[asterisk-users] IAX2 bridge failing

Steven Davis stdavis at multiservice.com
Sat Dec 14 23:41:26 CST 2013


Did you change your network switch recently?  Some Digium IAX ATAs do not
behave well with Cisco equipment.


On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis <mdupuis at ocg.ca> wrote:

> meant to say restart didn't help either..
>
> ________________________________________
> From: asterisk-users-bounces at lists.digium.com [
> asterisk-users-bounces at lists.digium.com] On Behalf Of Michelle Dupuis [
> mdupuis at ocg.ca]
> Sent: Saturday, December 14, 2013 11:20 PM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] IAX2 bridge failing
>
> Ok just restart
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] On Behalf Of Michelle Dupuis
> Sent: Friday, December 13, 2013 11:46 AM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] IAX2 bridge failing
>
> I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload"
> inbetween)....same result
>
> I agree it sounds like something either end is using the wrong IP/port
> address somewhere in the call (yet signalling works fine).
>
> Anything else to suggest?  I was hoping for an externalip type setting but
> not in iax2 (at least not in 1.4.x.x)
> ________________________________________
> From: asterisk-users-bounces at lists.digium.com [
> asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp [
> jcolp at digium.com]
> Sent: Friday, December 13, 2013 11:44 AM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] IAX2 bridge failing
>
> Michelle Dupuis wrote:
> > Some more details...I noticed that the call is bridged, and audio goes
> > one way. However, the dial command still times out after 35 seconds
> > (approx), and exists non-zero.
> > While the channels are up, I did an core show channel xxx and found
> > Blocking in:
> > ast_waitfor_nandfds
> > Is this a bug? Or something I can fix through config?
>
> Hola,
>
> Set "transfer=no" under the entries in iax.conf for the
> peers/users/friends/etc in question, reload, retry, and see if that changes
> the behavior. If it does then something involved may not like
> IAX2 native transfers.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
> www.digium.com  & www.asterisk.org
>
> --
> _____________________________________________________________________
> -- 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
> --
> _____________________________________________________________________
> -- 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
>
> --
> _____________________________________________________________________
> -- 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
> --
> _____________________________________________________________________
> -- 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
>



-- 
*Steven Davis*
VoIP Engineer
Multi Service

+1-913-663-9748 o
+1-913-871-5155 m

stdavis at multiservice.com

<http://www.multiservice.com/>

-- 


------------------------------------------------------------------
This email is intended solely for the use of the addressee and may
contain information that is confidential, proprietary, or both.
If you receive this email in error please immediately notify the
sender and delete the email..
------------------------------------------------------------------

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131214/fe75f13a/attachment.html>


More information about the asterisk-users mailing list