Inbound SIP Traffic CODEC Selection

Asterisk Asterisk Support
1

My G729 (snipe) hunt continues. MalcomD has graciously pointed out the codec can set in the dialplan:

[from-callcentric] ;exten => s, 1, Dial(SIP/101, 180, D(:1)); fail exten => 1777XXXXXXX,1,Set(SIP_CODEC_INBOUND=g729:60) exten => 1777XXXXXXX,n,Dial(SIP/101)

Inbound sip traffic still is ulaw (expecting g729). I am hoping that someone can point out the glaring error in the configuration code above.

[quote][Sep 29 00:05:15] NOTICE[23307]: chan_sip.c:6147 try_suggested_sip_codec: Changing codec to ‘g729’ for this call because of ${SIP_CODEC} variable
[Sep 29 00:05:15] NOTICE[23307]: chan_sip.c:6152 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because it is not shared by both ends.
[Sep 29 00:06:48] NOTICE[23374]: chan_sip.c:6147 try_suggested_sip_codec: Changing codec to ‘g729’ for this call because of ${SIP_CODEC} variable
[Sep 29 00:06:48] NOTICE[23374]: chan_sip.c:6152 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because it is not shared by both ends.
[Sep 29 00:06:48] NOTICE[23374]: chan_sip.c:6147 try_suggested_sip_codec: Changing codec to ‘g729’ for this call because of ${SIP_CODEC} variable
[Sep 29 00:06:48] NOTICE[23374]: chan_sip.c:6152 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because it is not shared by both ends.
– Remotely bridging SIP/callcentric.com-0000001a and SIP/101-0000001b
[Sep 29 00:06:48] NOTICE[22985]: chan_sip.c:6147 try_suggested_sip_codec: Changing codec to ‘g729’ for this call because of ${SIP_CODEC} variable
[Sep 29 00:06:48] NOTICE[22985]: chan_sip.c:6152 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because it is not shared by both ends.

athomehost*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
204.11.192.37 88351000XXXXXXX e4c1db-6998d4f2 0x4 (ulaw) No Tx: ACK
204.11.192.37 1777XXXXXX 4826fe866119cd4 0x0 (nothing) No
192.168.8.110 101 339577bc3c0db22 0x4 (ulaw) No Tx: ACK 101
3 active SIP dialogs[/quote]

2

Either your incoming channel hasn’t offered G.729 or you have not listed it as an allowed codec for the incoming channel.

What is the verbose and debug output when your run sip set debug on for the incoming channel?

3

Do:

exten => 1777XXXXXXX,1,Set(__SIP_CODEC_INBOUND=g729:60)
4

OBSERVATIONS

Gtalk outbound is not working because of codec negotiation problems, despite working nicely for several months. Gtalk Inbound is OK. Apparently mastering Codec negotiation is the root cause of my * challenges.

TEST RESULTS

Thanks for the suggest Malcom. Here’s the test results:

  • Reloaded Dialplan: changed inbound SIP per MalcolmD

[from-callcentric] exten => 1777XXXYYYY,1,Set(__SIP_CODEC_INBOUND=g729:60) exten => 1777XXXYYYY,n,Dial(SIP/101)
I do not know what the _ in front of SIP_CODEC… implies. I was also wondering about the 60 in g729:60. Google did not help me here.

  • Outbound SIP test results: G729 codec used (Asterisk Callcentric account call to another Callcentric account on Line1)
  • Tested inbound SIP: ulaw codec used
    Question: is the expectation that inbound SIP calls should use G729 unfounded \ unreasonable?

@david55

SIP traffic details (vebose?):

[quote][Sep 29 16:19:24] NOTICE[8110]: chan_sip.c:6154 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): g729:60
[Sep 29 16:19:24] NOTICE[8110]: chan_sip.c:6154 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): g729:60
– Remotely bridging SIP/callcentric.com-00000036 and SIP/101-00000037
[Sep 29 16:19:24] NOTICE[22985]: chan_sip.c:6154 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): g729:60

athomehost*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
192.168.8.110 101 06e4c73d0790198 0x4 (ulaw) No Tx: ACK 101
204.11.192.23 1777xxxyyyy 4826fe866119cd4 0x0 (nothing) No
204.11.192.23 88351000XXXXYY 1ef0f89b-f56dd3 0x4 (ulaw) No Tx: ACK
204.11.192.38 1777xxxyyyy 070f709f19a652d 0x0 (nothing) No Rx: BYE callcentri
4 active SIP dialogs
[Sep 29 16:19:34] NOTICE[8110]: chan_sip.c:6154 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): g729:60
at[/quote]

5

You haven’t run sip set debug on.

