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Summary:ASTERISK-17539: after picking up a call, phones stop working...
Reporter:Francisco Javier Cintrón Olguín (fcintron)Labels:
Date Opened:2011-03-10 14:01:11.000-0600Date Closed:2011-04-09 04:52:29
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:1.8.3 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) extensions.conf
( 1) features.conf
( 2) full
( 3) sip.conf
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@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)
[Mar  7 20:38:54] VERBOSE[3764] asterisk.c:     -- Remote UNIX connection disconnected

Comments:By: Francisco Javier Cintrón Olguín (fcintron) 2011-03-10 14:09:36.000-0600

Sorry for write conf files and /var/log/asterisk/full into description section. I didn´t see any attachment options when I was writing this issue. This is my first one.


By: Francisco Javier Cintrón Olguín (fcintron) 2011-03-11 14:34:38.000-0600

I just installed asterisk 1.6.2.17 with exactly same configuration to test this issue and It worked without any problem.
By: Francesco Romano (francesco_r) 2011-03-11 15:29:39.000-0600

The same bug was already reported on issue ASTERISK-1840654
By: Alec Davis (alecdavis) 2011-04-09 04:52:29

fcintron: possible patch on ASTERISK-17264 or if you prefer a workaround is to disable pickupsounds in features.conf