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[Freeswitch-users] Both phone rang, but no voice

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adeel.gnome at gmail.com
Guest

Posted: Thu Aug 21, 2008 11:23 pm Post subject: [Freeswitch-users] Both phone rang, but no voice

Got it, under /conf/autoload_configs/switch.conf.xml
But why its rounding odd port to even?? I got this message on restart after changing the port range.

=====
2008-08-22 12:19:38 [WARNING] switch_core_port_allocator.c:78 switch_core_port_allocator_new() Rounding odd end port 10039 to 10038
=====

Thanks.

On Fri, Aug 22, 2008 at 9:45 AM, Adeel Ansari <adeel.gnome@gmail.com (adeel.gnome@gmail.com)> wrote:

Quote:

Yes, its working. Actually, one of my colleague was working on something else, so he did something stupid on the public gateway settings. Is there any way to control the rtp port generation? I mean, something where we can define minimum and maximum port range for RTP. I have seen it generates, usually, 20000 - 35000. But I am not sure.

Thanks for your helpful responses.

On Thu, Aug 21, 2008 at 11:00 PM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote:

Quote:

We are already auto-adjusting to the nat issue here, but you likely have a firewall on blocking the rtp traffic, please turn off your firewall and test again.

Mike

On Aug 21, 2008, at 4:02 AM, Adeel Ansari wrote:

Quote:

I am not using any ip phone. Its normal mobile phones.
Moreover, I tried to change rtp-ip parameter to my public ip. Now its showing my public ip in the log, not the local one. But crashed with the error below.

=====
2008-08-21 16:00:03 [DEBUG] sofia_glue.c:1756 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/01160176905074] 124.82.137.118 port 25456 -> 130.94.88.94 port 13678 codec: 0 ms: 20
2008-08-21 16:00:03 [DEBUG] switch_rtp.c:813 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms
2008-08-21 16:00:03 [ERR] sofia_glue.c:1975 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-08-21 16:00:03 [NOTICE] sofia_glue.c:1976 sofia_glue_activate_rtp() Hangup sofia/external/01160176905074 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
2008-08-21 16:00:03 [DEBUG] switch_channel.c:1362 switch_channel_perform_hangup() Kill sofia/external/01160176905074 [KILL]
2008-08-21 16:00:03 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Kill sofia/external/01160176905074 [BREAK]
2008-08-21 16:00:03 [ERR] sofia.c:2509 sofia_handle_sip_i_state() RTP Error!
2008-08-21 16:00:03 [DEBUG] sofia.c:208 sofia_event_callback() Channel is already hungup.
=====

Thanks.

On Thu, Aug 21, 2008 at 3:49 PM, Jonas Gauffin <jonas.gauffin@gmail.com (jonas.gauffin@gmail.com)> wrote:

Quote:

Your phones should be configured to use STUN. (in your phone web admin
interfaces)

On Thu, Aug 21, 2008 at 9:46 AM, Adeel Ansari <adeel.gnome@gmail.com (adeel.gnome@gmail.com)> wrote:

Quote:

Right. I guess, both phones are reporting local ips. Whats next. I have
tried to change the STUN to stun:stun.freeswitch.org, but didn't work,
nothing changed same local ip. I have tried to give ext-rtp-ip my public ip,
but no work. Any suggestions?

Thanks for your efforts.

On Thu, Aug 21, 2008 at 3:26 PM, Jonas Gauffin <jonas.gauffin@gmail.com (jonas.gauffin@gmail.com)>
wrote:

Quote:

None of the phones are on the same lan as freeswitch, right?
If so, one of the phones do not use stun, but reports it's local ip to FS.

2008-08-21 15:15:58 [DEBUG] sofia_glue.c:1756
sofia_glue_activate_rtp() AUDIO RTP [sofia/external/01160122263828]
192.168.253.101 port 23678 -> 130.94.88.93 port 10116 codec: 0 ms: 20

On Thu, Aug 21, 2008 at 9:16 AM, Adeel Ansari <adeel.gnome@gmail.com (adeel.gnome@gmail.com)>
wrote:

Quote:

Below is the log in debug mode.

