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 _______________________________________________ -- 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
