Hi All
Fairly new to Asterisk ,so please bear with me on this ,i have setup a SIP trunk from my Asterisk server to a Cisco SBC which connects to the SP. I have an xlite phone registered to my Asterisk server ,on net calls work between xlite softphones. When i dial a number across the SIP trunk i see all the relevant SIP INVITES being sent by Asterisk and received by the Cisco CUBE ,but then the Asterisk server just randomly sends a BYE message to the Xlite softphone before RTP is sent ,BYE message reason being =16 ,normal call clearing. So now my question is ,other than the tcpdump i ran on the Asterisk server ,is there any other logs i can gather that could show me issues somehwere in the SIP stack ?
Ok so i have managed to get the SIP stack debugged in Asterisk server ,see logs below
<------------->
— (16 headers 12 lines) —
sip_route_dump: route/path hop: sip:XXXXXX@XXXXXX:5060
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port XXXXXXXXX:25972
[2017-02-13 07:37:47] WARNING[5340][C-00000002]: channel.c:5540 set_format: Unable to find a codec translation path: (g729) -> (ulaw)
[2017-02-13 07:37:47] WARNING[5340][C-00000002]: channel.c:5540 set_format: Unable to find a codec translation path: (ulaw) -> (g729)
[2017-02-13 07:37:47] WARNING[25223][C-00000002]: channel.c:5540 set_format: Unable to find a codec translation path: (g729) -> (ulaw|alaw)
[2017-02-13 07:37:47] WARNING[25223][C-00000002]: app_dial.c:1619 wait_for_answer: Unable to write frametype: 2
[2017-02-13 07:37:47] WARNING[25223][C-00000002]: app_dial.c:1619 wait_for_answer: Unable to write frametype: 2
[2017-02-13 07:37:47] WARNING[25223][C-00000002]: app_dial.c:1619 wait_for_answer: Unable to write frametype: 2
[2017-02-13 07:37:47] WARNING[25223][C-00000002]: app_dial.c:1619 wait_for_answer: Unable to write frametype: 2
[2017-02-13 07:37:47] WARNING[25223][C-00000002]: app_dial.c:1619 wait_for_answer: Unable to write frametype: 2
[2017-02-13 07:37:47] WARNING[25223][C-00000002]: app_dial.c:1619 wait_for_answer: Unable to write frametype: 2
So from the above it looks like the Asterisk server is trying to transcode ,would this mean that the Xlite softphone i am using is not capable of doing G729 natively ?
You need a valid G729 license or Pass through this last one method will only work if both call legs support G.729.
Note using Pass through monitor, whisper, local generation of tones, originates etc wont work
Yeah ,that is what i thought ,thanks for the feedback
Given that it costs the developer to licence G.729, and XLite is the free of charge loss leader product, I doubt that it supports G.729.