The number after g729 controls the frame size.

6

The underscores cause the value of the channel variable to propagate to channels that are created. As this is for the incoming channel, I didn’t think they were necessary.

One underscore causes the value to propagate once. Two cause it to propagate indefinitely.

7

oh…gtalk. The gtalk channel driver in Asterisk doesn’t negotiate G.729. Doesn’t matter if Google supports it; Asterisk won’t negotiate it.

8

@david55: thanks for the code clarification. Good stuff.

@malcom, I miscommunicated: I had used GV’s PSTN to make a SIP call to my Asterisk server. Sorry for the confusion. I also tested inbound Asterisk SIP by calling from another SIP account. As I understand it, G729 is not supported by Gtalk at either the Gtalk channel driver or Google side.

As of today, my inbound Gtalk channel works, HOWEVER outbound Gtalk is not working because of Codec Negotation Problems . I find it curious that outbound Gtalk codec negotiation (G711) fails whereas inbound codec negotiation is successful.

Interesting that outbound G711 codec negotiation is broken and inbound G729 codec negotiation is failing as well. The former causes more problems in the short term, however, in the long term the latter is more important to me.

I appreciate the generous support and any suggestions that would help with codec negotiation mastery.

9

@david55: Inbound SIP call to Asterisk, originated from callcentric.com. Here is the verbose traffic :

[quote]athomehost*CLI> sip set debug on
SIP Debugging enabled
Really destroying SIP dialog ‘4826fe866119cd4d1e7fe3d162b9a4f1@[c0a8:801:78fb:8a7f::]’ Method: REGISTER

<— SIP read from UDP:204.11.192.34:5060 —>
INVITE sip:1777#######@98.71.227.233:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.34:5060;branch=z9hG4bK-1c1d8b201a2cfe56b85377bc28782eeb
f: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
t: sip:1777#######@callcentric.com
i: 209add14-474c9bbf@192.168.8.110
CSeq: 102 INVITE
Max-Forwards: 13
m: sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp
Supported: x-sipura, replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Expires: 240
c: application/sdp
l: 488

v=0
o=1777*******1 1 IN IP4 204.11.192.34
s=-
c=IN IP4 204.11.192.34
t=0 0
m=audio 53030 RTP/AVP 18 0 2 4 8 96 97 98 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
— (13 headers 22 lines) —
Sending to 204.11.192.34:5060 (no NAT)
Using INVITE request as basis request - 209add14-474c9bbf@192.168.8.110
No matching peer for ‘myCallerID’ from '204.11.192.34:5060’
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
Found audio description format G726-40 for ID 96
Found audio description format G726-24 for ID 97
Found audio description format G726-16 for ID 98
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
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)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 204.11.192.34:53030
Looking for 1777####### in from-callcentric (domain 98.71.227.233:5060)
list_route: hop: sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp

— Transmitting (no NAT) to 204.11.192.34:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.34:5060;branch=z9hG4bK-1c1d8b201a2cfe56b85377bc28782eeb;received=204.11.192.34
From: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
To: sip:1777#######@callcentric.com
Call-ID: 209add14-474c9bbf@192.168.8.110
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: sip:1777#######@98.71.227.233:5060
Content-Length: 0

