[asterisk-bugs] [Asterisk 0018955]: after picking up a call, phones stop working...
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Mar 11 14:34:38 CST 2011
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=18955
======================================================================
Reported By: fcintron
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 18955
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.8.3
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2011-03-10 14:01 CST
Last Modified: 2011-03-11 14:34 CST
======================================================================
Summary: after picking up a call, phones stop working...
Description:
**************************
*Software versions:
**************************
Centos 5.5 32 bits
Asterisk 1.8.3
Dahdi 2.4.1
Libpri 1.4.11.5
Libss7 1.0.2
**************************
*Phones used
**************************
2 linksys ip phones SPA921(firmware version 5.1.8 )
1 normal phone connected to a Cisco SPA8800(firmware version 6.1.7)
********************************************
*Extensions assigned to phones
********************************************
First spa921 is 401 extension.
Second spa921 is 402 extension.
Normal phone connected to SPA8800 is 404 extension.
****************************************
*Config files
****************************************
*****features.conf***********************
[general]
parkext => 700
parkpos => 701-720
context => parkedcalls
transferdigittimeout => 300
pickupexten = *7
pickupsound = beep
pickupfailsound = beeperr
atxfernoanswertimeout = 25
[featuremap]
atxfer => *2
[applicationmap]
*****extensions.conf***********************
[globals]
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=yes
[default]
exten => s,1,Verbose(1,Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait()
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
[incoming_calls]
[internal]
exten => _4XX,1,Dial(SIP/${EXTEN},180,tT)
[phones]
include => internal
*****sip.conf***********************
[general]
[401]
type=friend
context=phones
host=dynamic
callgroup=1
pickupgroup=1
[402]
type=friend
context=phones
host=dynamic
callgroup=1
pickupgroup=1
[403]
type=friend
context=phones
host=dynamic
callgroup=1
pickupgroup=1
[404]
type=friend
qualify=yes
nat=no
host=dynamic
careinvite=no
context=phones
regext=404
callgroup=1
pickupgroup=1
**********************************************
*Steps to repeat problem
**********************************************
Step 1.
Description: From extension 401 call to 404. Leave 404 extension to sound,
don´t answer the call.
Step2.
Description: From 402 press *8# to pick up the call made in step 1. 402
will show in its graphics display that it is connected and 404 will stop to
sound. Presumably at this point 401 could talk with 402 but this is not the
case, 401 will keep showing in its graphics display "CalledPartyRinging".
Step3.
Description: Wait 10 seconds.
Step 4.
Description: Hang up 402 and 401.
After these steps I can not neither send nor receive calls from anyone of
401, 402 or 404 until I restart asterisk.
****************************************************
/var/log/asterisk/full
****************************************************
[Mar 7 12:45:33] VERBOSE[3764] config.c: == Parsing
'/etc/asterisk/logger.conf': [Mar 7 12:45:33] VERBOSE[3764] config.c: ==
Found
[Mar 7 12:45:33] VERBOSE[3764] logger.c: Asterisk Queue Logger
restarted
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c:
<--- SIP read from UDP:10.10.100.21:5060 --->
INVITE sip:404 at 10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.21:5060;branch=z9hG4bK-d9161a7a
From: "401" <sip:401 at 10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404 at 10.10.100.20>
Call-ID: f9e2055-43844cde at 10.10.100.21
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "401" <sip:401 at 10.10.100.21:5060>
Expires: 240
User-Agent: Linksys/SPA921-5.1.8
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 870064 870064 IN IP4 10.10.100.21
s=-
c=IN IP4 10.10.100.21
t=0 0
m=audio 16388 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: --- (14 headers 18 lines) ---
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Sending to 10.10.100.21:5060
(no NAT)
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Using INVITE request as basis
request - f9e2055-43844cde at 10.10.100.21
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found peer '401' for '401'
from 10.10.100.21:5060
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 10.10.100.21:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.10.100.21:5060;branch=z9hG4bK-d9161a7a;received=10.10.100.21
From: "401" <sip:401 at 10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404 at 10.10.100.20>;tag=as077d9460
Call-ID: f9e2055-43844cde at 10.10.100.21
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="0fccf840"
Content-Length: 0
<------------>
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Scheduling destruction of SIP
dialog 'f9e2055-43844cde at 10.10.100.21' in 32000 ms (Method: INVITE)
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c:
<--- SIP read from UDP:10.10.100.21:5060 --->
ACK sip:404 at 10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.21:5060;branch=z9hG4bK-d9161a7a
From: "401" <sip:401 at 10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404 at 10.10.100.20>;tag=as077d9460
Call-ID: f9e2055-43844cde at 10.10.100.21
CSeq: 101 ACK
