[asterisk-bugs] [JIRA] (ASTERISK-23930) Call Barging/Whispering issue on SIP
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Fri Jun 27 06:59:03 CDT 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-23930?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-23930:
------------------------------------
Description:
Hi,
Call Barging and Whispering is not happening over SIP. Instead call listening is happening when I do barge(B) or whisper(w). FYI, Everything is working when I use DAHDI(PRI lines).
I am struck on this for the last two days. I am using latest asterisk 1.8.28. When I do Barge with the option 'Bqs', Barging is not happening. I am able to listen the existing conversation. But I am not able to participate. The same issue with whisper as well.
What might be the wrong. I am attaching the sip debug the log for reference. Please help me to find out the issue.
Please find the sip log below.
[Edit by Rusty - Removed reporters inline debug - reporter attached to issue]
was:
Hi,
Call Barging and Whispering is not happening over SIP. Instead call listening is happening when I do barge(B) or whisper(w). FYI, Everything is working when I use DAHDI(PRI lines).
I am struck on this for the last two days. I am using latest asterisk 1.8.28. When I do Barge with the option 'Bqs', Barging is not happening. I am able to listen the existing conversation. But I am not able to participate. The same issue with whisper as well.
What might be the wrong. I am attaching the sip debug the log for reference. Please help me to find out the issue.
Please find the sip log below.
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5057 --->
INVITE sip:08050103381 at obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5057;rport;branch=z9hG4bKafcxhmrg
Max-Forwards: 70
To: <sip:08050103381 at obelix.exotel.in>
From: "GovindTest -" <sip:Exotel08050103381 at obelix.exotel.in>;tag=zgnnp
Call-ID: rrcbxaikhlgqvdn at govind
CSeq: 12 INVITE
Contact: <sip:Exotel08050103381 at 192.168.33.1:5057>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.4.2
Content-Length: 307
v=0
o=twinkle 593607979 276015228 IN IP4 192.168.33.1
s=-
c=IN IP4 192.168.33.1
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: --- (13 headers 14 lines) ---
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5057 (NAT)
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Using INVITE request as basis request - rrcbxaikhlgqvdn at govind
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found peer 'Exotel08050103381' for 'Exotel08050103381' from 192.168.33.1:5057
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.33.1:5057 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.33.1:5057;branch=z9hG4bKafcxhmrg;received=192.168.33.1;rport=5057
From: "GovindTest -" <sip:Exotel08050103381 at obelix.exotel.in>;tag=zgnnp
To: <sip:08050103381 at obelix.exotel.in>;tag=as381c5d68
Call-ID: rrcbxaikhlgqvdn at govind
CSeq: 12 INVITE
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61fe0e0c"
Content-Length: 0
<------------>
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'rrcbxaikhlgqvdn at govind' in 32000 ms (Method: INVITE)
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5057 --->
ACK sip:08050103381 at obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5057;rport;branch=z9hG4bKafcxhmrg
Max-Forwards: 70
To: <sip:08050103381 at obelix.exotel.in>;tag=as381c5d68
From: "GovindTest -" <sip:Exotel08050103381 at obelix.exotel.in>;tag=zgnnp
Call-ID: rrcbxaikhlgqvdn at govind
CSeq: 12 ACK
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: --- (9 headers 0 lines) ---
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5057 --->
INVITE sip:08050103381 at obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5057;rport;branch=z9hG4bKhwexthks
Max-Forwards: 70
To: <sip:08050103381 at obelix.exotel.in>
From: "GovindTest -" <sip:Exotel08050103381 at obelix.exotel.in>;tag=zgnnp
Call-ID: rrcbxaikhlgqvdn at govind
CSeq: 13 INVITE
Contact: <sip:Exotel08050103381 at 192.168.33.1:5057>
Content-Type: application/sdp
Authorization: Digest username="Exotel08050103381",realm="asterisk",nonce="61fe0e0c",uri="sip:08050103381 at obelix.exotel.in",response="e61beb87c58a5052e4ca1d44abc88ba4",algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.4.2
Content-Length: 307
v=0
o=twinkle 593607979 276015228 IN IP4 192.168.33.1
s=-
c=IN IP4 192.168.33.1
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: --- (14 headers 14 lines) ---
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5057 (NAT)
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Using INVITE request as basis request - rrcbxaikhlgqvdn at govind
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found peer 'Exotel08050103381' for 'Exotel08050103381' from 192.168.33.1:5057
[Jun 24 16:54:59] VERBOSE[5776] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found RTP audio format 98
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found RTP audio format 97
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found RTP audio format 8
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found RTP audio format 0
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found RTP audio format 3