<------------>
We think we can do text
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.8.110:5060:
INVITE sip:101@192.168.8.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK2542a242;rport
Max-Forwards: 70
From: “myCallerID” sip:myCallerID@192.168.8.1;tag=as15b5a8d7
To: sip:101@192.168.8.110:5060
Contact: sip:myCallerID@192.168.8.1:5060
Call-ID: 27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.4
Date: Thu, 29 Sep 2011 17:08:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 374

v=0
o=root 1419777041 1419777041 IN IP4 192.168.8.1
s=Asterisk PBX 1.8.4
c=IN IP4 192.168.8.1
t=0 0
m=audio 13888 RTP/AVP 0 3 112 10 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:10 L16/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from UDP:192.168.8.110:5060 —>
SIP/2.0 100 Trying
To: sip:101@192.168.8.110:5060
From: “myCallerID” sip:myCallerID@192.168.8.1;tag=as15b5a8d7
Call-ID: 27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK2542a242
Server: Linksys/SPA2102-5.1.5(a)
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.8.110:5060 —>
SIP/2.0 180 Ringing
To: sip:101@192.168.8.110:5060;tag=7944f7a3b0f32aabi1
From: “myCallerID” sip:myCallerID@192.168.8.1;tag=as15b5a8d7
Call-ID: 27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK2542a242
Server: Linksys/SPA2102-5.1.5(a)
Remote-Party-ID: L2SPA2102 sip:101@192.168.8.1;screen=yes;party=called
Content-Length: 0

<------------->
— (9 headers 0 lines) —

<— Transmitting (no NAT) to 204.11.192.34:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 204.11.192.34:5060;branch=z9hG4bK-1c1d8b201a2cfe56b85377bc28782eeb;received=204.11.192.34
From: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
To: sip:1777#######@callcentric.com;tag=as000a614c
Call-ID: 209add14-474c9bbf@192.168.8.110
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: sip:1777#######@98.71.227.233:5060
Content-Length: 0

<------------>
[Sep 29 17:09:02] NOTICE[22985]: chan_sip.c:12365 sip_reregister: – Re-registration for 1777#######@callcentric.com
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 204.11.192.39:5060:
REGISTER sip:callcentric.com SIP/2.0
Via: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK434941e4
Max-Forwards: 70
From: sip:1777#######@callcentric.com;tag=as437e5b44
To: sip:1777#######@callcentric.com
Call-ID: 4826fe866119cd4d1e7fe3d162b9a4f1@[c0a8:801:78fb:8a7f::]
CSeq: 3999 REGISTER
User-Agent: Asterisk PBX 1.8.4
Authorization: Digest username=“1777#######”, realm=“callcentric.com”, algorithm=MD5, uri=“sip:callcentric.com”, nonce=“407a45a6e74aaa5913a1266734194338”, response="652aba3ad659cadf2ed959d446260039"
Expires: 120
Contact: sip:1777#######@98.71.227.233:5060
Content-Length: 0

<— SIP read from UDP:192.168.8.110:5060 —>
SIP/2.0 200 OK
To: sip:101@192.168.8.110:5060;tag=7944f7a3b0f32aabi1
From: “myCallerID” sip:myCallerID@192.168.8.1;tag=as15b5a8d7
Call-ID: 27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK2542a242
Contact: L2SPA2102 sip:101@192.168.8.110:5060
Server: Linksys/SPA2102-5.1.5(a)
Remote-Party-ID: L2SPA2102 sip:101@192.168.8.1;screen=yes;party=called
Content-Length: 255
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 312179 312179 IN IP4 192.168.8.110
s=-
c=IN IP4 192.168.8.110
t=0 0
m=audio 16440 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
— (13 headers 13 lines) —
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.8.110:16440
list_route: hop: sip:101@192.168.8.110:5060
set_destination: Parsing sip:101@192.168.8.110:5060 for address/port to send to
set_destination: set destination to 192.168.8.110:5060
Transmitting (NAT) to 192.168.8.110:5060:
ACK sip:101@192.168.8.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK66e8e701;rport
Max-Forwards: 70
From: “myCallerID” sip:myCallerID@192.168.8.1;tag=as15b5a8d7
To: sip:101@192.168.8.110:5060;tag=7944f7a3b0f32aabi1
Contact: sip:myCallerID@192.168.8.1:5060
Call-ID: 27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.4
Content-Length: 0