Max-Forwards: 70
Contact: "401" <sip:401 at 10.10.100.21:5060>
User-Agent: Linksys/SPA921-5.1.8
Content-Length: 0
<------------->
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: --- (10 headers 0 lines) ---
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c:
<--- SIP read from UDP:10.10.100.21:5060 --->
INVITE sip:404 at 10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.21:5060;branch=z9hG4bK-d9654c1b
From: "401" <sip:401 at 10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404 at 10.10.100.20>
Call-ID: f9e2055-43844cde at 10.10.100.21
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest
username="401",realm="asterisk",nonce="0fccf840",uri="sip:404 at 10.10.100.20",algorithm=MD5,response="9bcdad8f85a785f01798e76d3b6be2bb"
Contact: "401" <sip:401 at 10.10.100.21:5060>
Expires: 240
User-Agent: Linksys/SPA921-5.1.8
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 870064 870064 IN IP4 10.10.100.21
s=-
c=IN IP4 10.10.100.21
t=0 0
m=audio 16388 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: --- (15 headers 18 lines) ---
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Sending to 10.10.100.21:5060
(no NAT)
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Using INVITE request as basis
request - f9e2055-43844cde at 10.10.100.21
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found peer '401' for '401'
from 10.10.100.21:5060
[Mar 7 12:45:40] VERBOSE[3734] netsock2.c: == Using SIP RTP CoS mark 5
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found RTP audio format 0
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found RTP audio format 2
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found RTP audio format 4
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found RTP audio format 8
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found RTP audio format 18
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found RTP audio format 96
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found RTP audio format 97
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found RTP audio format 98
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found RTP audio format 101
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found audio description format
PCMU for ID 0
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found audio description format
G726-32 for ID 2
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found audio description format
G723 for ID 4
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found audio description format
PCMA for ID 8
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found audio description format
G729a for ID 18
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found audio description format
G726-40 for ID 96
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found audio description format
G726-24 for ID 97
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found audio description format
G726-16 for ID 98
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Found audio description format
telephone-event for ID 101
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Capabilities: us -
0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x100d0d
(g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0
(nothing), combined - 0xc (ulaw|alaw)
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Non-codec capabilities (dtmf):
us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1
(telephone-event|)
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Peer audio RTP is at port
10.10.100.21:16388
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: Looking for 404 in phones
(domain 10.10.100.20)
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: list_route: hop:
<sip:401 at 10.10.100.21:5060>
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c:
<--- Transmitting (no NAT) to 10.10.100.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.10.100.21:5060;branch=z9hG4bK-d9654c1b;received=10.10.100.21
From: "401" <sip:401 at 10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404 at 10.10.100.20>
Call-ID: f9e2055-43844cde at 10.10.100.21
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:404 at 10.10.100.20:5060>
Content-Length: 0
<------------>
[Mar 7 12:45:40] VERBOSE[3770] pbx.c: -- Executing [404 at phones:1]
Dial("SIP/401-00000000", "SIP/404,180,tT") in new stack
[Mar 7 12:45:40] VERBOSE[3770] netsock2.c: == Using SIP RTP CoS mark 5
[Mar 7 12:45:40] VERBOSE[3770] chan_sip.c: Audio is at 5060
[Mar 7 12:45:40] VERBOSE[3770] chan_sip.c: Adding codec 0x4 (ulaw) to
SDP
[Mar 7 12:45:40] VERBOSE[3770] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[Mar 7 12:45:40] VERBOSE[3770] chan_sip.c: Adding codec 0x8 (alaw) to
SDP
[Mar 7 12:45:40] VERBOSE[3770] chan_sip.c: Adding codec 0x800000000000
(testlaw) to SDP
[Mar 7 12:45:40] VERBOSE[3770] chan_sip.c: Adding non-codec 0x1
(telephone-event) to SDP
[Mar 7 12:45:40] VERBOSE[3770] chan_sip.c: Reliably Transmitting (no NAT)
to 10.10.100.24:5060:
INVITE sip:404 at 10.10.100.24:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.20:5060;branch=z9hG4bK644fb7f0
Max-Forwards: 70
From: "401" <sip:401 at 10.10.100.20>;tag=as309652b0
To: <sip:404 at 10.10.100.24:5060>
Contact: <sip:401 at 10.10.100.20:5060>
Call-ID: 76d7de407cbc38ee5e786aec05e1349a at 10.10.100.20:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.3
