g729 License Rollover Issues

Asterisk Asterisk Support
1

When we hit the limit of our g729 licenses on the latest Asterisk (1.8.7.1) get the error:

[Oct 28 18:40:41] WARNING[11584]: translate.c:256 ast_translator_build_path: No translator path from alaw to unknown

over and over until the offending call is cut, but the call still works. How do we get it to choose ulaw/alaw instead. The inbound call that is coming in is coming from the PSTN and ending up a g729, we get audio and everything seems to work, except we get the error message. The next call after the first offender also get g.729 and works fine, except now we are way out of licenses. The SIP messaging looks like:

<— SIP read from UDP:x.x.x.x:5060 —>
INVITE sip:+18664929836@x.x.x.x SIP/2.0
Record-Route: sip:x.x.x.x;lr=on;ftag=gK05668215
Record-Route: sip:67.231.8.93;lr=on;ftag=gK05668215
Accept: application/sdp
Allow: INVITE,ACK,CANCEL,BYE
Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK92a6.263af072.0
Via: SIP/2.0/UDP 67.231.8.93;branch=z9hG4bK92a6.b87e9dc4.0
Via: SIP/2.0/UDP 4.55.4.227:5060;branch=z9hG4bK05Bfb9437ae4021da5a
From: sip:+115032221111@4.55.4.227:5060;isup-oli=0;tag=gK05668215
o: sip:+18664929836@67.231.8.93:5060
Call-ID: 1241885320_52209062@4.55.4.227
CSeq: 12700 INVITE
Max-Forwards: 91
Contact: sip:+115032221111@4.55.4.227:5060
Supported: 100rel
Content-Length: 300
Content-Disposition: session; handling=required
Content-Type: application/sdp
Remote-Party-ID: sip:+115032221111@4.55.4.227:5060;privacy=off;screen=no

v=0
o=Sonus_UAC 3049 4064 IN IP4 4.55.4.227
s=SIP Media Capabilities
c=IN IP4 4.55.4.194
t=0 0
m=audio 6500 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20

== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104
(ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 4.55.4.194:6500
Looking for +18664929836 in from-trunk (domain x.x.x.x)
list_route: hop: sip:x.x.x.x;lr=on;ftag=gK05668215
list_route: hop: sip:67.231.8.93;lr=on;ftag=gK05668215

<— Transmitting (NAT) to x.x.x.x:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
x.x.x.x;branch=z9hG4bK92a6.263af072.0;received=x.x.x.x;rport=5060
Via: SIP/2.0/UDP 67.231.8.93;branch=z9hG4bK92a6.b87e9dc4.0
Via: SIP/2.0/UDP 4.55.4.227:5060;branch=z9hG4bK05Bfb9437ae4021da5a
Record-Route: sip:x.x.x.x;lr=on;ftag=gK05668215
Record-Route: sip:67.231.8.93;lr=on;ftag=gK05668215
From: sip:+115032221111@4.55.4.227:5060;isup-oli=0;tag=gK05668215
To: sip:+18664929836@67.231.8.93:5060
Call-ID: 1241885320_52209062@4.55.4.227
CSeq: 12700 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: sip:+18664929836@x.x.x.x:5060
Content-Length: 0