[Sep 29 17:09:02] NOTICE[10644]: chan_sip.c:6154 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): g729:60
[Sep 29 17:09:02] NOTICE[10644]: chan_sip.c:6154 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): g729:60
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 204.11.192.34:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.34:5060;branch=z9hG4bK-1c1d8b201a2cfe56b85377bc28782eeb;received=204.11.192.34
From: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
To: sip:1777#######@callcentric.com;tag=as000a614c
Call-ID: 209add14-474c9bbf@192.168.8.110
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: sip:1777#######@98.71.227.233:5060
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1829221621 1829221621 IN IP4 98.71.227.233
s=Asterisk PBX 1.8.4
c=IN IP4 98.71.227.233
t=0 0
m=audio 16284 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<— SIP read from UDP:204.11.192.39:5060 —>
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK434941e4
f: sip:1777#######@callcentric.com;tag=as437e5b44
t: sip:1777#######@callcentric.com
i: 4826fe866119cd4d1e7fe3d162b9a4f1@[c0a8:801:78fb:8a7f::]
CSeq: 3999 REGISTER
Proxy-Authenticate: Digest realm=“callcentric.com”, domain=“sip:callcentric.com”, nonce=“c0222f1b39ce11c465a5b7b784ec075c”, opaque="", stale=TRUE, algorithm=MD5
l: 0

<------------->
— (8 headers 0 lines) —
Responding to challenge, registration to domain/host name callcentric.com
– Remotely bridging SIP/callcentric.com-0000003a and SIP/101-0000003b
set_destination: Parsing sip:101@192.168.8.110:5060 for address/port to send to
set_destination: set destination to 192.168.8.110:5060
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.8.110:5060:
INVITE sip:101@192.168.8.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK4547d880;rport
Max-Forwards: 70
From: “myCallerID” sip:myCallerID@192.168.8.1;tag=as15b5a8d7
To: sip:101@192.168.8.110:5060;tag=7944f7a3b0f32aabi1
Contact: sip:myCallerID@192.168.8.1:5060
Call-ID: 27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1419777041 1419777042 IN IP4 204.11.192.34
s=Asterisk PBX 1.8.4
c=IN IP4 204.11.192.34
t=0 0
m=audio 53030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 204.11.192.39:5060:
REGISTER sip:callcentric.com SIP/2.0
Via: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK4e2f6232
Max-Forwards: 70
From: sip:1777#######@callcentric.com;tag=as150c9838
To: sip:1777#######@callcentric.com
Call-ID: 4826fe866119cd4d1e7fe3d162b9a4f1@[c0a8:801:78fb:8a7f::]
CSeq: 4000 REGISTER
User-Agent: Asterisk PBX 1.8.4
Proxy-Authorization: Digest username=“1777#######”, realm=“callcentric.com”, algorithm=MD5, uri=“sip:callcentric.com”, nonce=“c0222f1b39ce11c465a5b7b784ec075c”, response="470cf4040ef8e0ee862a00bfd1b1e2a0"
Expires: 120
Contact: sip:1777#######@98.71.227.233:5060
Content-Length: 0

<— SIP read from UDP:192.168.8.110:5060 —>
SIP/2.0 200 OK
To: sip:101@192.168.8.110:5060;tag=7944f7a3b0f32aabi1
From: “myCallerID” sip:myCallerID@192.168.8.1;tag=as15b5a8d7
Call-ID: 27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK4547d880
Contact: L2SPA2102 sip:101@192.168.8.110:5060
Server: Linksys/SPA2102-5.1.5(a)
Content-Length: 282
Content-Type: application/sdp

v=0
o=- 312624 312624 IN IP4 192.168.8.110
s=-
c=IN IP4 192.168.8.110
t=0 0
m=audio 16440 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=silenceSupp:off - - - -
<------------->
— (10 headers 14 lines) —
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.8.110:16440
set_destination: Parsing sip:101@192.168.8.110:5060 for address/port to send to
set_destination: set destination to 192.168.8.110:5060
Transmitting (NAT) to 192.168.8.110:5060:
ACK sip:101@192.168.8.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK2935a5f6;rport
Max-Forwards: 70
From: “myCallerID” sip:myCallerID@192.168.8.1;tag=as15b5a8d7
To: sip:101@192.168.8.110:5060;tag=7944f7a3b0f32aabi1
Contact: sip:myCallerID@192.168.8.1:5060
Call-ID: 27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.4
Content-Length: 0