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found RTP audio format 101
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found audio description format speex for ID 98
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found audio description format speex for ID 97
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found audio description format PCMA for ID 8
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found audio description format PCMU for ID 0
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found audio description format GSM for ID 3
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Found audio description format telephone-event for ID 101
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x20000020e (gsm|ulaw|alaw|speex|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Peer audio RTP is at port 192.168.33.1:8000
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: Looking for 08050103381 in exotelsip-incoming (domain obelix.exotel.in)
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c: list_route: hop: <sip:Exotel08050103381 at 192.168.33.1:5057>
[Jun 24 16:54:59] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5057 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.33.1:5057;branch=z9hG4bKhwexthks;received=192.168.33.1;rport=5057
From: "GovindTest -" <sip:Exotel08050103381 at obelix.exotel.in>;tag=zgnnp
To: <sip:08050103381 at obelix.exotel.in>
Call-ID: rrcbxaikhlgqvdn at govind
CSeq: 13 INVITE
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:08050103381 at 192.168.33.22:5060>
Content-Length: 0
<------------>
[Jun 24 16:54:59] VERBOSE[5965] pbx.c: -- Executing [08050103381 at exotelsip-incoming:1] Answer("SIP/Exotel08050103381-00000000", "") in new stack
[Jun 24 16:54:59] VERBOSE[5965] chan_sip.c: Audio is at 13752
[Jun 24 16:54:59] VERBOSE[5965] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[Jun 24 16:54:59] VERBOSE[5965] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Jun 24 16:54:59] VERBOSE[5965] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Jun 24 16:54:59] VERBOSE[5965] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jun 24 16:54:59] VERBOSE[5965] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.33.1:5057 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.1:5057;branch=z9hG4bKhwexthks;received=192.168.33.1;rport=5057
From: "GovindTest -" <sip:Exotel08050103381 at obelix.exotel.in>;tag=zgnnp
To: <sip:08050103381 at obelix.exotel.in>;tag=as11f87695
Call-ID: rrcbxaikhlgqvdn at govind
CSeq: 13 INVITE
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:08050103381 at 192.168.33.22:5060>
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 1816005428 1816005428 IN IP4 192.168.33.22
s=Asterisk PBX 1.8.28.2
c=IN IP4 192.168.33.22
t=0 0
m=audio 13752 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
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
<------------>
[Jun 24 16:55:00] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5057 --->
ACK sip:08050103381 at 192.168.33.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5057;rport;branch=z9hG4bKhglhongv
Max-Forwards: 70
To: <sip:08050103381 at obelix.exotel.in>;tag=as11f87695
From: "GovindTest -" <sip:Exotel08050103381 at obelix.exotel.in>;tag=zgnnp
Call-ID: rrcbxaikhlgqvdn at govind
CSeq: 13 ACK
Authorization: Digest username="Exotel08050103381",realm="asterisk",nonce="61fe0e0c",uri="sip:08050103381 at obelix.exotel.in",response="e61beb87c58a5052e4ca1d44abc88ba4",algorithm=MD5
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:55:00] VERBOSE[5776] chan_sip.c: --- (10 headers 0 lines) ---
[Jun 24 16:55:00] VERBOSE[5965] pbx.c: -- Executing [08050103381 at exotelsip-incoming:2] Dial("SIP/Exotel08050103381-00000000", "SIP/08050103381") in new stack
[Jun 24 16:55:00] VERBOSE[5965] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 24 16:55:00] VERBOSE[5965] chan_sip.c: Audio is at 10990
[Jun 24 16:55:00] VERBOSE[5965] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[Jun 24 16:55:00] VERBOSE[5965] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Jun 24 16:55:00] VERBOSE[5965] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Jun 24 16:55:00] VERBOSE[5965] chan_sip.c: Adding codec 0x800000000000 (testlaw) to SDP
[Jun 24 16:55:00] VERBOSE[5965] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jun 24 16:55:00] VERBOSE[5965] chan_sip.c: Reliably Transmitting (NAT) to 192.168.33.1:5051:
INVITE sip:08050103381 at 192.168.33.1:5051 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.22:5060;branch=z9hG4bK6437e232;rport
Max-Forwards: 70
From: "GovindTest -" <sip:Exotel08050103381 at 192.168.33.22>;tag=as19db0af8
To: <sip:08050103381 at 192.168.33.1:5051>
Contact: <sip:Exotel08050103381 at 192.168.33.22:5060>
Call-ID: 3c17b2c6757564c57820f3ad684a525d at 192.168.33.22:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.28.2
Date: Tue, 24 Jun 2014 11:25:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 1212243861 1212243861 IN IP4 192.168.33.22
s=Asterisk PBX 1.8.28.2
c=IN IP4 192.168.33.22
t=0 0
m=audio 10990 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
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
---
