[asterisk-users] Cut offs on outgoing SIP calls

[Daniel - Asterisk]

Current configuration follows:

[general]
context=default
allowguest=no
alwaysauthreject=yes
allowoverlap=yes
allowtransfer=yes
tcpenable=no
tlsenable=no
srvlookup=yes
vmexten=vm
rtcachefriends=yes
nat=no
directmedia=nonat
directrtpsetup=no
videosupport=yes
maxcallbitrate=384
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
allow=ilbc
allow=speex
allow=g726
allow=g723
mohinterpret=default
mohsuggest=default
dtmfmode=rfc2833
timer1b=60000
transport=udp

[carrier-1]
host=a.b.c.d
type=friend
context=from-pstn
disallow=all
allow=ulaw,alaw
qualify=yes
trunk=yes

[90102]
secret=xxxxxx
mailbox=90102 at default
cid_number=NXXXXXXXXX
accountcode=401
type=friend
host=dynamic
port=5060
qualify=yes
nat=yes
transport=udp
context=users
disallow=all
allow=ulaw,alaw,g729,gsm,speex,ilbc,h264,h263p,h263
directmedia=no
canreinvite=no
videosupport=no




On Wed, May 15, 2013 at 2:47 PM, Asghar Mohammad <asghar144 at gmail.com>wrote:

> 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
>>>
>>> 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
>
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