<— SIP read from UDP:204.11.192.39:5060 —>
SIP/2.0 200 Ok
v: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK4e2f6232
f: sip:1777#######@callcentric.com;tag=as150c9838
t: sip:1777#######@callcentric.com
i: 4826fe866119cd4d1e7fe3d162b9a4f1@[c0a8:801:78fb:8a7f::]
CSeq: 4000 REGISTER
m: sip:1777#######@98.71.227.233:5060;expires=60
l: 0

<------------->
— (8 headers 0 lines) —
Scheduling destruction of SIP dialog ‘4826fe866119cd4d1e7fe3d162b9a4f1@[c0a8:801:78fb:8a7f::]’ in 32000 ms (Method: REGISTER)
[Sep 29 17:09:02] NOTICE[22985]: chan_sip.c:19760 handle_response_register: Outbound Registration: Expiry for callcentric.com is 60 sec (Scheduling reregistration in 45 s)

<— SIP read from UDP:204.11.192.39:5060 —>
NOTIFY sip:1777#######@98.71.227.233:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.39:5060;branch=z9hG4bK-3e71df3555bb76cd6074a4371b1a5ad4
f: sip:1777#######@callcentric.com:5062
t: sip:1777#######@callcentric.com
i: 9fb7d4dfffcf8acac1129912c45bacb4-26257e … entric.com
CSeq: 1 NOTIFY
Max-Forwards: 15
m: sip:06e5e45741f34d840e67a79ca92fc07e@204.11.192.39:5060;transport=udp
Event: message-summary
c: application/simple-message-summary
l: 138

Messages-Waiting: no
Message-Account: sip:1777#######@callcentric.com
Voice-Message: 0/0 (0/0)
Fax-Message: 0/0 (0/0)
None: 0/0 (0/0)
<------------->
— (11 headers 5 lines) —

<— SIP read from UDP:204.11.192.34:5060 —>
ACK sip:1777#######@98.71.227.233:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.34:5060;branch=z9hG4bK-1e731acecca4cfec64741e3573befc74
f: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
t: sip:1777#######@callcentric.com;tag=as000a614c
i: 209add14-474c9bbf@192.168.8.110
CSeq: 102 ACK
Max-Forwards: 13
m: sip:60f0b31d81b89474c70d2c3b45ebd7c9@204.11.192.34:5060;transport=udp
l: 0

<------------->
— (9 headers 0 lines) —
set_destination: Parsing sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp for address/port to send to
set_destination: set destination to 204.11.192.34:5060
[Sep 29 17:09:02] NOTICE[22985]: chan_sip.c:6154 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): g729:60
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 204.11.192.34:5060:
INVITE sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK7283e0f1
Max-Forwards: 70
From: sip:1777#######@callcentric.com;tag=as000a614c
To: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
Contact: sip:1777#######@98.71.227.233:5060
Call-ID: 209add14-474c9bbf@192.168.8.110
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1829221621 1829221622 IN IP4 192.168.8.110
s=Asterisk PBX 1.8.4
c=IN IP4 192.168.8.110
t=0 0
m=audio 16440 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from UDP:204.11.192.34:5060 —>
SIP/2.0 100 Trying
v: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK7283e0f1
f: sip:1777#######@callcentric.com;tag=as000a614c
t: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
i: 209add14-474c9bbf@192.168.8.110
CSeq: 102 INVITE
l: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:204.11.192.34:5060 —>
SIP/2.0 200 OK
v: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK7283e0f1
f: sip:1777#######@callcentric.com;tag=as000a614c
t: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
i: 209add14-474c9bbf@192.168.8.110
CSeq: 102 INVITE
m: sip:0d4ce71cb9db410ba7f48a4669a7cf50@204.11.192.34:5060;transport=udp
c: application/sdp
l: 299

v=0
o=1777*******1 1 IN IP4 204.11.192.34
s=-
c=IN IP4 204.11.192.34
t=0 0
m=audio 53030 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
— (9 headers 15 lines) —
set_destination: Parsing sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp for address/port to send to
set_destination: set destination to 204.11.192.34:5060
Transmitting (no NAT) to 204.11.192.34:5060:
ACK sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK4297a38d
Max-Forwards: 70
From: sip:1777#######@callcentric.com;tag=as000a614c
To: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
Contact: sip:1777#######@98.71.227.233:5060
Call-ID: 209add14-474c9bbf@192.168.8.110
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.4
Content-Length: 0