==========
freeswitch@gnome> originate sofia/gateway/gizmo1/01160176905074
&bridge(sofia/gateway/gizmo2/01160122263828)
2008-08-21 15:15:54 [NOTICE] switch_channel.c:535
switch_channel_set_name()
New Channel sofia/external/01160176905074
[fdf9aab4-6f50-11dd-8cbb-f7afdd15cc69]
2008-08-21 15:15:54 [DEBUG] mod_sofia.c:2020 sofia_outgoing_channel()
sofia/external/01160176905074 State Change CS_NEW -> CS_INIT
2008-08-21 15:15:54 [DEBUG] switch_core_session.c:727
switch_core_session_signal_state_change() Kill
sofia/external/01160176905074
[BREAK]
2008-08-21 15:15:54 [DEBUG] switch_core_state_machine.c:365
switch_core_session_run() sofia/external/01160176905074 Running State
Change
CS_INIT
2008-08-21 15:15:54 [DEBUG] switch_core_state_machine.c:415
switch_core_session_run() (sofia/external/01160176905074) State INIT
2008-08-21 15:15:54 [DEBUG] mod_sofia.c:80 sofia_on_init()
sofia/external/01160176905074 SOFIA INIT
2008-08-21 15:15:54 [DEBUG] mod_sofia.c:107 sofia_on_init()
sofia/external/01160176905074 State Change CS_INIT -> CS_ROUTING
2008-08-21 15:15:54 [DEBUG] switch_core_session.c:727
switch_core_session_signal_state_change() Kill
sofia/external/01160176905074
[BREAK]
2008-08-21 15:15:54 [DEBUG] switch_core_state_machine.c:415
switch_core_session_run() (sofia/external/01160176905074) State INIT
going
to sleep
2008-08-21 15:15:54 [DEBUG] switch_core_state_machine.c:365
switch_core_session_run() sofia/external/01160176905074 Running State
Change
CS_ROUTING
2008-08-21 15:15:54 [DEBUG] switch_core_state_machine.c:420
switch_core_session_run() (sofia/external/01160176905074) State ROUTING
2008-08-21 15:15:54 [DEBUG] mod_sofia.c:119 sofia_on_routing()
sofia/external/01160176905074 SOFIA ROUTING
2008-08-21 15:15:54 [DEBUG] switch_ivr_originate.c:57
originate_on_routing()
sofia/external/01160176905074 State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2008-08-21 15:15:54 [DEBUG] switch_core_session.c:727
switch_core_session_signal_state_change() Kill
sofia/external/01160176905074
[BREAK]
2008-08-21 15:15:54 [DEBUG] switch_core_state_machine.c:420
switch_core_session_run() (sofia/external/01160176905074) State ROUTING
going to sleep
2008-08-21 15:15:54 [DEBUG] switch_core_state_machine.c:365
switch_core_session_run() sofia/external/01160176905074 Running State
Change
CS_CONSUME_MEDIA
2008-08-21 15:15:54 [DEBUG] switch_core_state_machine.c:442
switch_core_session_run() (sofia/external/01160176905074) State
CONSUME_MEDIA
2008-08-21 15:15:54 [DEBUG] sofia.c:200 sofia_event_callback() event
[nua_i_state] status [0][INVITE sent] session:
sofia/external/01160176905074
2008-08-21 15:15:54 [DEBUG] sofia.c:2145 sofia_handle_sip_i_state()
Channel
sofia/external/01160176905074 entering state [calling]
2008-08-21 15:15:54 [DEBUG] sofia.c:200 sofia_event_callback() event
[nua_r_invite] status [401][Unauthorized] session:
sofia/external/01160176905074
2008-08-21 15:15:54 [DEBUG] sofia_reg.c:1098
sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with
'Digest:"proxy01.sipphone.com":dasbit102:102'.
2008-08-21 15:15:54 [DEBUG] sofia.c:200 sofia_event_callback() event
[nua_i_state] status [0][INVITE sent] session:
sofia/external/01160176905074
2008-08-21 15:15:54 [DEBUG] sofia.c:2145 sofia_handle_sip_i_state()
Channel
sofia/external/01160176905074 entering state [calling]
2008-08-21 15:15:56 [DEBUG] sofia.c:2145 sofia_handle_sip_i_state()
Channel
sofia/external/01160176905074 entering state [ready]
2008-08-21 15:15:56 [DEBUG] sofia.c:2149 sofia_handle_sip_i_state()
Remote
SDP:
v=0
o=root 7709 7709 IN IP4 130.94.88.90
s=session
c=IN IP4 130.94.88.90
t=0 0
m=audio 19672 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

