Voicemail and G.729

[mbmessaoud]

Hi

Is it possible to use G.729 pass-thru mode when recording or playing voice message

Any comment is welcome

[malcolmd]

Provided that your recording format is G.729 and your playback source files are G.729-encoded files, sure.

[mbmessaoud]

Ok

I configured voicemail.conf as defined in http://www.voip-info.org/wiki/view/Asterisk+G.729+pass-thru
voicemail.conf

[general] format=g729 maxsilence=0
I can hear voicemail annonce in G.729
but I cannot record mail

see log below

[quote]
– <SIP/680-00000019> Playing ‘vm-intro.g729’ (language ‘en’)
– <SIP/680-00000019> Playing ‘beep.g729’ (language ‘en’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/default/888/tmp/UtgO5i format: g729, 0x23bc928
[Sep 20 17:50:25] WARNING[3932]: format_g729.c:78 g729_write: Invalid data length, 2, should be multiple of 10
[Sep 20 17:50:25] WARNING[3932]: file.c:173 ast_writestream: Natural write failed
[Sep 20 17:50:25] WARNING[3932]: app.c:890 __ast_play_and_record: Error writing frame[/quote]

What is the problem?
Is there some others parameters to configure ?

[malcolmd]

What’s the packet size that you’re phone’s sending? Try setting it to 20ms.

[mbmessaoud]

Thanks for the help

I use AAstra 55I phone and the ptime was set to 30ms by default , when I changed it to 20 ms it works

I can leave message and play recorded messages with G.729 codec

[malcolmd]

Yay

[mbmessaoud]

Hi

I have another issue now when I selected on AASTRA phone both codecs ( G.711u and G.729) with ptime = 20 ms
I get the same problem

herafter the Asterisk CLI log with sip debug enabled

<— SIP read from UDP:192.168.1.10:5060 —>
INVITE sip:680@192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK696bb787e04a91604.396745395afa121ba
Max-Forwards: 70
From: “” sip:680@192.168.1.11:5060;tag=a138e8711f
To: “680” sip:680@192.168.1.11:5060
Call-ID: c076054cbf9a3679
CSeq: 3548 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: “” sip:680@192.168.1.10:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BE07B”
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 55i/2.6.0.1007
Content-Type: application/sdp
Content-Length: 280

v=0
o=MxSIP 0 0 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 3000 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (14 headers 14 lines) —
Sending to 192.168.1.10:5060 (no NAT)
Using INVITE request as basis request - c076054cbf9a3679
Found peer ‘680’ for ‘680’ from 192.168.1.10:5060

<— Reliably Transmitting (no NAT) to 192.168.1.10:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK696bb787e04a91604.396745395afa121ba;received=192.168.1.10
From: “” sip:680@192.168.1.11:5060;tag=a138e8711f
To: “680” sip:680@192.168.1.11:5060;tag=as50a8030e
Call-ID: c076054cbf9a3679
CSeq: 3548 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“5fe2b707”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘c076054cbf9a3679’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.10:5060 —>
ACK sip:680@192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK696bb787e04a91604.396745395afa121ba
Max-Forwards: 70
From: “” sip:680@192.168.1.11:5060;tag=a138e8711f
To: “680” sip:680@192.168.1.11:5060;tag=as50a8030e
Call-ID: c076054cbf9a3679
CSeq: 3548 ACK
User-Agent: Aastra 55i/2.6.0.1007
Content-Length: 0

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

<— SIP read from UDP:192.168.1.10:5060 —>
INVITE sip:680@192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK69b4fc12f992b022a.c6af072d722444176
Max-Forwards: 70
From: “” sip:680@192.168.1.11:5060;tag=a138e8711f
To: “680” sip:680@192.168.1.11:5060
Call-ID: c076054cbf9a3679
CSeq: 3549 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username=“680”,realm=“asterisk”,nonce=“5fe2b707”,uri=“sip:680@192.168.1.11:5060”,response=“884840c793dc4b2e28859fc87ae50e8d”,algorithm=MD5
Contact: “” sip:680@192.168.1.10:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BE07B”
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 55i/2.6.0.1007
Content-Type: application/sdp
Content-Length: 280

v=0
o=MxSIP 0 0 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 3000 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (15 headers 14 lines) —
Sending to 192.168.1.10:5060 (no NAT)
Using INVITE request as basis request - c076054cbf9a3679
Found peer ‘680’ for ‘680’ from 192.168.1.10:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|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 192.168.1.10:3000
Looking for 680 in from-office (domain 192.168.1.11:5060)
list_route: hop: sip:680@192.168.1.10:5060;transport=udp

