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[Freeswitch-users] Polycom configuration problems?

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larclap at yahoo.com
Guest

Posted: Mon Jun 22, 2009 3:50 pm Post subject: [Freeswitch-users] Polycom configuration problems?

I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001@192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001@192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001@192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001@192.168.10.29) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001@192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001@192.168.10.29) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001@192.168.10.29) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001@192.168.10.29 SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001@192.168.10.29 Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001@192.168.10.29 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001@192.168.10.29 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001@192.168.10.29 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false

chris at cloudtel.com
Guest

Posted: Mon Jun 22, 2009 5:02 pm Post subject: [Freeswitch-users] Polycom configuration problems?

Sounds like a config issue in the <dialplan/> tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or .digitmap.timer settings. When you dial off-hook it sure will.

On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote:

Quote:

I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch.

When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic.

However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine.

You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch.

Thanks for any suggestions, Lars.

PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux

2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth.
2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth.
2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [661a46fa-5f63-11de-85d0-1157463d23c2]
2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_NEW
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) entering state [received][100]
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP:
v=0
o=- 1245682011 1245682011 IN IP4 192.168.10.101
s=Polycom IP Phone
c=IN IP4 192.168.10.101
t=0 0
m=audio 2254 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State NEW
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) PCMU/8000 20 ms 160 samples
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_NEW -> CS_INIT
2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK]
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_INIT
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA INIT
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_INIT -> CS_ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK]
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT going to sleep
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State ROUTING
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Standard ROUTING
2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) parsing [default->unloop] continue=false
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false

_______________________________________________
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Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
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larclap at yahoo.com
Guest

Posted: Tue Jun 23, 2009 8:28 am Post subject: [Freeswitch-users] Polycom configuration problems?

I’m sorry Chris, but I don’t know where the look for the “global sip.cfg and mac/phone specific cfg” settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Burns Sent: Monday, June 22, 2009 2:57 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the <dialplan/> tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ..digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) parsing [default->unloop] continue=false Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

rupa at rupa.com
Guest

Posted: Tue Jun 23, 2009 10:39 am Post subject: [Freeswitch-users] Polycom configuration problems?

How are you configuring your polycom?

On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote:

Quote:

I’m sorry Chris, but I don’t know where the look for the “global sip.cfg and mac/phone specific cfg” settings. I also looked for digitmap but could find nothing.

Can you be more specific?

Thanks, Lars

From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Chris Burns
Sent: Monday, June 22, 2009 2:57 PM
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Subject: Re: [Freeswitch-users] Polycom configuration problems?

Sounds like a config issue in the <dialplan/> tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ..digitmap.timer settings. When you dial off-hook it sure will.

On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote:
I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch.

When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic.

However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine.

You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch.

Thanks for any suggestions, Lars.

PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux

2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth.
2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth.
2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [661a46fa-5f63-11de-85d0-1157463d23c2]
2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_NEW
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) entering state [received][100]
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP:
v=0
o=- 1245682011 1245682011 IN IP4 192.168.10.101
s=Polycom IP Phone
c=IN IP4 192.168.10.101
t=0 0
m=audio 2254 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State NEW
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) PCMU/8000 20 ms 160 samples
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_NEW -> CS_INIT
2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK]
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_INIT
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA INIT
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_INIT -> CS_ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK]
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT going to sleep
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State ROUTING
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Standard ROUTING
2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) parsing [default->unloop] continue=false
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false

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--
-Rupa

chris at cloudtel.com
Guest

Posted: Tue Jun 23, 2009 11:47 am Post subject: [Freeswitch-users] Polycom configuration problems?

Basically read the polycom manual ... it is the polycom producing the dialtone and deciding when to dial the number you are entering, using its own dialplan and interdigit timers.

On Tue, Jun 23, 2009 at 10:38 AM, Rupa Schomaker <rupa@rupa.com (rupa@rupa.com)> wrote:

Quote:

How are you configuring your polycom?

On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote:

Quote:

--
-Rupa

_______________________________________________
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larclap at yahoo.com
Guest

Posted: Tue Jun 23, 2009 12:27 pm Post subject: [Freeswitch-users] Polycom configuration problems?

Via a web browser. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 8:39 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote: I’m sorry Chris, but I don’t know where the look for the “global sip.cfg and mac/phone specific cfg” settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Chris Burns Sent: Monday, June 22, 2009 2:57 PM To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org) Subject: Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the <dialplan/> tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ...digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) parsing [default->unloop] continue=false Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa

rupa at rupa.com
Guest

Posted: Tue Jun 23, 2009 12:47 pm Post subject: [Freeswitch-users] Polycom configuration problems?

Ok, most of us configure the polycoms via a provisioning interface. usually ftp or http.

Anyway, when using the web interface, you want to look at:

Goto the web interface, Click on SIP.

Scroll down to the Local Settings section and you need to modify digitmap and digitmap timeout. the syntax is in the polycom manuals which you can donwload from polycom.

On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote:

Quote:

Via a web browser.

From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Rupa Schomaker
Sent: Tuesday, June 23, 2009 8:39 AM

To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Subject: Re: [Freeswitch-users] Polycom configuration problems?

How are you configuring your polycom?
On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote:
I’m sorry Chris, but I don’t know where the look for the “global sip.cfg and mac/phone specific cfg” settings. I also looked for digitmap but could find nothing.

Can you be more specific?

