[asterisk-users] Cut offs on outgoing SIP calls

[Daniel - Asterisk]

Users (softphones) are behind a NAT, Asterisk has its own public ip address


On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad <asghar144 at gmail.com>wrote:

> 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
>>
>> --
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>
>
> --
> _____________________________________________________________________
> -- 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
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