Shariq Khan
2011-Apr-07 08:00 UTC
[asterisk-users] Asterisk Avaya SIP Trunking One Way Audio
I am facing one way audio problem in sip trunking between asterisk and
avaya.
+-------------+ +----+
| avaya sip |-------| P1 |
+-------------+ +----+
|
|
|
+-------------+
| Asterisk | WAN
-------------------------------------------------
| | LAN
+-------------+
|
/
+----+ /
| P2 |--+
+----+
When P1 dial P2, P2 hears voice clear but P1 could not hear any voice.
My sip.conf is
[avaya]
type=peer
fromdomain=xx.xx.xx.xx
host=xx.xx.xx.xx
disallow=all
allow=ulaw
dtmfmode=rfc2833
canreinvite=yes
--
Regards,
Shariq Khan
0333-3501125
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Lyle Giese
2011-Apr-08 01:38 UTC
[asterisk-users] Asterisk Avaya SIP Trunking One Way Audio
On 04/07/11 03:00, Shariq Khan wrote:> I am facing one way audio problem in sip trunking between asterisk and > avaya. > > +-------------+ +----+ > | avaya sip |-------| P1 | > +-------------+ +----+ > | > | > | > +-------------+ > | Asterisk | WAN > ------------------------------------------------- > | | LAN > +-------------+ > | > / > +----+ / > | P2 |--+ > +----+ > > When P1 dial P2, P2 hears voice clear but P1 could not hear any voice. > > My sip.conf is > > [avaya] > type=peer > fromdomain=xx.xx.xx.xx > host=xx.xx.xx.xx > disallow=all > allow=ulaw > dtmfmode=rfc2833 > canreinvite=yes > > > -- > Regards, > Shariq Khan > 0333-3501125 > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersTurn off reinvite on all extensions and SIP trunks involved and try again. Lyle Giese LCR Computer Services, Inc.
Shariq Khan
2011-Apr-08 10:05 UTC
[asterisk-users] Asterisk Avaya SIP Trunking One Way Audio
Yes, disabling reinvites solved the problem :) Thanks. -- Regards, Shariq Khan 0333-3501125 On Fri, Apr 8, 2011 at 6:38 AM, Lyle Giese <lyle at lcrcomputer.net> wrote:> On 04/07/11 03:00, Shariq Khan wrote: > >> I am facing one way audio problem in sip trunking between asterisk and >> avaya. >> >> +-------------+ +----+ >> | avaya sip |-------| P1 | >> +-------------+ +----+ >> | >> | >> | >> +-------------+ >> | Asterisk | WAN >> ------------------------------------------------- >> | | LAN >> +-------------+ >> | >> / >> +----+ / >> | P2 |--+ >> +----+ >> >> When P1 dial P2, P2 hears voice clear but P1 could not hear any voice. >> >> My sip.conf is >> >> [avaya] >> type=peer >> fromdomain=xx.xx.xx.xx >> host=xx.xx.xx.xx >> disallow=all >> allow=ulaw >> dtmfmode=rfc2833 >> canreinvite=yes >> >> >> -- >> Regards, >> Shariq Khan >> 0333-3501125 >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > Turn off reinvite on all extensions and SIP trunks involved and try again. > > Lyle Giese > LCR Computer Services, Inc. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110408/32cb4599/attachment.htm>