Thanks, Lars

From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Chris Burns
Sent: Monday, June 22, 2009 2:57 PM
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Subject: Re: [Freeswitch-users] Polycom configuration problems?

Sounds like a config issue in the <dialplan/> tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ...digitmap.timer settings. When you dial off-hook it sure will.
On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote:
I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch.

When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic.

However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine.

You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch.

Thanks for any suggestions, Lars.

PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux

2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth.
2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth.
2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [661a46fa-5f63-11de-85d0-1157463d23c2]
2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_NEW
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) entering state [received][100]
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP:
v=0
o=- 1245682011 1245682011 IN IP4 192.168.10.101
s=Polycom IP Phone
c=IN IP4 192.168.10.101
t=0 0
m=audio 2254 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State NEW
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) PCMU/8000 20 ms 160 samples
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_NEW -> CS_INIT
2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK]
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_INIT
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA INIT
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_INIT -> CS_ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK]
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT going to sleep
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State ROUTING
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Standard ROUTING
2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) parsing [default->unloop] continue=false
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false

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Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
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http://www.freeswitch.org

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--
-Rupa

_______________________________________________
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http://www.freeswitch.org

--
-Rupa

larclap at yahoo.com
Guest

Posted: Tue Jun 23, 2009 5:00 pm Post subject: [Freeswitch-users] Polycom configuration problems?

Thanks to Rupa and Chris for this help. I didn’t know enough to understand Chris was pointing me to the Polycom phone rather than FS. I would never have figured this out. Are Polycoms the only SIP phones which have this feature? Lars From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 10:46 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Ok, most of us configure the polycoms via a provisioning interface. usually ftp or http. Anyway, when using the web interface, you want to look at: Goto the web interface, Click on SIP. Scroll down to the Local Settings section and you need to modify digitmap and digitmap timeout. the syntax is in the polycom manuals which you can donwload from polycom. On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote: Via a web browser. From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 8:39 AM To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org) Subject: Re: [Freeswitch-users] Polycom configuration problems? How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote: I’m sorry Chris, but I don’t know where the look for the “global sip.cfg and mac/phone specific cfg” settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Chris Burns Sent: Monday, June 22, 2009 2:57 PM To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org) Subject: Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the <dialplan/> tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ....digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) parsing [default->unloop] continue=false Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa

brian at freeswitch.org
Guest

Posted: Tue Jun 23, 2009 5:05 pm Post subject: [Freeswitch-users] Polycom configuration problems?

Nope other phones have this also./b

On Jun 23, 2009, at 4:57 PM, Lars Zeb wrote:

Quote:

Brian West
brian@freeswitch.org (brian@freeswitch.org)

-- Meet us at ClueCon! http://www.cluecon.com

rupa at rupa.com
Guest

Posted: Tue Jun 23, 2009 5:06 pm Post subject: [Freeswitch-users] Polycom configuration problems?

Every sip phone I've used has this feature. Even ATAs -- though they tend to ship with more forgiving defaults.

On Tue, Jun 23, 2009 at 4:57 PM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote:

Quote:

Thanks to Rupa and Chris for this help. I didn’t know enough to understand Chris was pointing me to the Polycom phone rather than FS. I would never have figured this out.

Are Polycoms the only SIP phones which have this feature?

Lars

From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Rupa Schomaker
Sent: Tuesday, June 23, 2009 10:46 AM

To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Subject: Re: [Freeswitch-users] Polycom configuration problems?

Ok, most of us configure the polycoms via a provisioning interface. usually ftp or http.

Anyway, when using the web interface, you want to look at:

Goto the web interface, Click on SIP.

Scroll down to the Local Settings section and you need to modify digitmap and digitmap timeout. the syntax is in the polycom manuals which you can donwload from polycom.
On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote:
Via a web browser.

From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Rupa Schomaker
Sent: Tuesday, June 23, 2009 8:39 AM

To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Subject: Re: [Freeswitch-users] Polycom configuration problems?

How are you configuring your polycom?
On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote:
I’m sorry Chris, but I don’t know where the look for the “global sip.cfg and mac/phone specific cfg” settings. I also looked for digitmap but could find nothing.

Can you be more specific?

Thanks, Lars

From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Chris Burns
Sent: Monday, June 22, 2009 2:57 PM
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Subject: Re: [Freeswitch-users] Polycom configuration problems?

Sounds like a config issue in the <dialplan/> tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ....digitmap.timer settings. When you dial off-hook it sure will.
On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb <larclap@yahoo.com (larclap@yahoo.com)> wrote:
I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch.

When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic.

However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine.

You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch.

Thanks for any suggestions, Lars.

PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux

2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth.
2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth.
2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [661a46fa-5f63-11de-85d0-1157463d23c2]
2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_NEW
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) entering state [received][100]
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP:
v=0
o=- 1245682011 1245682011 IN IP4 192.168.10.101
s=Polycom IP Phone
c=IN IP4 192.168.10.101
t=0 0
m=audio 2254 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State NEW
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) PCMU/8000 20 ms 160 samples
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_NEW -> CS_INIT
2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK]
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_INIT
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA INIT
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State Change CS_INIT -> CS_ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) [BREAK]
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State INIT going to sleep
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) Running State Change CS_ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001@192.168.10.29 (1001@192.168.10.29)) State ROUTING
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) SOFIA ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Standard ROUTING
2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) parsing [default->unloop] continue=false
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001@192.168.10.29 (1001@192.168.10.29) Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false

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-Rupa

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-Rupa

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