[Jun 24 16:55:00] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5051 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.33.22:5060;received=192.168.33.22;rport=5060;branch=z9hG4bK6437e232
To: <sip:08050103381 at 192.168.33.1:5051>
From: "GovindTest -" <sip:Exotel08050103381 at 192.168.33.22>;tag=as19db0af8
Call-ID: 3c17b2c6757564c57820f3ad684a525d at 192.168.33.22:5060
CSeq: 102 INVITE
Server: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:55:00] VERBOSE[5776] chan_sip.c: --- (8 headers 0 lines) ---
[Jun 24 16:55:00] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5051 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.33.22:5060;received=192.168.33.22;rport=5060;branch=z9hG4bK6437e232
To: <sip:08050103381 at 192.168.33.1:5051>;tag=aplqe
From: "GovindTest -" <sip:Exotel08050103381 at 192.168.33.22>;tag=as19db0af8
Call-ID: 3c17b2c6757564c57820f3ad684a525d at 192.168.33.22:5060
CSeq: 102 INVITE
Contact: <sip:08050103381 at 192.168.33.1:5051>
Server: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:55:00] VERBOSE[5776] chan_sip.c: --- (9 headers 0 lines) ---
[Jun 24 16:55:00] VERBOSE[5776] chan_sip.c: list_route: hop: <sip:08050103381 at 192.168.33.1:5051>
[Jun 24 16:55:00] VERBOSE[5965] app_dial.c: -- Called SIP/08050103381
[Jun 24 16:55:00] VERBOSE[5965] app_dial.c: -- SIP/08050103381-00000001 is ringing
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5051 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.22:5060;received=192.168.33.22;rport=5060;branch=z9hG4bK6437e232
To: <sip:08050103381 at 192.168.33.1:5051>;tag=aplqe
From: "GovindTest -" <sip:Exotel08050103381 at 192.168.33.22>;tag=as19db0af8
Call-ID: 3c17b2c6757564c57820f3ad684a525d at 192.168.33.22:5060
CSeq: 102 INVITE
Contact: <sip:08050103381 at 192.168.33.1:5051>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.4.2
Supported: replaces,norefersub
Content-Length: 193
v=0
o=twinkle 825374987 1830279770 IN IP4 192.168.33.1
s=-
c=IN IP4 192.168.33.1
t=0 0
m=audio 7984 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: --- (12 headers 9 lines) ---
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: Found RTP audio format 3
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: Found RTP audio format 101
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: Found audio description format GSM for ID 3
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: Found audio description format telephone-event for ID 101
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: Peer audio RTP is at port 192.168.33.1:7984
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: list_route: hop: <sip:08050103381 at 192.168.33.1:5051>
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: set_destination: Parsing <sip:08050103381 at 192.168.33.1:5051> for address/port to send to
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: set_destination: set destination to 192.168.33.1:5051
[Jun 24 16:55:02] VERBOSE[5776] chan_sip.c: Transmitting (NAT) to 192.168.33.1:5051:
ACK sip:08050103381 at 192.168.33.1:5051 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.22:5060;branch=z9hG4bK320e940f;rport
Max-Forwards: 70
From: "GovindTest -" <sip:Exotel08050103381 at 192.168.33.22>;tag=as19db0af8
To: <sip:08050103381 at 192.168.33.1:5051>;tag=aplqe
Contact: <sip:Exotel08050103381 at 192.168.33.22:5060>
Call-ID: 3c17b2c6757564c57820f3ad684a525d at 192.168.33.22:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.28.2
Content-Length: 0
---
[Jun 24 16:55:02] VERBOSE[5965] app_dial.c: -- SIP/08050103381-00000001 answered SIP/Exotel08050103381-00000000
[Jun 24 16:55:02] VERBOSE[5965] rtp_engine.c: -- Locally bridging SIP/Exotel08050103381-00000000 and SIP/08050103381-00000001
[Jun 24 16:55:13] VERBOSE[5966] chan_sip.c: Really destroying SIP dialog '25ad08877adfd98134ed7a516123a931 at 127.0.1.1:5060' Method: INVITE
[Jun 24 16:55:13] NOTICE[5966] channel.c: Unable to request channel SIP/1000
[Jun 24 16:55:22] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
REGISTER sip:obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKjutvzswr
Max-Forwards: 70
To: "1000" <sip:1000 at obelix.exotel.in>
From: "1000" <sip:1000 at obelix.exotel.in>;tag=gczhh
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 840 REGISTER
Contact: <sip:1000 at 192.168.33.1:5054>;expires=3600
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:55:22] VERBOSE[5776] chan_sip.c: --- (11 headers 0 lines) ---
[Jun 24 16:55:22] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:55:22] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKjutvzswr;received=192.168.33.1;rport=5054
From: "1000" <sip:1000 at obelix.exotel.in>;tag=gczhh
To: "1000" <sip:1000 at obelix.exotel.in>;tag=as38e2a2cd
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 840 REGISTER
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f6d4e10"
Content-Length: 0
<------------>
[Jun 24 16:55:22] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'ijdxntqltqpdmjw at govind' in 32000 ms (Method: REGISTER)
[Jun 24 16:55:22] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
REGISTER sip:obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKwiwrqpfr
Max-Forwards: 70
To: "1000" <sip:1000 at obelix.exotel.in>