<— SIP read from UDP:204.11.192.34:5060 —>
SIP/2.0 200 OK
v: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK7283e0f1
f: sip:1777#######@callcentric.com;tag=as000a614c
t: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
i: 209add14-474c9bbf@192.168.8.110
CSeq: 102 INVITE
m: sip:0d4ce71cb9db410ba7f48a4669a7cf50@204.11.192.34:5060;transport=udp
c: application/sdp
l: 299

v=0
o=1777*******1 1 IN IP4 204.11.192.34
s=-
c=IN IP4 204.11.192.34
t=0 0
m=audio 53030 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
— (9 headers 15 lines) —
set_destination: Parsing sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp for address/port to send to
set_destination: set destination to 204.11.192.34:5060
Transmitting (no NAT) to 204.11.192.34:5060:
ACK sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK51eb2d6c
Max-Forwards: 70
From: sip:1777#######@callcentric.com;tag=as000a614c
To: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
Contact: sip:1777#######@98.71.227.233:5060
Call-ID: 209add14-474c9bbf@192.168.8.110
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.4
Content-Length: 0

<— SIP read from UDP:192.168.8.110:5060 —>
BYE sip:myCallerID@192.168.8.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-b8046255
From: sip:101@192.168.8.110:5060;tag=7944f7a3b0f32aabi1
To: “myCallerID” sip:myCallerID@192.168.8.1;tag=as15b5a8d7
Call-ID: 27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Linksys/SPA2102-5.1.5(a)
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.8.110:5060 (NAT)
Scheduling destruction of SIP dialog ‘27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060’ in 6400 ms (Method: BYE)

<— Transmitting (NAT) to 192.168.8.110:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-b8046255;received=192.168.8.110;rport=5060
From: sip:101@192.168.8.110:5060;tag=7944f7a3b0f32aabi1
To: “myCallerID” sip:myCallerID@192.168.8.1;tag=as15b5a8d7
Call-ID: 27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060
CSeq: 101 BYE
Server: Asterisk PBX 1.8.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0

<------------>
set_destination: Parsing sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp for address/port to send to
set_destination: set destination to 204.11.192.34:5060
[Sep 29 17:09:07] NOTICE[10644]: chan_sip.c:6154 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): g729:60
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 204.11.192.34:5060:
INVITE sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK62526466
Max-Forwards: 70
From: sip:1777#######@callcentric.com;tag=as000a614c
To: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
Contact: sip:1777#######@98.71.227.233:5060
Call-ID: 209add14-474c9bbf@192.168.8.110
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1829221621 1829221623 IN IP4 98.71.227.233
s=Asterisk PBX 1.8.4
c=IN IP4 98.71.227.233
t=0 0
m=audio 16284 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Scheduling destruction of SIP dialog ‘209add14-474c9bbf@192.168.8.110’ in 32000 ms (Method: ACK)

<— SIP read from UDP:204.11.192.34:5060 —>
SIP/2.0 100 Trying
v: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK62526466
f: sip:1777#######@callcentric.com;tag=as000a614c
t: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
i: 209add14-474c9bbf@192.168.8.110
CSeq: 103 INVITE
l: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:204.11.192.34:5060 —>
SIP/2.0 200 OK
v: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK62526466
f: sip:1777#######@callcentric.com;tag=as000a614c
t: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
i: 209add14-474c9bbf@192.168.8.110
CSeq: 103 INVITE
m: sip:0d4ce71cb9db410ba7f48a4669a7cf50@204.11.192.34:5060;transport=udp
c: application/sdp
l: 299

