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Summary:ASTERISK-12840: Audio not passing between two Asterisk boxes when OpenSER in the middle
Reporter:Matteo (mpiazzatnetbug)Labels:
Date Opened:2008-10-07 09:48:50Date Closed:2011-06-07 14:08:02
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) rtpdebug.txt
( 1) rtpdebug2.txt
( 2) trace-a-to-a.pcap
Description:This issue is similar to the BUG number 0010481. I have two asterisk identical each other, and a Openser in the middle to route the call based on prefix (in product will be noumerous asterisk and some other sip PBX).
Every device is in the same network, so no NAT issue.
Thia is the scenario

phone1 --> Asterisk1 --> Openser --> Asterisk2 --> phone2
The asterisk is registred on Openser with a Sip trunk, this is the sip conf for the asterisk:


[phone1]
type=friend
username=phone1
callerid=("" <2678>)
secret=1234
context=users-distretto
callgroup=1
pickupgroup=1
canreinvite=yes
nat=no
host=dynamic


[siptrunk]
type=friend
username=8889
secret=8889
fromuser=8889
host=siptrunk.tnet.it
dtmfmode=rfc2833
fromdomain=siptrunk.tnet.it
context=default
insecure=very
canreinvite=yes

and on the dialplan to dial
exten => _04.,n,Dial(SIP/${EXTEN}@siptrunk)
and to recieve
exten => _04.,1,Goto(user|${EXTEN}|1)

If I have enable canreinvite=yes in the general section, on the phone and on the siptrunk I can see a lot of 491 message and the RTP stream in not forwarded correctly.

Of course If I choose canreinvit=no every were evithing is fine.

If I select canreinvite=no in general , and in the user and on the siptrunk =yes the phone 1 send the rtp stream directly to the asterisk2 and the phone 2 send the RTP stream to asterisk2.




Comments:By: Leif Madsen (lmadsen) 2008-10-07 12:21:42

You will need to provide SIP traces and SIP history in order to move this bug along. Also please provide a very brief dump of the 'rtp debug' from the console. Please also provide the relevant portions of the OpenSER configuration in order to determine if this is a configuration issue or bug.

Thanks!
By: Matteo (mpiazzatnetbug) 2008-10-08 01:59:50

Isimplified the scenario. Now I have the two asterisk talking directly without openser in the middle.

So in a 172.25.18.0/24 network

Phone 1 --> Asterisk (Udine) --> Asterisk (Aviano) --> Phone 2
.60            .67                   .66                  .83
SNOM 300                                                 LInksys

I still see the issue
as Attachment a wireshark trace. In rtpdebug.txt a dump od SIP debug and in rtpdebug2.txt a dump of rtp debug.
Matteo
By: Leif Madsen (lmadsen) 2008-12-08 16:17:01.000-0600

Can you provide the configuration on the two boxes in order to reproduce the issue locally?
By: Leif Madsen (lmadsen) 2008-12-15 10:23:11.000-0600

Changing to feedback as we'd like to have the configuration between the two Asterisk boxes in order to reproduce the issue.
By: Leif Madsen (lmadsen) 2008-12-15 10:27:03.000-0600

Actually nevermind, I just looked at the pcap capture, and this looks to be a duplicate of 12013. Thanks!
By: Joshua C. Colp (jcolp) 2008-12-17 12:41:48.000-0600

Closed as duplicate, see other issue for progress.