On Tue, Feb 26, 2008 at 12:31 PM, Dirk Enrique Seiffert <[EMAIL PROTECTED]> wrote: > Hello, > > I am trying to add a sip-trunk to my Asterisk 1.4.15/Elastix 0.9.2 server. > The system is in production with local extensions, a zap trunk and a > working sip trunk with sipgate.de. > > My asterisk server is behind a NAT/Firewall, anyhow it registers and works > well with sipgate.de on incoming and outgoing calls. > > I aquired an account with a reseller net-voz.com: I did some testing with > the account directly from a Snom300 phone - works without a problem, > (behind the nat) I spent hours testing and adjusting the trunk > configuration for net-voz, maybe sombody on the list can give a helpful hint: > > First of all: Registry works! > > pbx*CLI> sip show registry > Host Username Refresh State > Reg.Time > sip.net-voz.com:5060 xxxxxx6168 585 Registered > Tue, 26 Feb 2008 10:47:58 > sipgate.de:5060 xxxx0823 105 Registered > Tue, 26 Feb 2008 10:56:22 > > This is my config: > > [ringtime] > username=5515816168 > type=peer > type=friend > secret=118873 > insecure=very > host=sip.net-voz.com > fromuser=5515816168 > fromdomain=sip.net-voz.com > canreinvite=no > call-limit=50 > > I tried faking the user agent (without success) > > useragent = Grandstream BT100 1.0.4.49 > externip=xx.xx.116.229 > localnet=192.168.8.0/255.255.255.0 > > On my gateway I can see the following with tcpdump: > > listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes > 11:05:57.386827 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: > 810 > 11:05:57.452414 IP 190.144.151.212.sip > pbx.lintec.sip: SIP, length: 442 > 11:05:57.453021 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: > 385 > 11:05:57.453587 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030 > 11:05:58.452868 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030 > 11:06:01.453814 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030 > > On the astersik CLI the logs show: > > Audio is at 192.168.8.3 port 14800 > Adding codec 0x4 (ulaw) to SDP > Adding codec 0x8 (alaw) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (no NAT) to 190.144.151.212:5060: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport > From: "901" <sip:[EMAIL PROTECTED]>;tag=as3c6dfee5 > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch", > algorithm=MD5, uri="sip:[EMAIL PROTECTED]", > nonce="120404195526111105702055508208", > response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque="" > Date: Tue, 26 Feb 2008 16:09:09 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 260 > > v=0 > o=root 2381 2382 IN IP4 192.168.8.3 > s=session > c=IN IP4 192.168.8.3 > t=0 0 > m=audio 14800 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > --- > Retransmitting #1 (no NAT) to 190.144.151.212:5060: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport > From: "901" <sip:[EMAIL PROTECTED]>;tag=as3c6dfee5 > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch", > algorithm=MD5, uri="sip:[EMAIL PROTECTED]", > nonce="120404195526111105702055508208", > response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque="" > Date: Tue, 26 Feb 2008 16:09:09 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 260 > > > It looks like the comuunication starts but then gets lost.?? > > Any idea is appreciated. > > Thanks > > Enrique > > > > Cartagena - Colombia > http://www.sipcolombia.com
Does it retransmit the invite six times and then hangup? When I have seen this it was a firewall issue on the remote (provider) side. Thanks, Steve Totaro _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
