G729 Problem

p0tp0t · March 7, 2012, 11:54am

Hi guys,

I have the latest FreePBX installed, with 11 G729 codecs.

The box is SIP registered to a VOIPTalk account, this is registered fine. When everything is set to ulaw/alaw I can make inbound and outbound calls with no problem.

If I set the extensions to be g729, they can call each other fine. When I call IN to the number, I see :

pbx01*CLI> g729 show licenses
1/1 encoders/decoders of 11 licensed channels are currently in use

And both sides can hear each other.

When I call OUT I see :

pbx01*CLI> g729 show licenses
0/1 encoders/decoders of 11 licensed channels are currently in use

In the SIP Trunk config i’ve tried allow=g729 and allow=all. Just seems like the encoding is having none of it!

p0tp0t · March 7, 2012, 12:43pm

Here is the SIP/Asterisk Debug info :

ere is my Asterisk output w/ SIP Debug.

[root@pbx01 ~]# asterisk -r
Asterisk 1.8.9.3, Copyright © 1999 - 2011 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 1.8.9.3 currently running on pbx01 (pid = 2906)
Verbosity is at least 10000
pbx01CLI> core set verbose 1000000000000
pbx01CLI> sip set debug on
SIP Debugging enabled
Reliably Transmitting (no NAT) to 192.168.10.254:1031:
OPTIONS sip:[email protected]:1031 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK14401f06
Max-Forwards: 70
From: “Unknown” ;tag=as0c053d10
To:
Contact:
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.0(1.8.9.3)
Date: Wed, 07 Mar 2012 12:04:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Reliably Transmitting (no NAT) to 192.168.10.254:1030:
OPTIONS sip:[email protected]:1030 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK4ce20f1d
Max-Forwards: 70
From: “Unknown” ;tag=as7fa915a7
To:
Contact:
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.0(1.8.9.3)
Date: Wed, 07 Mar 2012 12:04:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

SIP/2.0 200 OK
To: ;tag=e2e2004333f3fff1i0
From: “Unknown” ;tag=as0c053d10
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK14401f06
Server: Cisco/SPA504G-7.4.8a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

SIP/2.0 200 OK
To: ;tag=76576fa3f63e57a9i0
From: “Unknown” ;tag=as7fa915a7
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK4ce20f1d
Server: Cisco/SPA504G-7.4.8a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.10.254:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK77d5731d
Max-Forwards: 70
From: “Unknown” ;tag=as768268b4
To:
Contact:
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.0(1.8.9.3)
Date: Wed, 07 Mar 2012 12:04:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

SIP/2.0 200 OK
To: ;tag=8d5fdc2a5e6dfbb9i0
From: “Unknown” ;tag=as768268b4
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK77d5731d
Server: Cisco/SPA525G2-7.4.9c
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-de7244ec
From: “Test Extension” ;tag=be9e0da23f38457co0
To: "01299252388"
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Test Extension"
Expires: 240
User-Agent: Cisco/SPA525G2-7.4.9c
Content-Length: 215
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 7050295 7050295 IN IP4 192.168.10.254
s=-
c=IN IP4 192.168.10.254
t=0 0
m=audio 16402 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

— (14 headers 11 lines) —
Sending to 192.168.10.254:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘777’ for ‘777’ from 192.168.10.254:5060

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-de7244ec;received=192.168.10.254
From: “Test Extension” ;tag=be9e0da23f38457co0
To: “01299252388” ;tag=as1482b4d2
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="671ad371"
Content-Length: 0

Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-de7244ec
From: “Test Extension” ;tag=be9e0da23f38457co0
To: “01299252388” ;tag=as1482b4d2
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Test Extension"
User-Agent: Cisco/SPA525G2-7.4.9c
Content-Length: 0

— (10 headers 0 lines) —

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-e36dbee4
From: “Test Extension” ;tag=be9e0da23f38457co0
To: “01299252388"
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username=“777”,realm=“asterisk”,nonce=“671ad371”,uri="sip:[email protected]”,algorithm=MD5,response="9dcfc6cffb5434ae1d0d0a8ac17acdf4"
Contact: "Test Extension"
Expires: 240
User-Agent: Cisco/SPA525G2-7.4.9c
Content-Length: 215
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 7050295 7050295 IN IP4 192.168.10.254
s=-
c=IN IP4 192.168.10.254
t=0 0
m=audio 16402 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

— (15 headers 11 lines) —
Sending to 192.168.10.254:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘777’ for ‘777’ from 192.168.10.254:5060
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.10.254:16402
Looking for 01299252388 in from-internal (domain 192.168.10.200)
list_route: hop:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-e36dbee4;received=192.168.10.254
From: “Test Extension” ;tag=be9e0da23f38457co0
To: "01299252388"
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