2008-08-21 15:15:56 [DEBUG] sofia_glue.c:2297 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000]
2008-08-21 15:15:56 [DEBUG] sofia_glue.c:1547
sofia_glue_tech_set_codec()
Set Codec sofia/external/01160176905074 PCMU/8000 20 ms 160 samples
2008-08-21 15:15:56 [DEBUG] sofia_glue.c:2260 sofia_glue_negotiate_sdp()
Set
2833 dtmf payload to 101
2008-08-21 15:15:56 [DEBUG] sofia_glue.c:1756 sofia_glue_activate_rtp()
AUDIO RTP [sofia/external/01160176905074] 192.168.253.101 port 31444 ->
130.94.88.90 port 19672 codec: 0 ms: 20
2008-08-21 15:15:56 [DEBUG] switch_rtp.c:813 switch_rtp_create()
Starting
timer [soft] 160 bytes per 20000ms
2008-08-21 15:15:56 [DEBUG] switch_ivr_originate.c:1370
switch_ivr_originate() Originate Resulted in Success:
[sofia/external/01160176905074]
2008-08-21 15:15:56 [DEBUG] mod_commands.c:1612 originate_function()
sofia/external/01160176905074 State Change CS_CONSUME_MEDIA ->
CS_EXECUTE
2008-08-21 15:15:56 [DEBUG] switch_core_session.c:727
switch_core_session_signal_state_change() Kill
sofia/external/01160176905074
[BREAK]
2008-08-21 15:15:56 [NOTICE] sofia.c:2507 sofia_handle_sip_i_state()
Channel
[sofia/external/01160176905074] has been answered
API CALL [originate(sofia/gateway/gizmo1/01160176905074
&bridge(sofia/gateway/gizmo2/01160122263828))] output:
+OK fdf9aab4-6f50-11dd-8cbb-f7afdd15cc69