<— Transmitting (no NAT) to 192.168.1.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK69b4fc12f992b022a.c6af072d722444176;received=192.168.1.10
From: “” sip:680@192.168.1.11:5060;tag=a138e8711f
To: “680” sip:680@192.168.1.11:5060
Call-ID: c076054cbf9a3679
CSeq: 3549 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:680@192.168.1.11:5060
Content-Length: 0

<------------>
– Executing [680@from-office:1] VoiceMail(“SIP/680-00000022”, “888@default”) in new stack
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.1.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK69b4fc12f992b022a.c6af072d722444176;received=192.168.1.10
From: “” sip:680@192.168.1.11:5060;tag=a138e8711f
To: “680” sip:680@192.168.1.11:5060;tag=as14512ddd
Call-ID: c076054cbf9a3679
CSeq: 3549 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:680@192.168.1.11:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 846127932 846127932 IN IP4 192.168.1.11
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.1.11
t=0 0
m=audio 10560 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

<— SIP read from UDP:192.168.1.10:5060 —>
ACK sip:680@192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bKfc8ba66881f88f99c.654dcdfa8cced1a5a
Max-Forwards: 70
From: “” sip:680@192.168.1.11:5060;tag=a138e8711f
To: “680” sip:680@192.168.1.11:5060;tag=as14512ddd
Call-ID: c076054cbf9a3679
CSeq: 3549 ACK
Authorization: Digest username=“680”,realm=“asterisk”,nonce=“5fe2b707”,uri=“sip:680@192.168.1.11:5060”,response=“884840c793dc4b2e28859fc87ae50e8d”,algorithm=MD5
User-Agent: Aastra 55i/2.6.0.1007
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– <SIP/680-00000022> Playing ‘vm-intro.g729’ (language ‘en’)
– <SIP/680-00000022> Playing ‘beep.g729’ (language ‘en’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/default/888/tmp/FMDYeb format: g729, 0x287c818
[Sep 21 14:20:44] WARNING[3220]: format_g729.c:78 g729_write: Invalid data length, 2, should be multiple of 10
[Sep 21 14:20:44] WARNING[3220]: file.c:173 ast_writestream: Natural write failed
[Sep 21 14:20:44] WARNING[3220]: app.c:890 __ast_play_and_record: Error writing frame
Scheduling destruction of SIP dialog ‘3937d0d47c82ca9962aa71f464f0e240@192.168.1.11:5060’ in 32000 ms (Method: NOTIFY)
== Spawn extension (from-office, 680, 1) exited non-zero on ‘SIP/680-00000022’

[mbmessaoud]

Hi
I made another test
when configuring G729 before G.711u ==> it works

<— SIP read from UDP:192.168.1.10:5060 —>
INVITE sip:680@192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bKf611c7d99ed6c7a3d.b20ebd330b041aed2
Max-Forwards: 70
From: “” sip:680@192.168.1.11:5060;tag=07b8845676
To: “680” sip:680@192.168.1.11:5060
Call-ID: ac3b3261b9879f64
CSeq: 9756 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: “” sip:680@192.168.1.10:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BE07B”;isfocus
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 55i/2.6.0.1007
Content-Type: application/sdp
Content-Length: 280

v=0
o=MxSIP 0 0 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 3000 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (14 headers 14 lines) —
Sending to 192.168.1.10:5060 (no NAT)
Using INVITE request as basis request - ac3b3261b9879f64
Found peer ‘680’ for ‘680’ from 192.168.1.10:5060

