[asterisk-users] Cisco AS5300 and Digium g729A codec
Roi Stork
roi.stork at gmail.com
Mon Jan 9 03:18:37 CST 2012
Hi Alex, here's the config and the sip debug output.
Guide:
xxx.xxx.xxx.xxx - the provider's cisco as5300 trunk ip add
yyy.yy.yy.yy - our asterisk 1.6.2.14 server
sip config:
type=peer
disallow=all
allow=g729
host=xxx.xxx.xxx.xxx
fromdomain=xxx.xxx.xxx.xxx
dtmfmode=rfc2833
nat=no
canreinvite=yes
context=from-trunk-sip-iaccess
sip debug:
v=0
o=root 249777024 249777024 IN IP4 yyy.yy.yy.yy
s=Asterisk PBX 1.6.2.14
c=IN IP4 yyy.yy.yy.yy
t=0 0
m=audio 13702 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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=ptime:20
a=sendrecv
---
<--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport
From: "6598715968" <sip:6598715968 at yyy.yy.yy.yy>;tag=as6e218907
To: <sip:34546598715968 at xxx.xxx.xxx.xxx>
Date: Fri, 06 Jan 2012 04:51:39 GMT
Call-ID: 7e54da423b0e6e457475ab17694e5165 at yyy.yy.yy.yy
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Retransmitting #3 (no NAT) to zzz.zz.zz.zz:5060:
OPTIONS sip:zzz.zz.zz.zz SIP/2.0
Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown at yyy.yy.yy.yy>;tag=as5c8e3f97
To: <sip:zzz.zz.zz.zz>
Contact: <sip:Unknown at yyy.yy.yy.yy>
Call-ID: 7bdb028f789e3afa58b22db472d9dfb5 at yyy.yy.yy.yy
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.14
Date: Fri, 06 Jan 2012 06:23:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:69.90.209.57:5060 --->
<------------->
Retransmitting #4 (no NAT) to zzz.zz.zz.zz:5060:
OPTIONS sip:zzz.zz.zz.zz SIP/2.0
Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown at yyy.yy.yy.yy>;tag=as5c8e3f97
To: <sip:zzz.zz.zz.zz>
Contact: <sip:Unknown at yyy.yy.yy.yy>
Call-ID: 7bdb028f789e3afa58b22db472d9dfb5 at yyy.yy.yy.yy
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.14
Date: Fri, 06 Jan 2012 06:23:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
Really destroying SIP dialog '7bdb028f789e3afa58b22db472d9dfb5 at yyy.yy.yy.yy'
Method: OPTIONS
<--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport
From: "6598715968" <sip:6598715968 at yyy.yy.yy.yy>;tag=as6e218907
To: <sip:34546598715968 at xxx.xxx.xxx.xxx>;tag=B6534850-EC6
Date: Fri, 06 Jan 2012 04:51:39 GMT
Call-ID: 7e54da423b0e6e457475ab17694e5165 at yyy.yy.yy.yy
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "6598715968"
<sip:1234#6598715968 at xxx.xxx.xxx.xxx>;party=called;screen=no;privacy=off
Contact: <sip:34546598715968 at xxx.xxx.xxx.xxx:5060>
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 223
v=0
o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx
s=SIP Call
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18132 RTP/AVP 18
c=IN IP4 xxx.xxx.xxx.xxx
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
<------------->
--- (15 headers 10 lines) ---
Found RTP audio format 18
Found audio description format G729 for ID 18
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100
(g729)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0
(nothing)
Peer audio RTP is at port xxx.xxx.xxx.xxx:18132
<--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport
From: "6598715968" <sip:6598715968 at yyy.yy.yy.yy>;tag=as6e218907
To: <sip:34546598715968 at xxx.xxx.xxx.xxx>;tag=B6534850-EC6
Date: Fri, 06 Jan 2012 04:51:39 GMT
Call-ID: 7e54da423b0e6e457475ab17694e5165 at yyy.yy.yy.yy
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:34546598715968 at xxx.xxx.xxx.xxx:5060>
Supported: replaces
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 223
v=0
o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx
s=SIP Call
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18132 RTP/AVP 18
c=IN IP4 xxx.xxx.xxx.xxx
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
<------------->
--- (15 headers 10 lines) ---
list_route: hop: <sip:34546598715968 at xxx.xxx.xxx.xxx:5060>
set_destination: Parsing <sip:34546598715968 at xxx.xxx.xxx.xxx:5060> for
address/port to send to
set_destination: set destination to xxx.xxx.xxx.xxx, port 5060
Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
ACK sip:34546598715968 at xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK17854b94;rport
Max-Forwards: 70
From: "6598715968" <sip:6598715968 at yyy.yy.yy.yy>;tag=as6e218907
To: <sip:34546598715968 at xxx.xxx.xxx.xxx>;tag=B6534850-EC6
Contact: <sip:6598715968 at yyy.yy.yy.yy>
Call-ID: 7e54da423b0e6e457475ab17694e5165 at yyy.yy.yy.yy
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.14
Content-Length: 0
---
> Channel SIP/xxx.xxx.xxx.xxx-00003693 was answered.