Date: Mon, 07 Mar 2011 18:45:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 279
v=0
o=root 735388317 735388317 IN IP4 10.10.100.20
s=Asterisk PBX 1.8.3
c=IN IP4 10.10.100.20
t=0 0
m=audio 16550 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Mar 7 12:45:40] VERBOSE[3770] app_dial.c: -- Called 404
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c:
<--- SIP read from UDP:10.10.100.24:5060 --->
SIP/2.0 100 Trying
To: <sip:404 at 10.10.100.24:5060>
From: "401" <sip:401 at 10.10.100.20>;tag=as309652b0
Call-ID: 76d7de407cbc38ee5e786aec05e1349a at 10.10.100.20:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.10.100.20:5060;branch=z9hG4bK644fb7f0
Server: Cisco/SPA8800-6.1.7(GW)
Content-Length: 0
<------------->
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: --- (8 headers 0 lines) ---
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c:
<--- SIP read from UDP:10.10.100.24:5060 --->
SIP/2.0 180 Ringing
To: <sip:404 at 10.10.100.24:5060>;tag=57d02b562c2d5498i0
From: "401" <sip:401 at 10.10.100.20>;tag=as309652b0
Call-ID: 76d7de407cbc38ee5e786aec05e1349a at 10.10.100.20:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.10.100.20:5060;branch=z9hG4bK644fb7f0
Contact: "404" <sip:404 at 10.10.100.24:5060>
Server: Cisco/SPA8800-6.1.7(GW)
Remote-Party-ID: "404" <sip:404 at 10.10.100.20>;screen=yes;party=called
Content-Length: 0
<------------->
[Mar 7 12:45:40] VERBOSE[3734] chan_sip.c: --- (10 headers 0 lines) ---
[Mar 7 12:45:40] VERBOSE[3770] app_dial.c: -- SIP/404-00000001 is
ringing
[Mar 7 12:45:40] VERBOSE[3770] chan_sip.c:
<--- Transmitting (no NAT) to 10.10.100.21:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
10.10.100.21:5060;branch=z9hG4bK-d9654c1b;received=10.10.100.21
From: "401" <sip:401 at 10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404 at 10.10.100.20>;tag=as393690ba
Call-ID: f9e2055-43844cde at 10.10.100.21
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:404 at 10.10.100.20:5060>
Content-Length: 0
<------------>
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c:
<--- SIP read from UDP:10.10.100.22:5060 --->
INVITE sip:*7 at 10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.22:5060;branch=z9hG4bK-befd924d
From: "402" <sip:402 at 10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7 at 10.10.100.20>
Call-ID: 2a893106-30b58fd1 at 10.10.100.22
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "402" <sip:402 at 10.10.100.22:5060>
Expires: 240
User-Agent: Linksys/SPA921-5.1.8
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 674464 674464 IN IP4 10.10.100.22
s=-
c=IN IP4 10.10.100.22
t=0 0
m=audio 16422 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: --- (14 headers 18 lines) ---
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Sending to 10.10.100.22:5060
(no NAT)
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Using INVITE request as basis
request - 2a893106-30b58fd1 at 10.10.100.22
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found peer '402' for '402'
from 10.10.100.22:5060
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 10.10.100.22:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.10.100.22:5060;branch=z9hG4bK-befd924d;received=10.10.100.22
From: "402" <sip:402 at 10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7 at 10.10.100.20>;tag=as21b41cb0
Call-ID: 2a893106-30b58fd1 at 10.10.100.22
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="7c20bd27"
Content-Length: 0
<------------>
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Scheduling destruction of SIP
dialog '2a893106-30b58fd1 at 10.10.100.22' in 32000 ms (Method: INVITE)
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c:
<--- SIP read from UDP:10.10.100.22:5060 --->
ACK sip:*7 at 10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.22:5060;branch=z9hG4bK-befd924d
From: "402" <sip:402 at 10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7 at 10.10.100.20>;tag=as21b41cb0
Call-ID: 2a893106-30b58fd1 at 10.10.100.22
CSeq: 101 ACK
Max-Forwards: 70
Contact: "402" <sip:402 at 10.10.100.22:5060>
User-Agent: Linksys/SPA921-5.1.8
Content-Length: 0
<------------->
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: --- (10 headers 0 lines) ---
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c:
<--- SIP read from UDP:10.10.100.22:5060 --->
INVITE sip:*7 at 10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.22:5060;branch=z9hG4bK-83cecf31
From: "402" <sip:402 at 10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7 at 10.10.100.20>
Call-ID: 2a893106-30b58fd1 at 10.10.100.22
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest
username="402",realm="asterisk",nonce="7c20bd27",uri="sip:*7 at 10.10.100.20",algorithm=MD5,response="0666e728d9b46007267a11b56bd32960"
Contact: "402" <sip:402 at 10.10.100.22:5060>
Expires: 240
User-Agent: Linksys/SPA921-5.1.8
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 674464 674464 IN IP4 10.10.100.22
s=-
c=IN IP4 10.10.100.22
t=0 0
m=audio 16422 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: --- (15 headers 18 lines) ---
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Sending to 10.10.100.22:5060
(no NAT)
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Using INVITE request as basis
request - 2a893106-30b58fd1 at 10.10.100.22
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found peer '402' for '402'
from 10.10.100.22:5060
[Mar 7 12:45:49] VERBOSE[3734] netsock2.c: == Using SIP RTP CoS mark 5
