Hey, guys!
Recently I used G729 on one of our servers, so we decided to implement it on most of our solutions.
Asterisk 11.11.0
Following the guide, I installed the codecs and licensed it.
CLI is able to use g729 commands, however I do not see the codec in the translate table
Moreover, any call that was made from/to the SIP that has
disallow=all
allow=g729
gives me
and immediately drops the call.
Any other calls, using regular (alaw, ulaw, gsm) codecs went well with no errors/warnings.
g729 show licenses
[code]0/0 encoders/decoders of 0 licensed channels are currently in use
Licenses Found:
File: info that you dont need
Channels: 5 (Expires: 2034-09-11) (OK)[/code]
When I do module reload codec_g729a.so I get
sip debug for you
[code]—
– Called SIP/789
[2014-09-17 13:10:56] WARNING[32688][C-00000182]: channel.c:6208 ast_channel_make_compatible_helper: No path to translate from SIP/789-000003c1 to SIP/edm-000003c0
Scheduling destruction of SIP dialog ‘229ef0f27d73dcc50e9c3cb15a33dce4@90.0.3.7:5060’ in 11904 ms (Method: INVITE)
== Spawn extension (defaultin, 789, 1) exited non-zero on ‘SIP/edm-000003c0’
– Executing [h@defaultin:1] Set(“SIP/edm-000003c0”, “CDR(accountcode)=ER:0”) in new stack
Scheduling destruction of SIP dialog ‘0f1fbd9a6d3973a24b10c2af70c91a40@phtelin.asp.secure5.net’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:4036087153@90.0.3.6:5060 for address/port to send to
set_destination: set destination to 90.0.3.6:5060
Reliably Transmitting (NAT) to 90.0.3.6:5060:
BYE sip:4036087153@90.0.3.6:5060 SIP/2.0
Via: SIP/2.0/UDP 90.0.3.7:5060;branch=z9hG4bK629c63e2;rport
Max-Forwards: 70
From: sip:4032970270@phtelin.asp.secure5.net;tag=as5fce66ad
To: sip:4036087153@phtelin.asp.secure5.net;tag=as5093121d
Call-ID: 0f1fbd9a6d3973a24b10c2af70c91a40@phtelin.asp.secure5.net
CSeq: 102 BYE
User-Agent: FPBX-AsteriskNOW-2.11.0(11.11.0)
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0
== MixMonitor close filestream (mixed)
<— SIP read from UDP:90.0.3.6:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 90.0.3.7:5060;branch=z9hG4bK629c63e2;received=90.0.3.7;rport=5060
From: sip:4032970270@phtelin.asp.secure5.net;tag=as5fce66ad
To: sip:4036087153@phtelin.asp.secure5.net;tag=as5093121d
Call-ID: 0f1fbd9a6d3973a24b10c2af70c91a40@phtelin.asp.secure5.net
CSeq: 102 BYE
Server: FPBX-AsteriskNOW-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------->
— (10 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘0f1fbd9a6d3973a24b10c2af70c91a40@phtelin.asp.secure5.net’ Method: ACK
== Executing [/home/nj/mon.sh “1410981049.1037” “IN|4036087153|4036087153|4032970270”]
== End MixMonitor Recording SIP/edm-000003c0
Retransmitting #1 (NAT) to 90.0.5.55:5060:
INVITE sip:789@90.0.5.55:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 90.0.3.7:5060;branch=z9hG4bK588b02ea;rport
Max-Forwards: 70
From: sip:4036087153@90.0.3.7;tag=as051d62c3
To: sip:789@90.0.5.55:5060;transport=udp
Contact: sip:4036087153@90.0.3.7:5060
Call-ID: 229ef0f27d73dcc50e9c3cb15a33dce4@90.0.3.7:5060
CSeq: 102 INVITE
User-Agent: FPBX-AsteriskNOW-2.11.0(11.11.0)
Date: Wed, 17 Sep 2014 19:10:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 1054200832 1054200832 IN IP4 90.0.3.7
s=Asterisk PBX 11.11.0
c=IN IP4 90.0.3.7
t=0 0
m=audio 19878 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 read from UDP:90.0.5.55:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 90.0.3.7:5060;branch=z9hG4bK588b02ea;rport=5060;received=90.0.3.7
From: sip:4036087153@90.0.3.7;tag=as051d62c3
To: sip:789@90.0.5.55:5060;transport=udp;tag=1261343697
Call-ID: 229ef0f27d73dcc50e9c3cb15a33dce4@90.0.3.7:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: “789” sip:789@90.0.5.55:5060;transport=udp;+sip.instance="urn:uuid:00000000-0000-1000-8000-00085D266E81"
Server: Aastra 9480i/3.2.2.3109
Supported: path
Content-Length: 0
<------------->
— (12 headers 0 lines) —
list_route: hop: sip:789@90.0.5.55:5060;transport=udp
Reliably Transmitting (NAT) to 90.0.5.55:5060:
CANCEL sip:789@90.0.5.55:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 90.0.3.7:5060;branch=z9hG4bK588b02ea;rport
Max-Forwards: 70
From: sip:4036087153@90.0.3.7;tag=as051d62c3
To: sip:789@90.0.5.55:5060;transport=udp
Call-ID: 229ef0f27d73dcc50e9c3cb15a33dce4@90.0.3.7:5060
CSeq: 102 CANCEL
User-Agent: FPBX-AsteriskNOW-2.11.0(11.11.0)
Content-Length: 0
[/code]
What am I missing in my config?