Problem picking up a call in asterisk

Asterisk Asterisk Support
1

Hi there. I am trying to configure an asterisk server for my office but I have a very nasty problem picking up a call.

I am using:

Centos 5.5 32 bits
Asterisk 1.8.3
Dahdi 2.4.1
Libpri 1.4.11.5
Libss7 1.0.2

Linux pbx.local 2.6.18-194.32.1.el5 #1 SMP Wed Jan 5 17:53:09 EST 2011 i686 i686 i386 GNU/Linux
I have 2 linksys ip phones spa921 and 1 normal phone connected to a cisco spa8800, all them are internal lines with these extensions:

ext 401 spa921
ext 402 spa921
ext 403 normal phone connected to spa8800

Testing “pick up call” functionality I came across this problem, these are steps to repeat it:

1.- call from 401 to 403
2.- with 403 doesn’t answer it.
3.- with 402 press *8# to pick up this call.
4.- 402 says that it is connected.
5.- 403 stops to sound.
6.- 401 keeps ringing, 402 said it is connected but 401 keeps ringing.
7.- Hang up 402
8.- Hang up 401

After these steps I can not neither send nor receive calls from anyone of 401, 402 or 403 until I restart asterisk.

/var/log/asterisk/messages, doesn´t show anything strange.

If a try to look a specific SIP channel using: sip show channel
asterisk cli hangs up.

I have tried this:

1.- Install last firmware for SPA921
2.- Install last firmware for SAP8800
3.- Install latest versions of asterisk and dahdi.
4.- Install all update for Centos
5.- Change pickup extension from *8 to *7
6.- Change this normal phone.

Thank you for your kind help.

2

sip history and or sip debug output needed.

3

Here is sip set debug on output from /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@10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.21:5060;branch=z9hG4bK-d9161a7a
From: "401" <sip:401@10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404@10.10.100.20>
Call-ID: f9e2055-43844cde@10.10.100.21
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "401" <sip:401@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@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@10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404@10.10.100.20>;tag=as077d9460
Call-ID: f9e2055-43844cde@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@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@10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.21:5060;branch=z9hG4bK-d9161a7a
From: "401" <sip:401@10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404@10.10.100.20>;tag=as077d9460
Call-ID: f9e2055-43844cde@10.10.100.21
CSeq: 101 ACK
Max-Forwards: 70
Contact: "401" <sip:401@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@10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.21:5060;branch=z9hG4bK-d9654c1b
From: "401" <sip:401@10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404@10.10.100.20>
Call-ID: f9e2055-43844cde@10.10.100.21
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="401",realm="asterisk",nonce="0fccf840",uri="sip:404@10.10.100.20",algorithm=MD5,response="9bcdad8f85a785f01798e76d3b6be2bb"
Contact: "401" <sip:401@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@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@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@10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404@10.10.100.20>
Call-ID: f9e2055-43844cde@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@10.10.100.20:5060>
Content-Length: 0


<------------>
[Mar  7 12:45:40] VERBOSE[3770] pbx.c:     -- Executing [404@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@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@10.10.100.20>;tag=as309652b0
To: <sip:404@10.10.100.24:5060>
Contact: <sip:401@10.10.100.20:5060>
Call-ID: 76d7de407cbc38ee5e786aec05e1349a@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@10.10.100.24:5060>
From: "401" <sip:401@10.10.100.20>;tag=as309652b0
Call-ID: 76d7de407cbc38ee5e786aec05e1349a@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@10.10.100.24:5060>;tag=57d02b562c2d5498i0
From: "401" <sip:401@10.10.100.20>;tag=as309652b0
Call-ID: 76d7de407cbc38ee5e786aec05e1349a@10.10.100.20:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.10.100.20:5060;branch=z9hG4bK644fb7f0
Contact: "404" <sip:404@10.10.100.24:5060>
Server: Cisco/SPA8800-6.1.7(GW)
Remote-Party-ID: "404" <sip:404@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@10.10.100.20>;tag=25c861e9c42ad85ao0
To: <sip:404@10.10.100.20>;tag=as393690ba
Call-ID: f9e2055-43844cde@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@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@10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.22:5060;branch=z9hG4bK-befd924d
From: "402" <sip:402@10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7@10.10.100.20>
Call-ID: 2a893106-30b58fd1@10.10.100.22
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "402" <sip:402@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@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@10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7@10.10.100.20>;tag=as21b41cb0
Call-ID: 2a893106-30b58fd1@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@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@10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.22:5060;branch=z9hG4bK-befd924d
From: "402" <sip:402@10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7@10.10.100.20>;tag=as21b41cb0
Call-ID: 2a893106-30b58fd1@10.10.100.22
CSeq: 101 ACK
Max-Forwards: 70
Contact: "402" <sip:402@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@10.10.100.20 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.22:5060;branch=z9hG4bK-83cecf31
From: "402" <sip:402@10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7@10.10.100.20>
Call-ID: 2a893106-30b58fd1@10.10.100.22
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="402",realm="asterisk",nonce="7c20bd27",uri="sip:*7@10.10.100.20",algorithm=MD5,response="0666e728d9b46007267a11b56bd32960"
Contact: "402" <sip:402@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@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@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@10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7@10.10.100.20>
Call-ID: 2a893106-30b58fd1@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@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@10.10.100.20>;tag=6b56478224790b85o0
To: <sip:*7@10.10.100.20>;tag=as778b6ec7
Call-ID: 2a893106-30b58fd1@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@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@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@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@10.10.100.20>;tag=as309652b0
To: <sip:404@10.10.100.24:5060>
Call-ID: 76d7de407cbc38ee5e786aec05e1349a@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@10.10.100.20:5060' in 6400 ms (Method: INVITE)
4

This is a bug, I just finished to install asterisk 1.6 and it works perfectly with same configuration. How could I report this bug?

5

I haven’t reviewed the logs, but assuming they demonstrate a problem, submit the same information to issues.aterisk.org/. You would normally attach the logs. Strictly speaking, for SIP problems, you should provide sip history as well as the debug log.

Refer back to the forum in the additional information.

Ideally, search it for duplicates, first.