-- Executing [6591394459 at a2billing-callback:1]
DeadAGI("SIP/xxx.xxx.xxx.xxx-00003693",
"a2billing.php,1,callback") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- AGI Script Executing Application: (DIAL) Options:
(SIP/xxx.xxx.xxx.xxx/34546591394459,60,HRrL(370239000:61000:30000))
-- Limit Data for this call:
> timelimit = 370239000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound =
> warning_sound = timeleft
> end_sound =
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at yyy.yy.yy.yy port 14212
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
INVITE sip:34546591394459 at xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK4ea95f20;rport
Max-Forwards: 70
From: "6598715968" <sip:6598715968 at yyy.yy.yy.yy>;tag=as492477b7
To: <sip:34546591394459 at xxx.xxx.xxx.xxx>
Contact: <sip:6598715968 at yyy.yy.yy.yy>
Call-ID: 4d866149766030b331fee79f62bc2030 at yyy.yy.yy.yy
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.14
Date: Fri, 06 Jan 2012 06:23:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 331
v=0
o=root 1686167830 1686167830 IN IP4 yyy.yy.yy.yy
s=Asterisk PBX 1.6.2.14
c=IN IP4 yyy.yy.yy.yy
t=0 0
m=audio 14212 RTP/AVP 18 0 8 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
To: <sip:34546598715968 at xxx.xxx.xxx.xxx>;tag=B6534850-EC6
Date: Fri, 06 Jan 2012 04:51:39 GMT
Call-ID: 7e54da423b0e6e457475ab17694e5165 at yyy.yy.yy.yy
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "6598715968"
<sip:1234#6598715968 at xxx.xxx.xxx.xxx>;party=called;screen=no;privacy=off
Contact: <sip:34546598715968 at xxx.xxx.xxx.xxx:5060>
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 223
v=0
o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx
s=SIP Call
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18132 RTP/AVP 18
c=IN IP4 xxx.xxx.xxx.xxx
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
On Mon, Jan 9, 2012 at 4:33 PM, Alex Balashov <abalashov at evaristesys.com>wrote:
> You are hereby encouraged to post your AS5300 IOS config, sip.conf peer
> declaration, and packet capture. Those three things would aid greatly in
> diagnosis, especially the capture.
>
> --
> This message was painstakingly thumbed out on my mobile, so apologies for
> brevity, errors, and general sloppiness.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> On Jan 9, 2012, at 3:20 AM, Roi Stork <roi.stork at gmail.com> wrote:
>
> > Hi,
> >
> > We have a problem connecting to a Cisco AS5300 trunk.
> >
> > We set the sip peer to allow only g729. The call attempt is able to
> connect, but when answered, no audio is heard or transmitted.
> >
> > Our asterisk version is 1.6.2.14 . Codec is licensed, bought from Digium.
> >
> > We do not have this problem on our other providers using asterisk and
> other non-cisco systems.
> > Anyone else having this same problem?
> > --
> > _____________________________________________________________________
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>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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