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found RTP audio format 0
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found RTP audio format 2
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found RTP audio format 4
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found RTP audio format 8
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found RTP audio format 18
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found RTP audio format 96
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found RTP audio format 97
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found RTP audio format 98
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found RTP audio format 101
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found audio description format
PCMU for ID 0
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found audio description format
G726-32 for ID 2
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found audio description format
G723 for ID 4
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found audio description format
PCMA for ID 8
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found audio description format
G729a for ID 18
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found audio description format
G726-40 for ID 96
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found audio description format
G726-24 for ID 97
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found audio description format
G726-16 for ID 98
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Found audio description format
telephone-event for ID 101
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Capabilities: us -
0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x100d0d
(g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0
(nothing), combined - 0xc (ulaw|alaw)
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Non-codec capabilities (dtmf):
us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1
(telephone-event|)
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Peer audio RTP is at port
10.10.100.22:16422
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Looking for *7 in phones
(domain 10.10.100.20)
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: list_route: hop:
<sip:402 at 10.10.100.22:5060>
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c:
<--- Transmitting (no NAT) to 10.10.100.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.10.100.22:5060;branch=z9hG4bK-83cecf31;received=10.10.100.22
From: "402" <sip:402 at 10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7 at 10.10.100.20>
Call-ID: 2a893106-30b58fd1 at 10.10.100.22
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:*7 at 10.10.100.20:5060>
Content-Length: 0
<------------>
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Audio is at 5060
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Adding codec 0x4 (ulaw) to
SDP
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Adding codec 0x8 (alaw) to
SDP
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c: Adding non-codec 0x1
(telephone-event) to SDP
[Mar 7 12:45:49] VERBOSE[3734] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 10.10.100.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.10.100.22:5060;branch=z9hG4bK-83cecf31;received=10.10.100.22
From: "402" <sip:402 at 10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7 at 10.10.100.20>;tag=as778b6ec7
Call-ID: 2a893106-30b58fd1 at 10.10.100.22
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:*7 at 10.10.100.20:5060>
Content-Type: application/sdp
Content-Length: 258
v=0
o=root 1058168652 1058168652 IN IP4 10.10.100.20
s=Asterisk PBX 1.8.3
c=IN IP4 10.10.100.20
t=0 0
m=audio 18166 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=ptime:20
a=sendrecv
<------------>
[Mar 7 12:45:49] VERBOSE[3770] chan_sip.c: Scheduling destruction of SIP
dialog '76d7de407cbc38ee5e786aec05e1349a at 10.10.100.20:5060' in 6400 ms
(Method: INVITE)
[Mar 7 12:45:49] VERBOSE[3770] chan_sip.c: Reliably Transmitting (no NAT)
to 10.10.100.24:5060:
CANCEL sip:404 at 10.10.100.24:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.20:5060;branch=z9hG4bK644fb7f0
Max-Forwards: 70
From: "401" <sip:401 at 10.10.100.20>;tag=as309652b0
To: <sip:404 at 10.10.100.24:5060>
Call-ID: 76d7de407cbc38ee5e786aec05e1349a at 10.10.100.20:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.3
Content-Length: 0
---
[Mar 7 12:45:49] VERBOSE[3770] chan_sip.c: Scheduling destruction of SIP
dialog '76d7de407cbc38ee5e786aec05e1349a at 10.10.100.20:5060' in 6400 ms
(Method: INVITE)
[Mar 7 20:38:54] VERBOSE[3764] asterisk.c: -- Remote UNIX connection
disconnected
======================================================================
----------------------------------------------------------------------
(0132869) fcintron (reporter) - 2011-03-11 14:34
https://issues.asterisk.org/view.php?id=18955#c132869
----------------------------------------------------------------------
I just installed asterisk 1.6.2.17 with exactly same configuration to test
this issue and It worked without any problem.
Issue History
Date Modified Username Field Change
======================================================================
2011-03-11 14:34 fcintron Note Added: 0132869
======================================================================
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