freeswitch@gnome> 2008-08-21 15:15:56 [DEBUG]
switch_core_state_machine.c:442 switch_core_session_run()
(sofia/external/01160176905074) State CONSUME_MEDIA going to sleep
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:365
switch_core_session_run() sofia/external/01160176905074 Running State
Change
CS_EXECUTE
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:430
switch_core_session_run() (sofia/external/01160176905074) State EXECUTE
2008-08-21 15:15:56 [DEBUG] mod_sofia.c:156 sofia_on_execute()
sofia/external/01160176905074 SOFIA EXECUTE
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:129
switch_core_standard_on_execute() Standard EXECUTE
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:140
switch_core_standard_on_execute() sofia/external/01160176905074 Execute
bridge(sofia/gateway/gizmo2/01160122263828)
2008-08-21 15:15:56 [NOTICE] switch_channel.c:535
switch_channel_set_name()
New Channel sofia/external/01160122263828
[ff8093e8-6f50-11dd-8cbb-f7afdd15cc69]
2008-08-21 15:15:56 [DEBUG] mod_sofia.c:2020 sofia_outgoing_channel()
sofia/external/01160122263828 State Change CS_NEW -> CS_INIT
2008-08-21 15:15:56 [DEBUG] switch_core_session.c:727
switch_core_session_signal_state_change() Kill
sofia/external/01160122263828
[BREAK]
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:365
switch_core_session_run() sofia/external/01160122263828 Running State
Change
CS_INIT
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:415
switch_core_session_run() (sofia/external/01160122263828) State INIT
2008-08-21 15:15:56 [DEBUG] mod_sofia.c:80 sofia_on_init()
sofia/external/01160122263828 SOFIA INIT
2008-08-21 15:15:56 [DEBUG] mod_sofia.c:107 sofia_on_init()
sofia/external/01160122263828 State Change CS_INIT -> CS_ROUTING
2008-08-21 15:15:56 [DEBUG] switch_core_session.c:727
switch_core_session_signal_state_change() Kill
sofia/external/01160122263828
[BREAK]
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:415
switch_core_session_run() (sofia/external/01160122263828) State INIT
going
to sleep
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:365
switch_core_session_run() sofia/external/01160122263828 Running State
Change
CS_ROUTING
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:420
switch_core_session_run() (sofia/external/01160122263828) State ROUTING
2008-08-21 15:15:56 [DEBUG] mod_sofia.c:119 sofia_on_routing()
sofia/external/01160122263828 SOFIA ROUTING
2008-08-21 15:15:56 [DEBUG] switch_ivr_originate.c:57
originate_on_routing()
sofia/external/01160122263828 State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2008-08-21 15:15:56 [DEBUG] switch_core_session.c:727
switch_core_session_signal_state_change() Kill
sofia/external/01160122263828
[BREAK]
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:420
switch_core_session_run() (sofia/external/01160122263828) State ROUTING
going to sleep
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:365
switch_core_session_run() sofia/external/01160122263828 Running State
Change
CS_CONSUME_MEDIA
2008-08-21 15:15:56 [DEBUG] switch_core_state_machine.c:442
switch_core_session_run() (sofia/external/01160122263828) State
CONSUME_MEDIA
2008-08-21 15:15:56 [DEBUG] sofia.c:200 sofia_event_callback() event
[nua_i_state] status [0][INVITE sent] session:
sofia/external/01160122263828
2008-08-21 15:15:56 [DEBUG] sofia.c:2145 sofia_handle_sip_i_state()
Channel
sofia/external/01160122263828 entering state [calling]
2008-08-21 15:15:56 [DEBUG] sofia.c:200 sofia_event_callback() event
[nua_r_invite] status [401][Unauthorized] session:
sofia/external/01160122263828
2008-08-21 15:15:56 [DEBUG] sofia_reg.c:1098
sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with
'Digest:"proxy01.sipphone.com":dasbit103:103'.
2008-08-21 15:15:56 [DEBUG] sofia.c:200 sofia_event_callback() event
[nua_i_state] status [0][INVITE sent] session:
sofia/external/01160122263828
2008-08-21 15:15:56 [DEBUG] sofia.c:2145 sofia_handle_sip_i_state()
Channel
sofia/external/01160122263828 entering state [calling]
2008-08-21 15:15:58 [DEBUG] sofia.c:2145 sofia_handle_sip_i_state()
Channel
sofia/external/01160122263828 entering state [ready]
2008-08-21 15:15:58 [DEBUG] sofia.c:2149 sofia_handle_sip_i_state()
Remote
SDP:
v=0
o=root 9902 9902 IN IP4 130.94.88.93
s=session
c=IN IP4 130.94.88.93
t=0 0
m=audio 10116 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