<— Reliably Transmitting (no NAT) to 192.168.1.10:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bKf611c7d99ed6c7a3d.b20ebd330b041aed2;received=192.168.1.10
From: “” sip:680@192.168.1.11:5060;tag=07b8845676
To: “680” sip:680@192.168.1.11:5060;tag=as466c97e0
Call-ID: ac3b3261b9879f64
CSeq: 9756 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“3a0ad2fe”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ac3b3261b9879f64’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.10:5060 —>
ACK sip:680@192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bKf611c7d99ed6c7a3d.b20ebd330b041aed2
Max-Forwards: 70
From: “” sip:680@192.168.1.11:5060;tag=07b8845676
To: “680” sip:680@192.168.1.11:5060;tag=as466c97e0
Call-ID: ac3b3261b9879f64
CSeq: 9756 ACK
User-Agent: Aastra 55i/2.6.0.1007
Content-Length: 0

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

<— SIP read from UDP:192.168.1.10:5060 —>
INVITE sip:680@192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK7db387b31ee8acfa9.86d121759c96029b3
Max-Forwards: 70
From: “” sip:680@192.168.1.11:5060;tag=07b8845676
To: “680” sip:680@192.168.1.11:5060
Call-ID: ac3b3261b9879f64
CSeq: 9757 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username=“680”,realm=“asterisk”,nonce=“3a0ad2fe”,uri=“sip:680@192.168.1.11:5060”,response=“4ccab0f538ffe677f1e6c1c700d075ad”,algorithm=MD5
Contact: “” sip:680@192.168.1.10:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BE07B”;isfocus
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 55i/2.6.0.1007
Content-Type: application/sdp
Content-Length: 280

v=0
o=MxSIP 0 0 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 3000 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (15 headers 14 lines) —
Sending to 192.168.1.10:5060 (no NAT)
Using INVITE request as basis request - ac3b3261b9879f64
Found peer ‘680’ for ‘680’ from 192.168.1.10:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|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 192.168.1.10:3000
Looking for 680 in from-office (domain 192.168.1.11:5060)
list_route: hop: sip:680@192.168.1.10:5060;transport=udp

<— Transmitting (no NAT) to 192.168.1.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK7db387b31ee8acfa9.86d121759c96029b3;received=192.168.1.10
From: “” sip:680@192.168.1.11:5060;tag=07b8845676
To: “680” sip:680@192.168.1.11:5060
Call-ID: ac3b3261b9879f64
CSeq: 9757 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:680@192.168.1.11:5060
Content-Length: 0

<------------>
– Executing [680@from-office:1] VoiceMail(“SIP/680-00000000”, “888@default”) in new stack
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.1.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK7db387b31ee8acfa9.86d121759c96029b3;received=192.168.1.10
From: “” sip:680@192.168.1.11:5060;tag=07b8845676
To: “680” sip:680@192.168.1.11:5060;tag=as6e2b12ce
Call-ID: ac3b3261b9879f64
CSeq: 9757 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:680@192.168.1.11:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 945601046 945601046 IN IP4 192.168.1.11
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.1.11
t=0 0
m=audio 11380 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

<— SIP read from UDP:192.168.1.10:5060 —>
ACK sip:680@192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bKbea09c187f6ce3aee.73618cbf2490bdf91
Max-Forwards: 70
From: “” sip:680@192.168.1.11:5060;tag=07b8845676
To: “680” sip:680@192.168.1.11:5060;tag=as6e2b12ce
Call-ID: ac3b3261b9879f64
CSeq: 9757 ACK
Authorization: Digest username=“680”,realm=“asterisk”,nonce=“3a0ad2fe”,uri=“sip:680@192.168.1.11:5060”,response=“4ccab0f538ffe677f1e6c1c700d075ad”,algorithm=MD5
User-Agent: Aastra 55i/2.6.0.1007
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– <SIP/680-00000000> Playing ‘vm-intro.g729’ (language ‘en’)
– <SIP/680-00000000> Playing ‘beep.g729’ (language ‘en’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/default/888/tmp/XtmqKE format: g729, 0x26f0608

Why asterisk behave differently according to the codecs order ?