From: "1000" <sip:1000 at obelix.exotel.in>;tag=gczhh
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 841 REGISTER
Contact: <sip:1000 at 192.168.33.1:5054>;expires=3600
Authorization: Digest username="1000",realm="asterisk",nonce="2f6d4e10",uri="sip:obelix.exotel.in",response="3478e770cb3760e3dc627131b3f4e50e",algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:55:22] VERBOSE[5776] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 24 16:55:22] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:55:22] VERBOSE[5776] chan_sip.c: -- Registered SIP '1000' at 192.168.33.1:5054
[Jun 24 16:55:22] VERBOSE[5776] chan_sip.c: > Saved useragent "Twinkle/1.4.2" for peer 1000
[Jun 24 16:55:23] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKwiwrqpfr;received=192.168.33.1;rport=5054
From: "1000" <sip:1000 at obelix.exotel.in>;tag=gczhh
To: "1000" <sip:1000 at obelix.exotel.in>;tag=as38e2a2cd
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 841 REGISTER
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:1000 at 192.168.33.1:5054>;expires=3600
Date: Tue, 24 Jun 2014 11:25:23 GMT
Content-Length: 0
<------------>
[Jun 24 16:55:23] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'ijdxntqltqpdmjw at govind' in 32000 ms (Method: REGISTER)
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
REGISTER sip:obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKiaqnvnka
Max-Forwards: 70
To: "1000" <sip:1000 at obelix.exotel.in>
From: "1000" <sip:1000 at obelix.exotel.in>;tag=nohie
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 842 REGISTER
Contact: <sip:1000 at 192.168.33.1:5054>;expires=3600
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c: --- (11 headers 0 lines) ---
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKiaqnvnka;received=192.168.33.1;rport=5054
From: "1000" <sip:1000 at obelix.exotel.in>;tag=nohie
To: "1000" <sip:1000 at obelix.exotel.in>;tag=as1f0fb381
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 842 REGISTER
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="651d657e"
Content-Length: 0
<------------>
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'ijdxntqltqpdmjw at govind' in 32000 ms (Method: REGISTER)
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
REGISTER sip:obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKankufzbx
Max-Forwards: 70
To: "1000" <sip:1000 at obelix.exotel.in>
From: "1000" <sip:1000 at obelix.exotel.in>;tag=nohie
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 843 REGISTER
Contact: <sip:1000 at 192.168.33.1:5054>;expires=3600
Authorization: Digest username="1000",realm="asterisk",nonce="651d657e",uri="sip:obelix.exotel.in",response="0b333878b9a15e5e459f68e2e3deb194",algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:55:51] NOTICE[5776] chan_sip.c: Correct auth, but based on stale nonce received from '"1000" <sip:1000 at obelix.exotel.in>;tag=nohie'
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKankufzbx;received=192.168.33.1;rport=5054
From: "1000" <sip:1000 at obelix.exotel.in>;tag=nohie
To: "1000" <sip:1000 at obelix.exotel.in>;tag=as38e2a2cd
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 843 REGISTER
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2c469ba7", stale=true
Content-Length: 0
<------------>
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'ijdxntqltqpdmjw at govind' in 32000 ms (Method: REGISTER)
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
REGISTER sip:obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKdqxvefwc
Max-Forwards: 70
To: "1000" <sip:1000 at obelix.exotel.in>
From: "1000" <sip:1000 at obelix.exotel.in>;tag=nohie
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 844 REGISTER
Contact: <sip:1000 at 192.168.33.1:5054>;expires=3600
Authorization: Digest username="1000",realm="asterisk",nonce="2c469ba7",uri="sip:obelix.exotel.in",response="37ccfccf2b9b3445979f1aa6f6beadb6",algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKdqxvefwc;received=192.168.33.1;rport=5054
From: "1000" <sip:1000 at obelix.exotel.in>;tag=nohie
To: "1000" <sip:1000 at obelix.exotel.in>;tag=as38e2a2cd
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 844 REGISTER
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:1000 at 192.168.33.1:5054>;expires=3600
Date: Tue, 24 Jun 2014 11:25:51 GMT
Content-Length: 0
<------------>
[Jun 24 16:55:51] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'ijdxntqltqpdmjw at govind' in 32000 ms (Method: REGISTER)
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
REGISTER sip:obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKiwaebbia
Max-Forwards: 70
To: "1000" <sip:1000 at obelix.exotel.in>
From: "1000" <sip:1000 at obelix.exotel.in>;tag=bcggq
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 845 REGISTER
Contact: <sip:1000 at 192.168.33.1:5054>;expires=0
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c: --- (11 headers 0 lines) ---
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKiwaebbia;received=192.168.33.1;rport=5054
From: "1000" <sip:1000 at obelix.exotel.in>;tag=bcggq
To: "1000" <sip:1000 at obelix.exotel.in>;tag=as68243801
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 845 REGISTER