2008-08-21 15:15:58 [DEBUG] sofia_glue.c:2297 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000]
2008-08-21 15:15:58 [DEBUG] sofia_glue.c:1547
sofia_glue_tech_set_codec()
Set Codec sofia/external/01160122263828 PCMU/8000 20 ms 160 samples
2008-08-21 15:15:58 [DEBUG] sofia_glue.c:2260 sofia_glue_negotiate_sdp()
Set
2833 dtmf payload to 101
2008-08-21 15:15:58 [DEBUG] sofia_glue.c:1756 sofia_glue_activate_rtp()
AUDIO RTP [sofia/external/01160122263828] 192.168.253.101 port 23678 ->
130.94.88.93 port 10116 codec: 0 ms: 20
2008-08-21 15:15:58 [DEBUG] switch_rtp.c:813 switch_rtp_create()
Starting
timer [soft] 160 bytes per 20000ms
2008-08-21 15:15:58 [DEBUG] switch_channel.c:1541
switch_channel_perform_mark_answered() Kill
sofia/external/01160176905074
[BREAK]
2008-08-21 15:15:58 [NOTICE] sofia.c:2507 sofia_handle_sip_i_state()
Channel
[sofia/external/01160122263828] has been answered
2008-08-21 15:15:58 [DEBUG] switch_ivr_originate.c:1370
switch_ivr_originate() Originate Resulted in Success:
[sofia/external/01160122263828]
2008-08-21 15:15:58 [DEBUG] switch_core_session.c:435
switch_core_session_receive_message() Kill sofia/external/01160122263828
[BREAK]
2008-08-21 15:15:58 [DEBUG] switch_core_session.c:435
switch_core_session_receive_message() Kill sofia/external/01160176905074
[BREAK]
2008-08-21 15:15:58 [DEBUG] switch_ivr_bridge.c:778
switch_ivr_multi_threaded_bridge() sofia/external/01160122263828 State
Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2008-08-21 15:15:58 [DEBUG] switch_core_session.c:727
switch_core_session_signal_state_change() Kill
sofia/external/01160122263828
[BREAK]
2008-08-21 15:15:58 [DEBUG] switch_core_state_machine.c:442
switch_core_session_run() (sofia/external/01160122263828) State
CONSUME_MEDIA going to sleep
2008-08-21 15:15:58 [DEBUG] switch_core_state_machine.c:365
switch_core_session_run() sofia/external/01160122263828 Running State
Change
CS_EXCHANGE_MEDIA
2008-08-21 15:15:58 [DEBUG] switch_core_state_machine.c:433
switch_core_session_run() (sofia/external/01160122263828) State
EXCHANGE_MEDIA
2008-08-21 15:15:58 [DEBUG] mod_sofia.c:365 sofia_on_exchange_media()
SOFIA
LOOPBACK
==========

Thanks.

On Thu, Aug 21, 2008 at 3:07 PM, Jonas Gauffin <jonas.gauffin@gmail.com (jonas.gauffin@gmail.com)>
wrote:

Quote:

Show us logs of a call attempt.

On Thu, Aug 21, 2008 at 8:42 AM, Adeel Ansari <adeel.gnome@gmail.com (adeel.gnome@gmail.com)>
wrote:

Quote:

5080 and 5060 both are open. Moreover, other PBX is working fine on
5060.

Further, I have tried by setting the parameters sip-ip and rtp-ip to
auto;
ext-sip-ip and ext-sip-ip to my public ip. Nothing is working. Both
the
phone rang always, but no voice.

Anyother suggestions??
Thanks.

On Thu, Aug 21, 2008 at 1:01 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)>
wrote:

Quote:

Chech and see if a firewall is on! Also are you behind nat?
/b

Sent from my iPhone
On Aug 20, 2008, at 11:27 PM, "Adeel Ansari" <adeel.gnome@gmail.com (adeel.gnome@gmail.com)>
wrote:

Hi all,

I have tried the command below,

====
originate sofia/gateway/gizmo1/01160176xxxxxx
&bridge(sofia/gateway/gizmo2/01160122xxxxxx)
====

Its working, phone rang, but no voice. Am I missing something?

Thanks.

