[asterisk-users] g729a compatibility

Thu Jul 2 10:43:49 CDT 2009

Hello Everybody!

Here are the full SIP logs!

<--- SIP read from 216.48.184.50:5060 --->
INVITE sip:6587972772285297 at 82.80.231.238:5060;user=phone SIP/2.0
Call-ID: 6998640000475636237-1246542986-18105
From: <sip:7188894321 at 216.48.184.50:5060;user=phone>;tag=24794
To: <sip:6587972772285297 at 82.80.231.238:5060;user=phone>
Content-Type: application/sdp
CSeq: 1 INVITE
Via: SIP/2.0/UDP 216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-1
Contact:  <sip:7188894321 at 216.48.184.50:5060;user=phone>
Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO
Supported: timer,100rel
Max-Forwards: 70
Content-Length: 255

v=0
o=MG4000|2.0 42386 70624 IN IP4 216.48.184.30
s=-
c=IN IP4 216.48.184.30
t=0 0
m=audio 33068 RTP/AVP 18 98 101 13
a=rtpmap:98 G.729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=ptime:20
a=rtpmap:13 CN/8000

<------------->
[Jul  2 16:56:26] VERBOSE[13420] logger.c: --- (12 headers 12 lines) ---
[Jul  2 16:56:26] VERBOSE[13420] logger.c: Sending to 216.48.184.50 :
5060 (no NAT)
[Jul  2 16:56:26] VERBOSE[13420] logger.c: Using INVITE request as
basis request - 6998640000475636237-1246542986-18105
[Jul  2 16:56:26] VERBOSE[13420] logger.c: Found no matching peer or
user for '216.48.184.50:5060'
[Jul  2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 18
[Jul  2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 98
[Jul  2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 101
[Jul  2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 13
[Jul  2 16:56:26] VERBOSE[13420] logger.c: Capabilities: us - 0x8000e
(gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing)
, combined - 0x0 (nothing)
[Jul  2 16:56:26] VERBOSE[13420] logger.c: Non-codec capabilities
(dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN),
co
mbined - 0x1 (telephone-event)
[Jul  2 16:56:26] VERBOSE[13420] logger.c: Looking for
6587972772285297 in didx-to-mor (domain 82.80.231.238)
[Jul  2 16:56:26] VERBOSE[13420] logger.c: list_route: hop:
<sip:7188894321 at 216.48.184.50:5060;user=phone>
[Jul  2 16:56:26] VERBOSE[13420] logger.c:
<--- Transmitting (no NAT) to 216.48.184.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-1;received=216.48.184.50
From: <sip:7188894321 at 216.48.184.50:5060;user=phone>;tag=24794
To: <sip:6587972772285297 at 82.80.231.238:5060;user=phone>
Call-ID: 6998640000475636237-1246542986-18105
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6587972772285297 at 82.80.231.238>
Content-Length: 0

<------------>
[Jul  2 16:56:26] VERBOSE[7001] logger.c:     -- Executing
[6587972772285297 at didx-to-mor:1] Set("SIP/5060-bc068a30",
"CDR(accountcode)=32
9") in new stack
[Jul  2 16:56:26] VERBOSE[7001] logger.c:     -- Executing
[6587972772285297 at didx-to-mor:2] Answer("SIP/5060-bc068a30", "") in
new stack
[Jul  2 16:56:26] VERBOSE[7001] logger.c: Audio is at 82.80.231.238 port 19616
[Jul  2 16:56:26] VERBOSE[7001] logger.c:
<--- Reliably Transmitting (no NAT) to 216.48.184.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-1;received=216.48.184.50
From: <sip:7188894321 at 216.48.184.50:5060;user=phone>;tag=24794
To: <sip:6587972772285297 at 82.80.231.238:5060;user=phone>;tag=as2a5d9a27
Call-ID: 6998640000475636237-1246542986-18105
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6587972772285297 at 82.80.231.238>
Content-Type: application/sdp
Content-Length: 150

v=0
o=root 13368 13368 IN IP4 82.80.231.238
s=session
c=IN IP4 82.80.231.238
t=0 0
m=audio 19616 RTP/AVP
a=silenceSupp:off - - - -
a=sendrecv

<------------>
[Jul  2 16:56:26] VERBOSE[7001] logger.c:     -- Executing
[6587972772285297 at didx-to-mor:3] SayDigits("SIP/5060-bc068a30",
"123456789") i
n new stack
[Jul  2 16:56:26] VERBOSE[7001] logger.c:     -- <SIP/5060-bc068a30>
Playing 'digits/1' (language 'en')
[Jul  2 16:56:26] VERBOSE[13420] logger.c:
<--- SIP read from 216.48.184.50:5060 --->
ACK sip:6587972772285297 at 82.80.231.238:5060;user=phone SIP/2.0
Call-ID: 6998640000475636237-1246542986-18105
From: <sip:7188894321 at 216.48.184.50:5060;user=phone>;tag=24794
To: <sip:6587972772285297 at 82.80.231.238:5060;user=phone>;tag=as2a5d9a27
CSeq: 1 ACK
Via: SIP/2.0/UDP 216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-1
Max-Forwards: 70
Content-Length: 0

<------------->
[Jul  2 16:56:26] VERBOSE[13420] logger.c: --- (8 headers 0 lines) ---
[Jul  2 16:56:27] VERBOSE[13420] logger.c:
<--- SIP read from 216.48.184.50:5060 --->
BYE sip:6587972772285297 at 82.80.231.238:5060;user=phone SIP/2.0
Call-ID: 6998640000475636237-1246542986-18105
From: <sip:7188894321 at 216.48.184.50:5060;user=phone>;tag=24794
To: <sip:6587972772285297 at 82.80.231.238:5060;user=phone>;tag=as2a5d9a27
CSeq: 2 BYE
Via: SIP/2.0/UDP 216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-2
Supported: timer,100rel
Max-Forwards: 70
Content-Length: 0

<------------->
[Jul  2 16:56:27] VERBOSE[13420] logger.c: --- (9 headers 0 lines) ---
[Jul  2 16:56:27] VERBOSE[13420] logger.c: Sending to 216.48.184.50 :
5060 (no NAT)
[Jul  2 16:56:27] VERBOSE[13420] logger.c:
<--- Transmitting (no NAT) to 216.48.184.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-2;received=216.48.184.50
From: <sip:7188894321 at 216.48.184.50:5060;user=phone>;tag=24794
To: <sip:6587972772285297 at 82.80.231.238:5060;user=phone>;tag=as2a5d9a27
Call-ID: 6998640000475636237-1246542986-18105
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6587972772285297 at 82.80.231.238>
Content-Length: 0

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

Thanks!
Elliot