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2614b6fc"
Content-Length: 0
<------------>
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'ijdxntqltqpdmjw at govind' in 32000 ms (Method: REGISTER)
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
REGISTER sip:obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKsrhnxmjb
Max-Forwards: 70
To: "1000" <sip:1000 at obelix.exotel.in>
From: "1000" <sip:1000 at obelix.exotel.in>;tag=bcggq
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 846 REGISTER
Contact: <sip:1000 at 192.168.33.1:5054>;expires=0
Authorization: Digest username="1000",realm="asterisk",nonce="2614b6fc",uri="sip:obelix.exotel.in",response="346805c58f252a34b51eb44c8b18b199",algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:56:00] NOTICE[5776] chan_sip.c: Correct auth, but based on stale nonce received from '"1000" <sip:1000 at obelix.exotel.in>;tag=bcggq'
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKsrhnxmjb;received=192.168.33.1;rport=5054
From: "1000" <sip:1000 at obelix.exotel.in>;tag=bcggq
To: "1000" <sip:1000 at obelix.exotel.in>;tag=as38e2a2cd
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 846 REGISTER
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="22f93f3e", stale=true
Content-Length: 0
<------------>
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'ijdxntqltqpdmjw at govind' in 32000 ms (Method: REGISTER)
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
REGISTER sip:obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKhgwzieni
Max-Forwards: 70
To: "1000" <sip:1000 at obelix.exotel.in>
From: "1000" <sip:1000 at obelix.exotel.in>;tag=bcggq
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 847 REGISTER
Contact: <sip:1000 at 192.168.33.1:5054>;expires=0
Authorization: Digest username="1000",realm="asterisk",nonce="22f93f3e",uri="sip:obelix.exotel.in",response="88d6ae65e43799d827493a45d2b303eb",algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c: -- Unregistered SIP '1000'
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKhgwzieni;received=192.168.33.1;rport=5054
From: "1000" <sip:1000 at obelix.exotel.in>;tag=bcggq
To: "1000" <sip:1000 at obelix.exotel.in>;tag=as38e2a2cd
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 847 REGISTER
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 0
Date: Tue, 24 Jun 2014 11:26:00 GMT
Content-Length: 0
<------------>
[Jun 24 16:56:00] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'ijdxntqltqpdmjw at govind' in 32000 ms (Method: REGISTER)
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
REGISTER sip:obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKpysexepz
Max-Forwards: 70
To: "1000" <sip:1000 at obelix.exotel.in>
From: "1000" <sip:1000 at obelix.exotel.in>;tag=dbxgj
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 848 REGISTER
Contact: <sip:1000 at 192.168.33.1:5054>;expires=3600
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c: --- (11 headers 0 lines) ---
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKpysexepz;received=192.168.33.1;rport=5054
From: "1000" <sip:1000 at obelix.exotel.in>;tag=dbxgj
To: "1000" <sip:1000 at obelix.exotel.in>;tag=as5d297ff6
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 848 REGISTER
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c422174"
Content-Length: 0
<------------>
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'ijdxntqltqpdmjw at govind' in 32000 ms (Method: REGISTER)
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
REGISTER sip:obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKalfzukne
Max-Forwards: 70
To: "1000" <sip:1000 at obelix.exotel.in>
From: "1000" <sip:1000 at obelix.exotel.in>;tag=dbxgj
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 849 REGISTER
Contact: <sip:1000 at 192.168.33.1:5054>;expires=3600
Authorization: Digest username="1000",realm="asterisk",nonce="5c422174",uri="sip:obelix.exotel.in",response="f7d3d4e31bd100dd5228048aba73e0e9",algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:56:05] NOTICE[5776] chan_sip.c: Correct auth, but based on stale nonce received from '"1000" <sip:1000 at obelix.exotel.in>;tag=dbxgj'
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKalfzukne;received=192.168.33.1;rport=5054
From: "1000" <sip:1000 at obelix.exotel.in>;tag=dbxgj
To: "1000" <sip:1000 at obelix.exotel.in>;tag=as38e2a2cd
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 849 REGISTER
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0f0458c8", stale=true
Content-Length: 0
<------------>
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'ijdxntqltqpdmjw at govind' in 32000 ms (Method: REGISTER)
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
REGISTER sip:obelix.exotel.in SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKxlstqdja
Max-Forwards: 70
To: "1000" <sip:1000 at obelix.exotel.in>
From: "1000" <sip:1000 at obelix.exotel.in>;tag=dbxgj
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 850 REGISTER
Contact: <sip:1000 at 192.168.33.1:5054>;expires=3600
Authorization: Digest username="1000",realm="asterisk",nonce="0f0458c8",uri="sip:obelix.exotel.in",response="4d330b735585f1a707585f23e81793d2",algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c: -- Registered SIP '1000' at 192.168.33.1:5054