<------------>
– Executing [+18664929836@from-trunk:1] GOTO(“SIP/BW-SIP-A-00000005”,
“from-synclio,+18664929836,1”)
– Goto (from-synclio,+18664929836,1)
– Executing [+18664929836@from-synclio:1] AGI(“SIP/BW-SIP-A-00000005”,
“synclioparser.php”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/synclioparser.php
– AGI Script Executing Application: (MixMonitor) Options:
(1319752923.9.wav)
== Begin MixMonitor Recording SIP/BW-SIP-A-00000005
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to x.x.x.x:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP
x.x.x.x;branch=z9hG4bK92a6.263af072.0;received=x.x.x.x;rport=5060
Via: SIP/2.0/UDP 67.231.8.93;branch=z9hG4bK92a6.b87e9dc4.0
Via: SIP/2.0/UDP 4.55.4.227:5060;branch=z9hG4bK05Bfb9437ae4021da5a
Record-Route: sip:x.x.x.x;lr=on;ftag=gK05668215
Record-Route: sip:67.231.8.93;lr=on;ftag=gK05668215
From: sip:+115032221111@4.55.4.227:5060;isup-oli=0;tag=gK05668215
To: sip:+18664929836@67.231.8.93:5060;tag=as5f0e9f46
Call-ID: 1241885320_52209062@4.55.4.227
CSeq: 12700 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: sip:+18664929836@x.x.x.x:5060
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 806425519 806425519 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.7.1
c=IN IP4 x.x.x.x
t=0 0
m=audio 13636 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
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:x.x.x.x:5060 —>
ACK sip:+18664929836@x.x.x.x:5060 SIP/2.0
Record-Route: sip:x.x.x.x;lr=on;ftag=gK05668215
Record-Route: sip:67.231.8.93;lr=on;ftag=gK05668215
Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK92a6.263af072.2
Via: SIP/2.0/UDP 67.231.8.93;branch=z9hG4bK92a6.b87e9dc4.2
Via: SIP/2.0/UDP 4.55.4.227:5060;branch=z9hG4bK05Bfb9717e24021da5a
From: sip:+115032221111@4.55.4.227:5060;isup-oli=0;tag=gK05668215
To: sip:+18664929836@67.231.8.93:5060;tag=as5f0e9f46
Call-ID: 1241885320_52209062@4.55.4.227
CSeq: 12700 ACK
Max-Forwards: 68
Content-Length: 0

Any help would be great, thank you!

2

Hi there,

This is not an error message, it’s only a Warning to remind you that your licenses are over quota.Your calls are being handled in this case by the next available and alloweded codec, until a call is ended to get available G729 to translate for more channels…
But there’s no worry about this message, it’s only a warning, and you get it maybe because you have high level verbosity, just turn verbosity off, and debug off, you will not get any more this warning, as this is not critical neither fatal error…
Regards,

3

HI

I have this problem

app_mixmonitor.c: == Begin MixMonitor Recording IAX2/ulsvoip-2634
[Jan 24 17:26:04] WARNING[5268] translate.c: No translator path from alaw to unknown
[Jan 24 17:26:04] WARNING[5268] channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x6 (gsm|ulaw)
[Jan 24 17:26:04] WARNING[5268] file.c: Unable to open vm-theperson (format 0x100 (g729)): No such file or directory
[Jan 24 17:26:04] VERBOSE[5268] app_macro.c: == Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on ‘IAX2/ulsvoip-2634’ in macro ‘vm’
[Jan 24 17:26:04] VERBOSE[5268] app_macro.c: == Spawn extension (macro-exten-vm, s, 15) exited non-zero on ‘IAX2/ulsvoip-2634’ in macro ‘exten-vm’
[Jan 24 17:26:04] VERBOSE[5268] pbx.c: == Spawn extension (from-did-direct, 100, 2) exited non-zero on ‘IAX2/ulsvoip-2634’
[Jan 24 17:26:04] VERBOSE[5268] pbx.c: – Executing [h@from-did-direct:1] Macro(“IAX2/ulsvoip-2634”, “hangupcall,”) in new stack
[Jan 24 17:26:04] VERBOSE[5268] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“IAX2/ulsvoip-2634”, “1?theend”) in new stack
[Jan 24 17:26:04] VERBOSE[5268] pbx.c: – Goto (macro-hangupcall,s,3)
[Jan 24 17:26:04] VERBOSE[5268] pbx.c: – Executing [s@macro-hangupcall:3] Hangup(“IAX2/ulsvoip-2634”, “”) in new stack
[Jan 24 17:26:04] VERBOSE[5268] app_macro.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘IAX2/ulsvoip-2634’ in macro ‘hangupcall’
[Jan 24 17:26:04] VERBOSE[5268] pbx.c: == Spawn extension (from-did-direct, h, 1) exited non-zero on ‘IAX2/ulsvoip-2634’
[Jan 24 17:26:04] VERBOSE[5268] chan_iax2.c: – Hungup ‘IAX2/ulsvoip-2634’
[Jan 24 17:26:04] VERBOSE[5269] app_mixmonitor.c: == End MixMonitor Recording IAX2/ulsvoip-2634

Can some one help.