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helmut.kuper at ewetel.de Guest
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Posted: Fri Oct 16, 2009 6:44 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello,
after updating FS to trunk a few days ago I found that callee's display
is updated serveral times to caller's name after callee picked up. The
first two equal INFO messages looks like this:
INFO sip:2850@85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bKU4803UHXaKD2r
Max-Forwards: 70
From: "Snom1 an PBX" <sip:4918@85.16.246.6>;tag=ccc4B2F9SD2rm
To: <sip:2850@85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
CSeq: 121727215 INFO
Contact: <sip:mod_sofia@85.16.246.6:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 27
From:
To: "unknown" 2850
The third one is this:
INFO sip:2850@85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK0H7vc35evZvZj
Max-Forwards: 70
From: "Snom1 an PBX" <sip:4918@85.16.246.6>;tag=ccc4B2F9SD2rm
To: <sip:2850@85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
CSeq: 121727216 INFO
Contact: <sip:mod_sofia@85.16.246.6:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 32
From:
To: "Snom1 an PBX" 4918
What calle sees is this:
1. Phone rings: caller's displayname
2. Callee picks up: switching from dislayname to unknown
3. Switching from unknown to displayname
After callee has picked up caller's display is also updated with
callee's name (this is a copy of callee's number rather than callee's
name) and number.
First Question:
Can this behaviour be disabled? Or can it be modified by dialplan, so
that there is no "unknown"?
Unfortunately when you do an outgoing call via openzap and callee picks
up the phone caller's display is updated with two equal INFO message
like this:
INFO sip:2850@85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK7jFa9Ztg39UUr
Max-Forwards: 70
From: <sip:890327@85.16.246.6;user=phone>;tag=9QZamXtmUyXvm
To: "Helmut Kuper" <sip:2850@85.16.246.6>;tag=5lf2e0q1on
Call-ID: 3c2a2539a966-0m3i9s85my46
CSeq: 121727707 INFO
Contact: <sip:890327@85.16.246.6:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 36
From:
To: "1/a/890327" 1/a/890327
Second Question:
Can I change the name and number via dialplan, so that the correkt name
and number is viewed to caller?
regards
helmut
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anthony.minessale at g... Guest
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Posted: Fri Oct 16, 2009 11:26 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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1) you should update again there were a few issues.
2) you can set the variable sip_callee_id_name and sip_callee_id number on the inbound leg before you answer to control what it says.
On Fri, Oct 16, 2009 at 6:31 AM, Helmut Kuper <helmut.kuper@ewetel.de (helmut.kuper@ewetel.de)> wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello,
after updating FS to trunk a few days ago I found that callee's display
is updated serveral times to caller's name after callee picked up. The
first two equal INFO messages looks like this:
INFO sip:2850@85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bKU4803UHXaKD2r
Max-Forwards: 70
From: "Snom1 an PBX" <sip:4918@85.16.246.6 ([email]sip%3A4918@85.16.246.6[/email])>;tag=ccc4B2F9SD2rm
To: <sip:2850@85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
CSeq: 121727215 INFO
Contact: <sip:mod_sofia@85.16.246.6:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 27
From:
To: "unknown" 2850
The third one is this:
INFO sip:2850@85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK0H7vc35evZvZj
Max-Forwards: 70
From: "Snom1 an PBX" <sip:4918@85.16.246.6 ([email]sip%3A4918@85.16.246.6[/email])>;tag=ccc4B2F9SD2rm
To: <sip:2850@85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
CSeq: 121727216 INFO
Contact: <sip:mod_sofia@85.16.246.6:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 32
From:
To: "Snom1 an PBX" 4918
What calle sees is this:
1. Phone rings: caller's displayname
2. Callee picks up: switching from dislayname to unknown
3. Switching from unknown to displayname
After callee has picked up caller's display is also updated with
callee's name (this is a copy of callee's number rather than callee's
name) and number.
First Question:
Can this behaviour be disabled? Or can it be modified by dialplan, so
that there is no "unknown"?