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKxlstqdja;received=192.168.33.1;rport=5054
From: "1000" <sip:1000 at obelix.exotel.in>;tag=dbxgj
To: "1000" <sip:1000 at obelix.exotel.in>;tag=as38e2a2cd
Call-ID: ijdxntqltqpdmjw at govind
CSeq: 850 REGISTER
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:1000 at 192.168.33.1:5054>;expires=3600
Date: Tue, 24 Jun 2014 11:26:05 GMT
Content-Length: 0
<------------>
[Jun 24 16:56:05] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog 'ijdxntqltqpdmjw at govind' in 32000 ms (Method: REGISTER)
[Jun 24 16:56:14] VERBOSE[5980] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 24 16:56:14] VERBOSE[5980] chan_sip.c: Audio is at 13424
[Jun 24 16:56:14] VERBOSE[5980] chan_sip.c: Adding codec 0x400 (ilbc) to SDP
[Jun 24 16:56:14] VERBOSE[5980] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[Jun 24 16:56:14] VERBOSE[5980] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Jun 24 16:56:14] VERBOSE[5980] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Jun 24 16:56:14] VERBOSE[5980] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jun 24 16:56:14] VERBOSE[5980] chan_sip.c: Reliably Transmitting (NAT) to 192.168.33.1:5054:
INVITE sip:1000 at 192.168.33.1:5054 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.22:5060;branch=z9hG4bK2732db5a;rport
Max-Forwards: 70
From: "MonAst Spyer" <sip:asterisk at 192.168.33.22>;tag=as51ba9a12
To: <sip:1000 at 192.168.33.1:5054>
Contact: <sip:asterisk at 192.168.33.22:5060>
Call-ID: 4f666e253fa7e77c5dc5971f52c75ee6 at 192.168.33.22:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.28.2
Date: Tue, 24 Jun 2014 11:26:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 329
v=0
o=root 312746186 312746186 IN IP4 192.168.33.22
s=Asterisk PBX 1.8.28.2
c=IN IP4 192.168.33.22
t=0 0
m=audio 13424 RTP/AVP 97 3 0 8 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:3 GSM/8000
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
---
[Jun 24 16:56:14] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.33.22:5060;received=192.168.33.22;rport=5060;branch=z9hG4bK2732db5a
To: <sip:1000 at 192.168.33.1:5054>
From: "MonAst Spyer" <sip:asterisk at 192.168.33.22>;tag=as51ba9a12
Call-ID: 4f666e253fa7e77c5dc5971f52c75ee6 at 192.168.33.22:5060
CSeq: 102 INVITE
Server: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:56:14] VERBOSE[5776] chan_sip.c: --- (8 headers 0 lines) ---
[Jun 24 16:56:14] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.33.22:5060;received=192.168.33.22;rport=5060;branch=z9hG4bK2732db5a
To: <sip:1000 at 192.168.33.1:5054>;tag=rkdsw
From: "MonAst Spyer" <sip:asterisk at 192.168.33.22>;tag=as51ba9a12
Call-ID: 4f666e253fa7e77c5dc5971f52c75ee6 at 192.168.33.22:5060
CSeq: 102 INVITE
Contact: <sip:1000 at 192.168.33.1:5054>
Server: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:56:14] VERBOSE[5776] chan_sip.c: --- (9 headers 0 lines) ---
[Jun 24 16:56:14] VERBOSE[5776] chan_sip.c: list_route: hop: <sip:1000 at 192.168.33.1:5054>
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.22:5060;received=192.168.33.22;rport=5060;branch=z9hG4bK2732db5a
To: <sip:1000 at 192.168.33.1:5054>;tag=rkdsw
From: "MonAst Spyer" <sip:asterisk at 192.168.33.22>;tag=as51ba9a12
Call-ID: 4f666e253fa7e77c5dc5971f52c75ee6 at 192.168.33.22:5060
CSeq: 102 INVITE
Contact: <sip:1000 at 192.168.33.1:5054>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.4.2
Supported: replaces,norefersub
Content-Length: 193
v=0
o=twinkle 396265683 2051223881 IN IP4 192.168.33.1
s=-
c=IN IP4 192.168.33.1
t=0 0
m=audio 8000 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: --- (12 headers 9 lines) ---
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: Found RTP audio format 3
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: Found RTP audio format 101
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: Found audio description format GSM for ID 3
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: Found audio description format telephone-event for ID 101
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: Peer audio RTP is at port 192.168.33.1:8000
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: list_route: hop: <sip:1000 at 192.168.33.1:5054>
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: set_destination: Parsing <sip:1000 at 192.168.33.1:5054> for address/port to send to
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: set_destination: set destination to 192.168.33.1:5054
[Jun 24 16:56:17] VERBOSE[5776] chan_sip.c: Transmitting (NAT) to 192.168.33.1:5054:
ACK sip:1000 at 192.168.33.1:5054 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.22:5060;branch=z9hG4bK2d7b0f3c;rport
Max-Forwards: 70
From: "MonAst Spyer" <sip:asterisk at 192.168.33.22>;tag=as51ba9a12
To: <sip:1000 at 192.168.33.1:5054>;tag=rkdsw
Contact: <sip:asterisk at 192.168.33.22:5060>
Call-ID: 4f666e253fa7e77c5dc5971f52c75ee6 at 192.168.33.22:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.28.2
Content-Length: 0
---
[Jun 24 16:56:17] VERBOSE[5980] pbx.c: > Channel SIP/1000-00000002 was answered.
