[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 15:17:18 MSD 2012


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
>>
>> 
>> 
>>
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>> http://www.freeswitch.org
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>>
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>>
>>
>
>
> --
> Sincerely
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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> http://wiki.freeswitch.org
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>
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