Unfortunately when you do an outgoing call via openzap and callee picks
up the phone caller's display is updated with two equal INFO message
like this:
INFO sip:2850@85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK7jFa9Ztg39UUr
Max-Forwards: 70
From: <sip:890327@85.16.246.6 ([email]sip%3A890327@85.16.246.6[/email]);user=phone>;tag=9QZamXtmUyXvm
To: "Helmut Kuper" <sip:2850@85.16.246.6 ([email]sip%3A2850@85.16.246.6[/email])>;tag=5lf2e0q1on
Call-ID: 3c2a2539a966-0m3i9s85my46
CSeq: 121727707 INFO
Contact: <sip:890327@85.16.246.6:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 36
From:
To: "1/a/890327" 1/a/890327
Second Question:
Can I change the name and number via dialplan, so that the correkt name
and number is viewed to caller?
regards
helmut
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=FGsp
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_______________________________________________
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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msc at freeswitch.org Guest
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Posted: Sat Oct 17, 2009 1:15 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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On Fri, Oct 16, 2009 at 9:18 AM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:
Quote: | 1) you should update again there were a few issues.
2) you can set the variable sip_callee_id_name and sip_callee_id number on the inbound leg before you answer to control what it says.
| Thanks for the heads up. I added these two vars to the wiki on the chan vars page, SIP-related vars section.
-MC |
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anthony.minessale at g... Guest
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Posted: Mon Oct 19, 2009 10:20 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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you only need to "set" it on the inbound leg and you must answer and bridge it somewhere.
On Mon, Oct 19, 2009 at 9:47 AM, Helmut Kuper <helmut.kuper@ewetel.de (helmut.kuper@ewetel.de)> wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello Anthony,
I updated and restarted my test FS to "FreeSWITCH Version 1.0.trunk
(15174M)". Callee's experience didn't change:
Quote: | 1. Phone rings: caller's displayname
2. Callee picks up: switching from dislayname to unknown
3. Switching from unknown to displayname
|
I used the two chvars you mentioned, set them via "set" and as well via
"export" but no change (neither on caller's nor in callee's display nor
in SIP INFO messages)
My dialplan portion for this is:
<extension name="Local_Extension">
<condition field="${ET_is_local}" expression="^true$">
<action application="set"
data="dialed_extension=${destination_number}"/>
<action application="set" data="sip_callee_id_number=1111"/>
<action application="set" data="sip_callee_id_name=hubu"/>
<action application="export" data="sip_callee_id_number=1111"/>
<action application="export" data="sip_callee_id_name=hubu"/>
<action application="info"/>
[...]
Here is the output of the info app after setting those chvars:
[INFO] mod_dptools.c:961 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001@85.16.246.12:5061]
Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [ringing]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001 an PBX1]
Caller-Caller-ID-Number: [1001]
Caller-Network-Addr: [85.16.245.206]
Caller-Destination-Number: [1000]
Caller-Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [1000]
Caller-Channel-Name: [sofia/internal/1001@85.16.246.12:5061]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1255963206242587]
Caller-Channel-Created-Time: [1255963206214959]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_sip_received_ip: [85.16.245.206]
variable_sip_received_port: [1024]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_from_user: [1001]
variable_sip_from_port: [5061]
variable_sip_from_uri: [1001@85.16.246.12:5061]
variable_sip_from_host: [85.16.246.12]
variable_sip_from_user_stripped: [1001]
variable_sip_from_tag: [snfuiue6ga]
variable_sofia_profile_name: [internal]
variable_sip_req_params: [user=phone]
variable_sip_req_user: [1000]
variable_sip_req_port: [5061]
variable_sip_req_uri: [1000@85.16.246.12:5061]
variable_sip_req_host: [85.16.246.12]
variable_sip_to_params: [user=phone]
variable_sip_to_user: [1000]
variable_sip_to_port: [5061]
variable_sip_to_uri: [1000@85.16.246.12:5061]
variable_sip_to_host: [85.16.246.12]
variable_sip_contact_params: [line=eg3wp69a]
variable_sip_contact_user: [1001]
variable_sip_contact_port: [1024]
variable_sip_contact_uri: [1001@85.16.245.206:1024]
variable_sip_contact_host: [85.16.245.206]
variable_channel_name: [sofia/internal/1001@85.16.246.12:5061]
variable_sip_call_id: [3c2d2d8f9a49-edzr2i2iezjp]
variable_sip_user_agent: [snom820/8.2.16]
variable_sip_via_host: [85.16.245.206]
variable_sip_via_port: [1024]
variable_sip_via_rport: [1024]
variable_presence_id: [1001@85.16.246.12 (1001@85.16.246.12)]
variable_sip_h_X-Serialnumber: [0004134002CB]
variable_sip_h_P-Key-Flags: [resolution="31x13", keys="4"]
variable_switch_r_sdp: [v=0
o=root 1331667919 1331667919 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 62882 RTP/SAVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:m6fas/KsLF57r9RnU7X0WEWeJw9Y6+a66YUIf9Dc
a=ptime:20
m=audio 62882 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
]
variable_ep_codec_string: [G722@8000h@20i,PCMA@8000h@20i]
variable_endpoint_disposition: [DELAYED NEGOTIATION]
variable_ET_is_local: [true]
variable_max_forwards: [69]
variable_domain_name: [85.16.246.12]
variable_dialed_extension: [1000]
variable_sip_callee_id_number: [1111]
variable_sip_callee_id_name: [hubu]
variable_export_vars: [sip_callee_id_number,sip_callee_id_name]
variable_current_application: [info]
On 16.10.2009 18:18, Anthony Minessale wrote:
Quote: | 1) you should update again there were a few issues.