[Jun 24 16:56:17] VERBOSE[5984] pbx.c: > Launching ChanSpy(SIP/Exotel08050103381-00000000,qBsS) on SIP/1000-00000002
[Jun 24 16:56:17] VERBOSE[5984] app_chanspy.c: == Spying on channel SIP/Exotel08050103381-00000000
[Jun 24 16:56:17] NOTICE[5984] app_chanspy.c: Attaching SIP/1000-00000002 to SIP/Exotel08050103381-00000000
[Jun 24 16:56:17] NOTICE[5984] app_chanspy.c: Attaching SIP/1000-00000002 to SIP/Exotel08050103381-00000000
[Jun 24 16:56:17] NOTICE[5984] app_chanspy.c: Attaching SIP/1000-00000002 to SIP/08050103381-00000001
[Jun 24 16:56:23] VERBOSE[5776] chan_sip.c: Really destroying SIP dialog 'ijdxntqltqpdmjw at govind' Method: REGISTER
[Jun 24 16:56:32] VERBOSE[5776] chan_sip.c: Really destroying SIP dialog 'ijdxntqltqpdmjw at govind' Method: REGISTER
[Jun 24 16:56:37] VERBOSE[5776] chan_sip.c: Really destroying SIP dialog 'ijdxntqltqpdmjw at govind' Method: REGISTER
[Jun 24 16:56:37] VERBOSE[5776] chan_sip.c: Really destroying SIP dialog 'ijdxntqltqpdmjw at govind' Method: REGISTER
[Jun 24 16:56:48] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5054 --->
BYE sip:asterisk at 192.168.33.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5054;rport;branch=z9hG4bKvswqmjtg
Max-Forwards: 70
To: "MonAst Spyer" <sip:asterisk at 192.168.33.22>;tag=as51ba9a12
From: <sip:1000 at 192.168.33.1:5054>;tag=rkdsw
Call-ID: 4f666e253fa7e77c5dc5971f52c75ee6 at 192.168.33.22:5060
CSeq: 350 BYE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:56:48] VERBOSE[5776] chan_sip.c: --- (9 headers 0 lines) ---
[Jun 24 16:56:48] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5054 (NAT)
[Jun 24 16:56:48] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog '4f666e253fa7e77c5dc5971f52c75ee6 at 192.168.33.22:5060' in 32000 ms (Method: BYE)
[Jun 24 16:56:48] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5054 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.1:5054;branch=z9hG4bKvswqmjtg;received=192.168.33.1;rport=5054
From: <sip:1000 at 192.168.33.1:5054>;tag=rkdsw
To: "MonAst Spyer" <sip:asterisk at 192.168.33.22>;tag=as51ba9a12
Call-ID: 4f666e253fa7e77c5dc5971f52c75ee6 at 192.168.33.22:5060
CSeq: 350 BYE
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
[Jun 24 16:56:52] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5051 --->
BYE sip:Exotel08050103381 at 192.168.33.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.1:5051;rport;branch=z9hG4bKmilzcsrh
Max-Forwards: 70
To: "GovindTest -" <sip:Exotel08050103381 at 192.168.33.22>;tag=as19db0af8
From: <sip:08050103381 at 192.168.33.1:5051>;tag=aplqe
Call-ID: 3c17b2c6757564c57820f3ad684a525d at 192.168.33.22:5060
CSeq: 758 BYE
User-Agent: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:56:52] VERBOSE[5776] chan_sip.c: --- (9 headers 0 lines) ---
[Jun 24 16:56:52] VERBOSE[5776] chan_sip.c: Sending to 192.168.33.1:5051 (NAT)
[Jun 24 16:56:52] VERBOSE[5776] chan_sip.c: Scheduling destruction of SIP dialog '3c17b2c6757564c57820f3ad684a525d at 192.168.33.22:5060' in 32000 ms (Method: BYE)
[Jun 24 16:56:52] VERBOSE[5776] chan_sip.c:
<--- Transmitting (NAT) to 192.168.33.1:5051 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.1:5051;branch=z9hG4bKmilzcsrh;received=192.168.33.1;rport=5051
From: <sip:08050103381 at 192.168.33.1:5051>;tag=aplqe
To: "GovindTest -" <sip:Exotel08050103381 at 192.168.33.22>;tag=as19db0af8
Call-ID: 3c17b2c6757564c57820f3ad684a525d at 192.168.33.22:5060
CSeq: 758 BYE
Server: Asterisk PBX 1.8.28.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
[Jun 24 16:56:52] VERBOSE[5965] pbx.c: -- Executing [h at exotelsip-incoming:1] Answer("SIP/Exotel08050103381-00000000", "") in new stack
