[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
>>>
>>> --
>>> _____________________________________________________________________
>>> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130515/ce1e9f1b/attachment.htm>
More information about the asterisk-users
mailing list