2) you can set the variable sip_callee_id_name and sip_callee_id number
on the inbound leg before you answer to control what it says.
|
regards
Helmut
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nNTz4r7+N7mI3Wj4GayFdTk=
=MYOb
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
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anthony.minessale at g... Guest
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Posted: Mon Oct 19, 2009 3:20 pm Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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please update and test trunk
1) I changed the core to remove the excess data by default in your scenario
2) I added variables you can use to control it origination_callee_id_name origination_callee_id_number which belong in {} in the dial string eg {origination_callee_id_number=1234}openzap/1/a/1234
On Mon, Oct 19, 2009 at 10:53 AM, Helmut Kuper <helmut.kuper@ewetel.de (helmut.kuper@ewetel.de)> wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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msc at freeswitch.org Guest
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Posted: Mon Oct 19, 2009 4:42 pm Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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On Mon, Oct 19, 2009 at 1:07 PM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:
Quote: | please update and test trunk
1) I changed the core to remove the excess data by default in your scenario
2) I added variables you can use to control it origination_callee_id_name origination_callee_id_number which belong in {} in the dial string eg {origination_callee_id_number=1234}openzap/1/a/1234
| After you test, please confirm the behavior and then we'll update the wiki on these two new chan vars.
-MC |
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msc at freeswitch.org Guest
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Posted: Tue Oct 20, 2009 11:34 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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On Tue, Oct 20, 2009 at 12:25 AM, Helmut Kuper <helmut.kuper@ewetel.de (helmut.kuper@ewetel.de)> wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello Anthony,
On 19.10.2009 22:07, Anthony Minessale wrote:
Quote: | please update and test trunk
1) I changed the core to remove the excess data by default in your scenario
2) I added variables you can use to control it
origination_callee_id_name origination_callee_id_number which belong in
{} in the dial string eg {origination_callee_id_number=1234}openzap/1/a/1234
|
updated and testet for SIP calls. "origination_callee_id_number=1234"
works within the dial string and via using "export" application but not
via "set" application. Didn't test it for openzap, yet but I guess it
will work, too.
|
Thanks for testing. BTW, according to Tony's instructions the user needs to put the variable definition inside the {} of the dialstring. This means that the vars are designed for the new leg, not the local leg, which means that "export" would work but "set" would not. (Remember, "set" is to set a chan var on the local channel, "export" will set a chan var locally *and* on the other call leg. See also the "nolocal" option of export.)
Quote: |
What left is that "unknown" thing on callee's display (SIP phones only I
guess) ... I would suggest to make sending "unknown" INFO message
optional, but to be honest, I have no idea for what it was invented, so
maybe my suggestion is nonsens.
|
Under what conditions did you see "unknown"? I'm wondering if the user can just pick a default other than "unknown" if he wants something else to be displayed.
Thoughts?
-MC |
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brian at freeswitch.org Guest
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Posted: Tue Oct 20, 2009 11:51 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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Or just set the var to what you want it to say?
/b
On Oct 20, 2009, at 11:19 AM, Michael Collins wrote:
Quote: |
Under what conditions did you see "unknown"? I'm wondering if the
user can just pick a default other than "unknown" if he wants
something else to be displayed.
Thoughts?
-MC
|
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msc at freeswitch.org Guest
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Posted: Wed Oct 21, 2009 12:57 pm Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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On Wed, Oct 21, 2009 at 10:43 AM, Helmut Kuper <helmut.kuper@ewetel.de (helmut.kuper@ewetel.de)> wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello Mike,
just updated my prod system. The "1/a/" problem is solved with Anthony's
"originate_callee_id_name" chvar.
thanks alot
So, last thing of this thread is still the "unknown" thing on callee's
display, which is (by now) NOT affected by the new chvars.
| Okay, you are able to reproduce that "unknown" thing? Can you pastebin a fresh debug log w/ SIP trace on, plus and relevant dp changes from the default dialplan?