v=0
o=1777*******1 1 IN IP4 204.11.192.34
s=-
c=IN IP4 204.11.192.34
t=0 0
m=audio 53030 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
— (9 headers 15 lines) —
set_destination: Parsing sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp for address/port to send to
set_destination: set destination to 204.11.192.34:5060
Transmitting (no NAT) to 204.11.192.34:5060:
ACK sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK1000568f
Max-Forwards: 70
From: sip:1777#######@callcentric.com;tag=as000a614c
To: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
Contact: sip:1777#######@98.71.227.233:5060
Call-ID: 209add14-474c9bbf@192.168.8.110
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.4
Content-Length: 0

set_destination: Parsing sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp for address/port to send to
set_destination: set destination to 204.11.192.34:5060
Reliably Transmitting (no NAT) to 204.11.192.34:5060:
BYE sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK2c63b7bc
Max-Forwards: 70
From: sip:1777#######@callcentric.com;tag=as000a614c
To: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
Call-ID: 209add14-474c9bbf@192.168.8.110
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.4
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

Scheduling destruction of SIP dialog ‘209add14-474c9bbf@192.168.8.110’ in 32000 ms (Method: ACK)
Retransmitting #1 (no NAT) to 204.11.192.34:5060:
BYE sip:4d29f3ad6eb8184e94afe7e8de2dcbf8@204.11.192.34:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK2c63b7bc
Max-Forwards: 70
From: sip:1777#######@callcentric.com;tag=as000a614c
To: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
Call-ID: 209add14-474c9bbf@192.168.8.110
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.4
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from UDP:204.11.192.34:5060 —>
SIP/2.0 100 Trying
v: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK2c63b7bc
f: sip:1777#######@callcentric.com;tag=as000a614c
t: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
i: 209add14-474c9bbf@192.168.8.110
CSeq: 104 BYE
l: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:204.11.192.34:5060 —>
SIP/2.0 100 Trying
v: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK2c63b7bc
f: sip:1777#######@callcentric.com;tag=as000a614c
t: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
i: 209add14-474c9bbf@192.168.8.110
CSeq: 104 BYE
l: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:204.11.192.34:5060 —>
SIP/2.0 200 OK
v: SIP/2.0/UDP 98.71.227.233:5060;branch=z9hG4bK2c63b7bc
f: sip:1777#######@callcentric.com;tag=as000a614c
t: “myCallerID” sip:myCallerID@callcentric.com;tag=b6b47510539183f3o0
i: 209add14-474c9bbf@192.168.8.110
CSeq: 104 BYE
l: 0

<------------->
— (7 headers 0 lines) —
Reliably Transmitting (NAT) to 192.168.8.110:5060:
OPTIONS sip:101@192.168.8.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK4aa0de64;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.8.1;tag=as56ab2f82
To: sip:101@192.168.8.110:5060
Contact: sip:asterisk@192.168.8.1:5060
Call-ID: 08588299135328970efdf0ff73398db0@192.168.8.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.4
Date: Thu, 29 Sep 2011 17:09:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0

Really destroying SIP dialog ‘209add14-474c9bbf@192.168.8.110’ Method: ACK

<— SIP read from UDP:192.168.8.110:5060 —>
SIP/2.0 200 OK
To: sip:101@192.168.8.110:5060;tag=f5214d5a1fdfb6d2i1
From: “asterisk” sip:asterisk@192.168.8.1;tag=as56ab2f82
Call-ID: 08588299135328970efdf0ff73398db0@192.168.8.1:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK4aa0de64
Server: Linksys/SPA2102-5.1.5(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘08588299135328970efdf0ff73398db0@192.168.8.1:5060’ Method: OPTIONS
Really destroying SIP dialog ‘27d745586f8e709425b03ecb6c5f5de7@192.168.8.1:5060’ Method: BYE
athomehostCLI> sip set debug off
SIP Debugging Disabled
athomehostCLI>[/quote]

10

You are only allowing:

gsm ulaw alaw h263 testlaw

The only ones in common with what it offers are ulaw and alaw.

You cannot force the use a codec that you didn’t allow in the first place.

On the outgoing side, they are offering ulaw and something they call NSE. Of those, you’ve only allowed ulaw.