[asterisk-users] Cut offs on outgoing SIP calls

[Asghar Mohammad]

asterisk is behind nat?


On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk <earohuanca at gmail.com>wrote:

> Hello everyone,
>
> I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
> calls are working fine, but outgoing ones show the gollowing messages when
> are being dropped:
>
> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
> Retransmission timeout reached on transmission
> ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical
> Response) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 6399ms with no response
> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt: Hanging
> up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to our
> critical packet (see
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
> This is happening with my PBX hosted on an external network and peers on
> my local network.
>
> It seems the SIP ACK is not being received properly.
>
> I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9
>
> Elder D. Arohuanca
> Lima - Peru
>
> --
> _____________________________________________________________________
> -- 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/20130515/2c24b46f/attachment.htm>