Thanks,
MC |
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brian at freeswitch.org Guest
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Posted: Thu Oct 22, 2009 11:36 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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I can't get what exactly you re talking about. Can you clarify ? Also
please include the packets of interest only not the full trace if its
not relevant to the bug.
/b
On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi Mike,
here it is:
Dialplan:
<extension name="Local_Extension">
<condition field="destination_number" expression="^(10[01][0-9])
$">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="dialed_extension=$1"/>
<action application="set" data="transfer_ringback=$$
{hold_music}"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>
</condition>
</extension>
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anthony.minessale at g... Guest
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Posted: Thu Oct 22, 2009 12:18 pm Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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I'm sure that problem is gone in svn trunk.
On Thu, Oct 22, 2009 at 11:25 AM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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anthony.minessale at g... Guest
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Posted: Fri Oct 23, 2009 11:02 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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should be even better in 15210
On Fri, Oct 23, 2009 at 6:35 AM, Helmut Kuper <helmut.kuper@ewetel.de (helmut.kuper@ewetel.de)> wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi Anthony,
thanks! The "unknown" thing is gone as long as you don't use
"originate_callee_id_name"
The "originate_callee_id_name" chvar causes caller's AND callee's
display ("caller name" line)to be updated with the content of that chvar
as soon as you set it.
Let's say you use "hubu" as originate_callee_id_name. As soon as callee
has picked up the phone it's display ("caller name" line) is updated
with "hubu" and right after that with caller's name. The caller's
display ("callee name" line) is updated to "hubu" (as expected).
So it's still not nice to see a name switch in callee's display for
caller's name after picking up.
In the following you see the SIP flows from
a) caller to FS
b) FS to callee
The more interesting one is b) I guess
#######################################################
This is the sip flow from caller to FS, which is OK:
INVITE sip:1000@85.16.246.12:5061;user=phone SIP/2.0
23/Oct/2009-13:26:44.730
INVITE sip:1000@85.16.246.12:5061;user=phone SIP/2.0
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport
From: "1001 an PBX1" <sip:1001@85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000@85.16.246.12:5061;user=phone>
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:1001@85.16.245.206:1024;line=eg3wp69a>;reg-id=1
X-Serialnumber: 0004134002CB
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom820/8.2.16
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 734
v=0
o=root 1420196469 1420196469 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 51286 RTP/SAVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:CjdPN1m2iLKAyXrQ2pWkQfCMtiISlF94ItTdsDis
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=audio 51286 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-ev
SIP/2.0 100 Trying
23/Oct/2009-13:26:44.731
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
From: "1001 an PBX1" <sip:1001@85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000@85.16.246.12:5061;user=phone>
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Content-Length: 0
SIP/2.0 180 Ringing
23/Oct/2009-13:26:45.3
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
From: "1001 an PBX1" <sip:1001@85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000@85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
Contact: <sip:1000@85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
X-FS-Display-Name: hubu
X-FS-Display-Number:
X-FS-Support: update_display
SIP/2.0 200 OK
23/Oct/2009-13:26:47.367
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
From: "1001 an PBX1" <sip:1001@85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000@85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
Contact: <sip:1000@85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Session-Expires: 3600;refresher=uas
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 354
X-FS-Display-Name: hubu
X-FS-Display-Number:
X-FS-Support: update_display
v=0
o=FreeSWITCH 1256285701 1256285702 IN IP4 85.16.246.12
s=FreeSWITCH
c=IN IP4 85.16.246.12
t=0 0
m=audio 11506 RTP/SAVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:/BFm/LUmDBuao0wqg6MS/l4acfzOHoHRFQr3N0R9
m=audio 0 RTP/AVP 19
ACK sip:1000@85.16.246.12:5061;transport=udp SIP/2.0
23/Oct/2009-13:26:47.392
ACK sip:1000@85.16.246.12:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-d474to81d7gh;rport
From: "1001 an PBX1" <sip:1001@85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000@85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:1001@85.16.245.206:1024;line=eg3wp69a>;reg-id=1
Content-Length: 0
INFO sip:1001@85.16.245.206:1024;line=eg3wp69a SIP/2.0
23/Oct/2009-13:26:47.416
INFO sip:1001@85.16.245.206:1024;line=eg3wp69a SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKB244eKZ08Uj3H
Max-Forwards: 70
From: <sip:1000@85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001@85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030267 INFO
Contact: <sip:1000@85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 24
From:
To: "hubu" 1000
SIP/2.0 200 Ok
23/Oct/2009-13:26:47.619
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKB244eKZ08Uj3H
From: <sip:1000@85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001@85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030267 INFO
Contact: <sip:1001@85.16.245.206:1024;line=eg3wp69a>;reg-id=1
Content-Length: 0
BYE sip:1001@85.16.245.206:1024;line=eg3wp69a SIP/2.0
23/Oct/2009-13:26:49.18
BYE sip:1001@85.16.245.206:1024;line=eg3wp69a SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKDmQpj9072DZ8r
Max-Forwards: 70
From: <sip:1000@85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001@85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030268 BYE
Contact: <sip:1000@85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
SIP/2.0 200 OK
23/Oct/2009-13:26:49.52
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKDmQpj9072DZ8r
From: <sip:1000@85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001@85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030268 BYE
Contact: <sip:1001@85.16.245.206:1024;line=eg3wp69a>;reg-id=1
User-Agent: snom820/8.2.16
RTP-RxStat: Total_Rx_Pkts=65,Rx_Pkts=65,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=94,Remote_Tx_Pkts=0
Content-Length: 0
####################################################################
This is the SIP flow from FS to callee.