--
Best,
Adeel Ansari

http://www.linkedin.com/in/adeelansari

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--
Best,
Adeel Ansari

http://www.linkedin.com/in/adeelansari

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--
Best,
Adeel Ansari

http://www.linkedin.com/in/adeelansari

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--
Best,
Adeel Ansari

http://www.linkedin.com/in/adeelansari

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--
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Adeel Ansari

http://www.linkedin.com/in/adeelansari

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Adeel Ansari

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brian at freeswitch.org
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Posted: Thu Aug 21, 2008 11:34 pm Post subject: [Freeswitch-users] Both phone rang, but no voice

RTP happens on even ports. RTCP happens on odd ports.

/b

On Aug 21, 2008, at 11:19 PM, Adeel Ansari wrote:

Quote:

Got it, under /conf/autoload_configs/switch.conf.xml
But why its rounding odd port to even?? I got this message on
restart after changing the port range.

Brian West
sip:brian@freeswitch.org

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adeel.gnome at gmail.com
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Posted: Thu Aug 21, 2008 11:41 pm Post subject: [Freeswitch-users] Both phone rang, but no voice

Thanks for the info. Do I assume that if the range is, say

===
<param name="rtp-start-port" value="10020"/>
<param name="rtp-end-port" value="10040"/>
===

and the rtp port picked up is 10020 then the rtcp port would be 10021?
If not, where and how we will know about the RTCP ports. Because, I suppose, we need to open those as well, in the firewall.

Thanks for your inputs.

On Fri, Aug 22, 2008 at 12:32 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

Quote:

RTP happens on even ports. RTCP happens on odd ports.

/b

On Aug 21, 2008, at 11:19 PM, Adeel Ansari wrote:

Quote:

Got it, under /conf/autoload_configs/switch.conf.xml
But why its rounding odd port to even?? I got this message on
restart after changing the port range.

Brian West
sip:brian@freeswitch.org ([email]sip%3Abrian@freeswitch.org[/email])

_______________________________________________
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--
Best,
Adeel Ansari

http://www.linkedin.com/in/adeelansari

brian at freeswitch.org
Guest

Posted: Thu Aug 21, 2008 11:52 pm Post subject: [Freeswitch-users] Both phone rang, but no voice

We don't support RTCP yet.. but if we did it would be like this 10020 RTP 10021 RTCP .. and so on.

/b

On Aug 21, 2008, at 11:40 PM, Adeel Ansari wrote:

Quote:

RTP happens on even ports. RTCP happens on odd ports.

/b

On Aug 21, 2008, at 11:19 PM, Adeel Ansari wrote:

Quote:

Got it, under /conf/autoload_configs/switch.conf.xml
But why its rounding odd port to even?? I got this message on
restart after changing the port range.

--
Best,
Adeel Ansari

http://www.linkedin.com/in/adeelansari

_______________________________________________
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

Brian West
[url=sip:brian@freeswitch.org]sip:brian@freeswitch.org[/url]

adeel.gnome at gmail.com
Guest

Posted: Fri Aug 22, 2008 12:02 am Post subject: [Freeswitch-users] Both phone rang, but no voice

Righto. Thanks buddy.

On Fri, Aug 22, 2008 at 12:50 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

Quote:

We don't support RTCP yet.. but if we did it would be like this 10020 RTP 10021 RTCP .. and so on.

/b

On Aug 21, 2008, at 11:40 PM, Adeel Ansari wrote:

Quote:

RTP happens on even ports. RTCP happens on odd ports.

/b

On Aug 21, 2008, at 11:19 PM, Adeel Ansari wrote:

Quote:

Got it, under /conf/autoload_configs/switch.conf.xml
But why its rounding odd port to even?? I got this message on
restart after changing the port range.

--
Best,
Adeel Ansari

http://www.linkedin.com/in/adeelansari

_______________________________________________
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

Brian West
sip:brian@freeswitch.org

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--
Best,
Adeel Ansari

http://www.linkedin.com/in/adeelansari

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