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@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@sip.mtt.ru;transport=udp>;tag=7USXQ5jg6caFB. To: <sip:89093848124@sip.mtt.ru>. Call-ID: 9ea5c38c-f2a0-122f-fd98-00163e000f5d. CSeq: 26087318 INVITE. Contact: <sip:100002000696@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@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@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@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org