The first INFO sent from FS to callee is the Problem. It contains "hubu"
as "To:" header. If there is s need for this message I guess the "From"
header should be changed, because from the callee-phone's point of view
the "To:" header is the caller and the "From" header is the callee. In
my opinion this INFO message shouldn't be sent at all, but I don't know
what you intend it to do.
INVITE sip:1000@85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:44.978
INVITE sip:1000@85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bK86rt91cpH1FBF
Max-Forwards: 68
From: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
To: <sip:1000@85.16.245.228:1024;line=5jkc7koy>
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:mod_sofia@85.16.246.12:5061>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 325
X-Serialnumber: 0004134002CB
P-Key-Flags: resolution="31x13", keys="4"
X-FS-Support: update_display
Remote-Party-ID: "1001 an PBX1"
<sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1256285400 1256285401 IN IP4 85.16.246.12
s=FreeSWITCH
c=IN IP4 85.16.246.12
t=0 0
m=audio 11804 RTP/AVP 95 103 9 8 101 13
a=rtpmap:95 CELT/48000
a=rtpmap:103 SPEEX/32000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:10
SIP/2.0 180 Ringing
23/Oct/2009-13:26:45.2
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
To: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0
SIP/2.0 180 Ringing
23/Oct/2009-13:26:45.504
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
To: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0
SIP/2.0 180 Ringing
23/Oct/2009-13:26:46.504
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
To: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0
SIP/2.0 200 Ok
23/Oct/2009-13:26:47.361
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
To: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;reg-id=1
User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 353690302 353690303 IN IP4 85.16.245.228
s=call
c=IN IP4 85.16.245.228
t=0 0
m=audio 60878 RTP/AVP 9 8 101
a=rtpmap:9 g722/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:10
a=sendrecv
ACK sip:1000@85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:47.362
ACK sip:1000@85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bK9FjKBXXSea6Xa
Max-Forwards: 70
From: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
To: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 ACK
Contact: <sip:mod_sofia@85.16.246.12:5061>
Content-Length: 0
INFO sip:1000@85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:47.367
INFO sip:1000@85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKaSBcDreXBKvgp
Max-Forwards: 70
From: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
To: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030267 INFO
Contact: <sip:mod_sofia@85.16.246.12:5061>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 24
From:
To: "hubu" 1000
SIP/2.0 200 Ok
23/Oct/2009-13:26:47.766
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKaSBcDreXBKvgp
From: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
To: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030267 INFO
Contact: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Content-Length: 0
INFO sip:1000@85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:47.766
INFO sip:1000@85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKcByXgeg4548ND
Max-Forwards: 70
From: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
To: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030268 INFO
Contact: <sip:mod_sofia@85.16.246.12:5061>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 32
From:
To: "1001 an PBX1" 1001
SIP/2.0 200 Ok
23/Oct/2009-13:26:47.860
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKcByXgeg4548ND
From: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
To: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030268 INFO
Contact: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Content-Length: 0
BYE sip:mod_sofia@85.16.246.12:5061 SIP/2.0
23/Oct/2009-13:26:49.3
BYE sip:mod_sofia@85.16.246.12:5061 SIP/2.0
Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-susgzpvc38s7;rport
From: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
To: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 1 BYE
Max-Forwards: 70
Contact: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;reg-id=1
User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=69,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=70,Tx_Pkts=65,Remote_Tx_Pkts=0
Content-Length: 0
SIP/2.0 200 OK
23/Oct/2009-13:26:49.4
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-susgzpvc38s7;rport=1024
From: <sip:1000@85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
To: "1001 an PBX1" <sip:1001@85.16.246.12 ([email]sip%3A1001@85.16.246.12[/email])>;tag=rp27e1KSe969D
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 1 BYE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0
On 22.10.2009 19:08, Anthony Minessale wrote:
Quote: | I'm sure that problem is gone in svn trunk.