We think we can do text
Audio is at 19018
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 77.240.48.94:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 212.74.46.35:5060;branch=z9hG4bK4c140597;rport
Max-Forwards: 70
From: “01179113714” ;tag=as4615ade9
To:
Contact:
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.0(1.8.9.3)
Date: Wed, 07 Mar 2012 12:04:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 519

v=0
o=root 363023207 363023207 IN IP4 212.74.46.35
s=Asterisk PBX 1.8.9.3
c=IN IP4 212.74.46.35
t=0 0
m=audio 19018 RTP/AVP 18 8 0 3 112 5 10 7 111 9 118 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK4c140597;rport=5060
From: “01179113714” ;tag=as4615ade9
To:
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

— (8 headers 0 lines) —

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK4c140597;rport=5060
From: “01179113714” ;tag=as4615ade9
To: ;tag=fd79486175647ed1617969929fdaad02.1661
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“voiptalk.org”, nonce="4f574ee500006c9b0eeaf8d508c32b3c1d3182e3554b486d"
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

— (9 headers 0 lines) —
set_destination: Parsing for address/port to send to
set_destination: set destination to 77.240.48.94:5060
Transmitting (NAT) to 77.240.48.94:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 212.74.46.35:5060;branch=z9hG4bK4c140597;rport
Max-Forwards: 70
From: “01179113714” ;tag=as4615ade9
To: ;tag=fd79486175647ed1617969929fdaad02.1661
Contact:
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-2.10.0(1.8.9.3)
Content-Length: 0

We think we can do text
Audio is at 19018
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 77.240.48.94:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 212.74.46.35:5060;branch=z9hG4bK6312a054;rport
Max-Forwards: 70
From: “01179113714” ;tag=as4615ade9
To:
Contact:
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.10.0(1.8.9.3)
Proxy-Authorization: Digest username=“844238829”, realm=“voiptalk.org”, algorithm=MD5, uri="sip:[email protected]", nonce=“4f574ee500006c9b0eeaf8d508c32b3c1d3182e3554b486d”, response="07d2a54d4a30a5070ecca60277e249dd"
Date: Wed, 07 Mar 2012 12:04:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 519

v=0
o=root 363023207 363023208 IN IP4 212.74.46.35
s=Asterisk PBX 1.8.9.3
c=IN IP4 212.74.46.35
t=0 0
m=audio 19018 RTP/AVP 18 8 0 3 112 5 10 7 111 9 118 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK6312a054;rport=5060
From: “01179113714” ;tag=as4615ade9
To:
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

— (8 headers 0 lines) —

SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK6312a054;rport=5060
From: “01179113714” ;tag=as4615ade9
To:
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

— (8 headers 0 lines) —

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.200:5060;received=192.168.10.200;branch=z9hG4bK6312a054;rport=5060
Record-Route:
From: “01179113714” ;tag=as4615ade9
To: ;tag=as01216cc7
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 339

v=0
o=voip 100122412 100122412 IN IP4 77.240.54.11
s=voip
c=IN IP4 77.240.54.11
t=0 0
m=audio 13426 RTP/AVP 8 0 3 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-e36dbee4;received=192.168.10.254
From: “Test Extension” ;tag=be9e0da23f38457co0
To: “01299252388” ;tag=as72e8a873
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1622965014 1622965014 IN IP4 192.168.10.200
s=Asterisk PBX 1.8.9.3
c=IN IP4 192.168.10.200
t=0 0
m=audio 15992 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.200:5060;received=192.168.10.200;branch=z9hG4bK6312a054;rport=5060
Record-Route:
From: “01179113714” ;tag=as4615ade9
To: ;tag=as01216cc7
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 339

v=0
o=voip 100122412 100122413 IN IP4 77.240.54.11
s=voip
c=IN IP4 77.240.54.11
t=0 0
m=audio 13426 RTP/AVP 8 0 3 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Audio is at 15992
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-e36dbee4;received=192.168.10.254
From: “Test Extension” ;tag=be9e0da23f38457co0
To: “01299252388” ;tag=as72e8a873
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1622965014 1622965015 IN IP4 192.168.10.200
s=Asterisk PBX 1.8.9.3
c=IN IP4 192.168.10.200
t=0 0
m=audio 15992 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-df017788
From: “Test Extension” ;tag=be9e0da23f38457co0
To: “01299252388” ;tag=as72e8a873
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“777”,realm=“asterisk”,nonce=“671ad371”,uri="sip:[email protected]",algorithm=MD5,response="9dcfc6cffb5434ae1d0d0a8ac17acdf4"
Contact: "Test Extension"
User-Agent: Cisco/SPA525G2-7.4.9c
Content-Length: 0