[Jun 24 16:56:52] VERBOSE[5965] features.c: == Spawn extension (exotelsip-incoming, h, 1) exited non-zero on 'SIP/Exotel08050103381-00000000'
[Jun 24 16:56:52] VERBOSE[5965] pbx.c: == Spawn extension (exotelsip-incoming, 08050103381, 2) exited non-zero on 'SIP/Exotel08050103381-00000000'
[Jun 24 16:56:52] VERBOSE[5984] app_chanspy.c: == Done Spying on channel SIP/Exotel08050103381-00000000
[Jun 24 16:56:52] VERBOSE[5965] chan_sip.c: Scheduling destruction of SIP dialog 'rrcbxaikhlgqvdn at govind' in 32000 ms (Method: ACK)
[Jun 24 16:56:52] VERBOSE[5965] chan_sip.c: set_destination: Parsing <sip:Exotel08050103381 at 192.168.33.1:5057> for address/port to send to
[Jun 24 16:56:52] VERBOSE[5965] chan_sip.c: set_destination: set destination to 192.168.33.1:5057
[Jun 24 16:56:52] VERBOSE[5965] chan_sip.c: Reliably Transmitting (NAT) to 192.168.33.1:5057:
BYE sip:Exotel08050103381 at 192.168.33.1:5057 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.22:5060;branch=z9hG4bK5e4f7407;rport
Max-Forwards: 70
From: <sip:08050103381 at obelix.exotel.in>;tag=as11f87695
To: "GovindTest -" <sip:Exotel08050103381 at obelix.exotel.in>;tag=zgnnp
Call-ID: rrcbxaikhlgqvdn at govind
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.28.2
Proxy-Authorization: Digest username="Exotel08050103381", realm="asterisk", algorithm=MD5, uri="sip:obelix.exotel.in", nonce="", response="cfcecebed14162b6f10f8dd6349e3f79"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[Jun 24 16:56:52] VERBOSE[5776] chan_sip.c:
<--- SIP read from UDP:192.168.33.1:5057 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.22:5060;received=192.168.33.22;rport=5060;branch=z9hG4bK5e4f7407
To: "GovindTest -" <sip:Exotel08050103381 at obelix.exotel.in>;tag=zgnnp
From: <sip:08050103381 at obelix.exotel.in>;tag=as11f87695
Call-ID: rrcbxaikhlgqvdn at govind
CSeq: 102 BYE
Server: Twinkle/1.4.2
Content-Length: 0
<------------->
[Jun 24 16:56:52] VERBOSE[5776] chan_sip.c: --- (8 headers 0 lines) ---
[Jun 24 16:56:52] VERBOSE[5776] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[Jun 24 16:56:52] VERBOSE[5776] chan_sip.c: Really destroying SIP dialog 'rrcbxaikhlgqvdn at govind' Method: ACK
[Jun 24 16:57:20] VERBOSE[5776] chan_sip.c: Really destroying SIP dialog '4f666e253fa7e77c5dc5971f52c75ee6 at 192.168.33.22:5060' Method: BYE
> Call Barging/Whispering issue on SIP
> ------------------------------------
>
> Key: ASTERISK-23930
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-23930
> Project: Asterisk
> Issue Type: Information Request
> Security Level: None
> Components: Channels/chan_sip/CodecHandling
> Environment: Tried with Ubuntu 12.04 / Asterisk 1.8.28 and CentOS-6 Asterisk 1.8.18
> Reporter: Govind Rajan
> Assignee: Govind Rajan
> Severity: Minor
> Attachments: debug.log
>
>
> Hi,
> Call Barging and Whispering is not happening over SIP. Instead call listening is happening when I do barge(B) or whisper(w). FYI, Everything is working when I use DAHDI(PRI lines).
> I am struck on this for the last two days. I am using latest asterisk 1.8.28. When I do Barge with the option 'Bqs', Barging is not happening. I am able to listen the existing conversation. But I am not able to participate. The same issue with whisper as well.
> What might be the wrong. I am attaching the sip debug the log for reference. Please help me to find out the issue.
> Please find the sip log below.
> [Edit by Rusty - Removed reporters inline debug - reporter attached to issue]
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