We have a system where calls are coming in from telcos via an opensips
server and then being redirected out to a regular sip destination.
There is no NAT, DTMF features, call recording, or codec translation
being performed so I would expect asterisk to issue a reinvite after the
call is answered and switch the audio however it is not happening.
Here is the sip peer information for the call coming from opensips.
Directmedia is not specifically defined so its using the asterisk
default value.
* Name : vmpubopensips3
Description :
Secret : <Not set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-pubopensips
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language :
Tonezone : <Not set>
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 0
Max forwards : 0
Dynamic : No
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : Auto (No)
Symmetric RTP: No
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : 88.x.x.x
Addr->IP : 88.x.x.x:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username:
SIP Options : (none)
Codecs : (gsm|ulaw|alaw)
Codec Order : (alaw:20,ulaw:20,gsm:20)
Auto-Framing : No
Status : Unmonitored
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
When the call comes in the SDP contains :-
v=0.
o=root 973184584 973184584 IN IP4 81.x.x.x
s=session.
c=IN IP4 81.x.x.x
t=0 0.
m=audio 11370 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
and we reply back with :-
v=0.
o=root 822402971 822402971 IN IP4 88.x.x.x
s=Asterisk PBX 11.2-cert2.
c=IN IP4 88.x.x.x
t=0 0.
m=audio 10428 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
When we send the outbound SIP information we advertise the following SDP :-
v=0.
o=root 431105643 431105643 IN IP4 88.x.x.x
s=Asterisk PBX 11.2-cert2.
c=IN IP4 88.x.x.x
t=0 0.
m=audio 10144 RTP/AVP 8 3 0 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
and the other end replies with :-
v=0.
o=hksbc1a 609621538 609621538 IN IP4 203.x.x.x
s=sip call.
c=IN IP4 203.x.x.x
t=0 0.
m=audio 34146 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=ptime:20.
a=fmtp:101 0-15.
In the Dial() command the only option we are using is M() which is used
to run a macro when the call is answered. This is used to update cdr
records and perform other features if they are enabled. In this case we
are just updating the cdr records so I would expect the audio to be
switched as soon as the macro finishes.
Any ideas what could be wrong?
We are running Asterisk PBX 11.2-cert2
Thanks
Gareth
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