— (11 headers 0 lines) —

REGISTER sip:192.168.10.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:1030;branch=z9hG4bK-b96851eb
From: “Test Phone” ;tag=9f4a1f5b8228ab51o0
To: "Test Phone"
Call-ID: [email protected]
CSeq: 16797 REGISTER
Max-Forwards: 70
Authorization: Digest username=“111”,realm=“asterisk”,nonce=“1965d0c3”,uri=“sip:192.168.10.200”,algorithm=MD5,response="2b586151b35b7b2a1128dc25b42c6db8"
Contact: “Test Phone” ;expires=3600
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

— (13 headers 0 lines) —
Sending to 192.168.10.254:1030 (NAT)

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.254:1030;branch=z9hG4bK-b96851eb;received=192.168.10.254
From: “Test Phone” ;tag=9f4a1f5b8228ab51o0
To: “Test Phone” ;tag=as3c3ec70a
Call-ID: [email protected]
CSeq: 16797 REGISTER
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5de8ad4f"
Content-Length: 0

Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

REGISTER sip:192.168.10.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:1030;branch=z9hG4bK-f1e73624
From: “Test Phone” ;tag=9f4a1f5b8228ab51o0
To: "Test Phone"
Call-ID: [email protected]
CSeq: 16798 REGISTER
Max-Forwards: 70
Authorization: Digest username=“111”,realm=“asterisk”,nonce=“5de8ad4f”,uri=“sip:192.168.10.200”,algorithm=MD5,response="b141e3dd9dc1661cf71dc7e0bd1a628f"
Contact: “Test Phone” ;expires=3600
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

— (13 headers 0 lines) —
Sending to 192.168.10.254:1030 (no NAT)
Reliably Transmitting (no NAT) to 192.168.10.254:1030:
OPTIONS sip:[email protected]:1030 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK0a7928a4
Max-Forwards: 70
From: “Unknown” ;tag=as23970de5
To:
Contact:
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.0(1.8.9.3)
Date: Wed, 07 Mar 2012 12:04:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.254:1030;branch=z9hG4bK-f1e73624;received=192.168.10.254
From: “Test Phone” ;tag=9f4a1f5b8228ab51o0
To: “Test Phone” ;tag=as3c3ec70a
Call-ID: [email protected]
CSeq: 16798 REGISTER
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: ;expires=3600
Date: Wed, 07 Mar 2012 12:04:47 GMT
Content-Length: 0

Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

SIP/2.0 200 OK
To: ;tag=76576fa3f63e57a9i0
From: “Unknown” ;tag=as23970de5
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK0a7928a4
Server: Cisco/SPA504G-7.4.8a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-fae82dc
From: “Test Extension” ;tag=be9e0da23f38457co0
To: “01299252388” ;tag=as72e8a873
Call-ID: [email protected]
CSeq: 103 BYE
Max-Forwards: 70
Authorization: Digest username=“777”,realm=“asterisk”,nonce=“671ad371”,uri=“sip:[email protected]:5060”,algorithm=MD5,response="f312ba90db214c70087458e208231eb6"
User-Agent: Cisco/SPA525G2-7.4.9c
Content-Length: 0

— (10 headers 0 lines) —
Sending to 192.168.10.254:5060 (no NAT)
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: BYE)

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.254:5060;branch=z9hG4bK-fae82dc;received=192.168.10.254
From: “Test Extension” ;tag=be9e0da23f38457co0
To: “01299252388” ;tag=as72e8a873
Call-ID: [email protected]
CSeq: 103 BYE
Server: FPBX-2.10.0(1.8.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 32000 ms (Method: INVITE)
set_destination: Parsing for address/port to send to
set_destination: set destination to 77.240.48.94:5060
Reliably Transmitting (NAT) to 77.240.48.94:5060:
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 212.74.46.35:5060;branch=z9hG4bK16b41477;rport
Route:
Max-Forwards: 70
From: “01179113714” ;tag=as4615ade9
To: ;tag=as01216cc7
Call-ID: [email protected]:5060
CSeq: 104 BYE
User-Agent: FPBX-2.10.0(1.8.9.3)
Proxy-Authorization: Digest username=“844238829”, realm=“voiptalk.org”, algorithm=MD5, uri="sip:[email protected]", nonce=“4f574ee500006c9b0eeaf8d508c32b3c1d3182e3554b486d”, response="6e37f5d170680404002529a35a4f2446"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.200:5060;received=192.168.10.200;branch=z9hG4bK16b41477;rport=5060
From: “01179113714” ;tag=as4615ade9
To: ;tag=as01216cc7
Call-ID: [email protected]:5060
CSeq: 104 BYE
Server: voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: INVITE
pbx01*CLI>
Disconnected from Asterisk server

p0tp0t · March 9, 2012, 9:16am

For anybody else that has this problem when operating behind a Cyberoam Firewall.

They come as standard with a SIP module loaded into their framework, this has to be disabled for RTP packets to pass through properly.

From the CLI (on the Cyberoam) run :

cyberoam system_modules sip unload