[Freeswitch-users] Problem with outbound call thru ITSP (MERA MVTS3G v.4.4.0-20)
Anton Kvashenkin
anton.jugatsu at gmail.com
Tue Mar 27 16:05:42 MSD 2012
Big thanks, Jay. Now it works as aspected ;)
originate {sip_cid_type=none}sofia/gateway/mtt.ru/XXXX&txfax(/home/kvashenkin/txfax-sample.tiff)
U 2012/03/27 11:47:03.635333 62.76.180.83:5080 -> 80.75.130.134:5060
INVITE sip:89093848124 at sip.mtt.ru SIP/2.0.
Via: SIP/2.0/UDP 62.76.180.83:5080;rport;branch=z9hG4bK79pvrF70rHvHN.
Max-Forwards: 70.
From: "" <sip:100002000696 at sip.mtt.ru;transport=udp>;tag=NB3BQ3tK7germ.
To: <sip:89093848124 at sip.mtt.ru>.
Call-ID: 6997f9d4-f2a5-122f-fd98-00163e000f5d.
CSeq: 26088347 INVITE.
Contact: <sip:100002000696 at 62.76.180.83:5080;transport=udp;gw=mtt.ru>.
User-Agent: IP-PBX.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 205.
X-FS-Support: update_display,send_info.
.
v=0.
o=FreeSWITCH 1332829045 1332829046 IN IP4 62.76.180.83.
s=FreeSWITCH.
c=IN IP4 62.76.180.83.
t=0 0.
m=audio 19778 RTP/AVP 0 8 3 101 13.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
I disable Remote-party-id by providing {sip_cid_type=none} variable. The
last thing is added _this_ config param (what param) to mtt.xlm gateway.
27 марта 2012 г. 15:33 пользователь jay binks <jaybinks at gmail.com> написал:
> they may also be rejecting based on your caller id..
> seen that before
>
> From: <sip:100002000696 at sip.mtt.ru;transport=udp>;tag=B36y48m258Uga
> and
> Remote-Party-ID: <sip:0000000000 at sip.mtt.ru
> >;party=calling;screen=yes;privacy=off.
>
> dont look like its legit caller id ...
>
>
> Jay
>
>
>
> On 27 March 2012 21:17, Anton Kvashenkin <anton.jugatsu at gmail.com> wrote:
>
>> Thanks for the reply. Here
>>
>> sofia status profile external http://pastebin.freeswitch.org/18754
>>
>> Hm... You are right... WTF with codecs @ SDP.
>>
>> originate {absolute_codec_string='PCMU,PCMA,G729'}sofia/gateway/
>> mtt.ru/89093848124 &playback()
>>
>> and INVITE i have
>>
>> U 2012/03/27 11:12:45.244216 62.76.180.83:5080 -> 80.75.130.134:5060
>> INVITE sip:89093848124 at sip.mtt.ru SIP/2.0.
>> Via: SIP/2.0/UDP 62.76.180.83:5080;rport;branch=z9hG4bKy0HX29D6rc86K.
>> Max-Forwards: 70.
>> From: "" <sip:100002000696 at sip.mtt.ru;transport=udp>;tag=7USXQ5jg6caFB.
>> To: <sip:89093848124 at sip.mtt.ru>.
>> Call-ID: 9ea5c38c-f2a0-122f-fd98-00163e000f5d.
>> CSeq: 26087318 INVITE.
>> Contact: <sip:100002000696 at 62.76.180.83:5080;transport=udp;gw=mtt.ru>.
>> User-Agent: IP-PBX.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, precondition, path, replaces.
>> Allow-Events: talk, hold, refer.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 201.
>> X-FS-Support: update_display,send_info.
>> Remote-Party-ID: <sip:0000000000 at sip.mtt.ru
>> >;party=calling;screen=yes;privacy=off.
>> .
>> v=0.
>> o=FreeSWITCH 1332822661 1332822662 IN IP4 62.76.180.83.
>> s=FreeSWITCH.
>> c=IN IP4 62.76.180.83.
>> t=0 0.
>> m=audio 24104 RTP/AVP 0 101 13.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:20.
>>
>>
>> The same... I'm little bit frustrating :)
>>
>> 27 марта 2012 г. 14:26 пользователь jay binks <jaybinks at gmail.com>написал:
>>
>> your not getting very far at all..
>>>
>>> try enabling more codecs... there is something in your invite ( or SDP )
>>> that they dont like.
>>> but their logs tell you nothing more than the pcap you provided.
>>>
>>> I can assure you MERA DOES work... I send millions of calls to MERA
>>> servers every day from my freeswitch boxes.
>>> so this is something specific to your setup.
>>>
>>> Id also suggest the ITSP should be able to give you more info and be a
>>> little more helpful.
>>>
>>> Jay
>>>
>>> On 27 March 2012 20:07, Anton Kvashenkin <anton.jugatsu at gmail.com>wrote:
>>>
>>>> Hello guys. I have setup the gateway for outbound calls
>>>> with credentials provided by my ITSP.
>>>>
>>>> http://pastebin.freeswitch.org/18749
>>>>
>>>> Trying to make test call using, for example, originate command
>>>>
>>>> originate sofia/gateway/mtt.ru/XXXX &txfax(/tmp/fax.tif)
>>>>
>>>> result:
>>>>
>>>> ITSP proxy sends 500 Gateway is Invalid
>>>> http://pastebin.freeswitch.org/18750
>>>>
>>>> But, trying to make call using asterisk with the same credentials
>>>> suprisenly works. Here dump of success call
>>>> http://pastebin.freeswitch.org/18751
>>>>
>>>> I also attach logs that provided mys ITSP. Don't mind that there is ip
>>>> address from privite LAN, the first try was from behind NAT. The second,
>>>> that i provided above (http://pastebin.freeswitch.org/18750), was from
>>>> vps with staticly assigned address. For the first glance I thought that the
>>>> root of this problem was NAT but IMO it's not.
>>>>
>>>> Also, my external.xml
>>>>
>>>> http://pastebin.freeswitch.org/18752
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Sincerely
>>>
>>> Jay
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Sincerely
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/bf8110d0/attachment.html
Join us at ClueCon 2011 Aug 9-11, 2011
More information about the FreeSWITCH-users
mailing list