On Thu, Oct 22, 2009 at 11:25 AM, Brian West <brian@freeswitch.org (brian@freeswitch.org)
|
Quote: | <mailto:brian@freeswitch.org (brian@freeswitch.org)>> wrote:
I can't get what exactly you re talking about. Can you clarify ? Also
please include the packets of interest only not the full trace if its
not relevant to the bug.
/b
On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Mike,
>
> here it is:
>
>
> Dialplan:
>
>
> <extension name="Local_Extension">
> <condition field="destination_number" expression="^(10[01][0-9])
> $">
> <action application="set" data="dialed_extension=$1"/>
> <action application="export" data="dialed_extension=$1"/>
> <action application="set" data="transfer_ringback=$$
> {hold_music}"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="bridge"
> data="user/${dialed_extension}@${domain_name}"/>
> </condition>
> </extension>
>
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Quote: | <mailto:MSN%3Aanthony_minessale@hotmail.com ([email]MSN%253Aanthony_minessale@hotmail.com[/email])>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
|
Quote: | <mailto:PAYPAL%3Aanthony.minessale@gmail.com ([email]PAYPAL%253Aanthony.minessale@gmail.com[/email])>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
|
- --
Mit freundlichen Grüßen
Helmut Kuper
Geschäftseinheit FD - Lösungen für Finanzdienstleister
Telefax: (0441) 8000-2799
mailto:helmut.kuper@ewetel.de (helmut.kuper@ewetel.de)
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Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole
Homepage: http://www.ewetel.de
___________________________________
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jkUQIoNN6CYJFp10ebA5MIU=
=cHej
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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anthony.minessale at g... Guest
|
Posted: Mon Oct 26, 2009 10:10 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
|
|
Could you maybe consolidate all of your problems into 1 thread. I am getting dizzy. You have 2 on the same subject and you say it works on one and does not on the other.
Last week we tested all of this with latest trunk and there is no longer any problems of any sort with the display related stuff.
On Mon, Oct 26, 2009 at 6:00 AM, Helmut Kuper <helmut.kuper@ewetel.de (helmut.kuper@ewetel.de)> wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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anthony.minessale at g... Guest
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Posted: Mon Oct 26, 2009 11:23 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
|
|
depending on your dialplan every time you bridge to a channel it changes the display to match who you are talking to. if you tried to set it with the variable then you call someone that is going to cause this. Take away the display app and/or any special variables and let it naturally work.
On Mon, Oct 26, 2009 at 10:51 AM, Helmut Kuper <helmut.kuper@ewetel.de (helmut.kuper@ewetel.de)> wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi Anthony,
sorry for making you dizzy ... bun in fact in my point of view I have
two different problems.
1.
One concerns the way using send_display in pre_answer mode. Simply to
send error texts to caller's display. This works again with latest trunk.
2.
The other one (this thread) concerns the (in my eyes newly introduced)
two INFO messages which FS sends to callee after callee picked up his
phone. The first INFO switches callee's display to a name set in
"originaton_callee_id_name" (a) and immediately after that the second
INFO switches it back to callee's real name (b). You can see this in
display only if (a) and (b) are not the same.
I used Snom 370 phones with FW 8.2.16 as caller and callee. I did an
internal call (sip to sip).
Maybe this is the same code problem, but on my level they are two
different problems, so sorry for confusing you. I hope this clears
things up.
On 26.10.2009 15:41, Anthony Minessale wrote:
Quote: | Could you maybe consolidate all of your problems into 1 thread. I am
getting dizzy. You have 2 on the same subject and you say it works on
one and does not on the other.
Last week we tested all of this with latest trunk and there is no longer
any problems of any sort with the display related stuff.
|
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fgVIZwAV/IthjWwvXzRO3TA=
=n7Sr
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http://www.freeswitch.org
|
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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