Hi,
I am fairly new to Asterisk and I ran into the following issue when attempting to enable G722 codec. Despite enabling G722 via Settings–>SIP Settings Asterisk, the outbound INVITE doesn’t include the G722 codec. Here is what /etc/asterisk/sip_general_additional.conf looks like:
...
alwaysauthreject=yes
useragent=FPBX-2.10.1(1.8.22.0)
disallow=all
allow=g722
allow=ulaw
allow=alaw
allow=gsm
...
Here is a snippet of the sip debug trace:
Phone call is placed from the phone and as witnessed by Asterisk
<--- SIP read from UDP:<<<phone IP addr>>> --->
INVITE sip:xxx@yyyyy.zzz;user=phone SIP/2.0
…
m=audio 53936 RTP/AVP 9 0 8 3 97 98 99 100 106 107 108 109 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
…
Asterisk turns around and forwards the call setup to the next hop
Reliably Transmitting (NAT) to ...<<<SIP server>>>
INVITE xxx@yyyyy.zzz SIP/2.0
…
m=audio 11938 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
…
Besides not include G722 codec, I also notice that the ;user=phone
Lastly, here are some relevant environment info:
PIAF Installed Version = 2.0.6.4 under *HARDWARE* │
│ FreePBX Version = 2.10.1.16 │
│ Running Asterisk Version = 1.8.22.0 │
│ Asterisk Source Version = 1.8.22.0
Any help to fix this issue will be much appreciated. I am looking forward to ramp up my knowledge of asterisk and contribute to the forum.
thanks,
-Syed