[asterisk-users] Cut offs on outgoing SIP calls

Asghar Mohammad asghar144 at gmail.com
Wed May 15 14:47:56 CDT 2013


please show us peer configuration.


On Wed, May 15, 2013 at 9:46 PM, Daniel - Asterisk <earohuanca at gmail.com>wrote:

> 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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>>
>
>
> --
> _____________________________________________________________________
> -- 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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>    http://lists.digium.com/mailman/listinfo/asterisk-users
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