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:5756646022 at sip.net-voz.com SIP/2.0 Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport From: "901" <sip:5515816168 at sip.net-voz.com>;tag=as3c6dfee5 To: <sip:5756646022 at sip.net-voz.com> Contact: <sip:5515816168 at 192.168.8.3> Call-ID: 5fc995c93d10f2a73186133377cafc88 at sip.net-voz.com CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch", algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com", 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:5756646022 at sip.net-voz.com SIP/2.0 Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport From: "901" <sip:5515816168 at sip.net-voz.com>;tag=as3c6dfee5 To: <sip:xxxxxx6022 at sip.net-voz.com> Contact: <sip:xxxxx6168 at 192.168.8.3> Call-ID: 5fc995c93d10f2a73186133377cafc88 at sip.net-voz.com CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch", algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com", 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
On Tue, Feb 26, 2008 at 12:31 PM, Dirk Enrique Seiffert <ds at caribenet.com> 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:5756646022 at sip.net-voz.com SIP/2.0 > Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport > From: "901" <sip:5515816168 at sip.net-voz.com>;tag=as3c6dfee5 > To: <sip:5756646022 at sip.net-voz.com> > Contact: <sip:5515816168 at 192.168.8.3> > Call-ID: 5fc995c93d10f2a73186133377cafc88 at sip.net-voz.com > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch", > algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com", > 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:5756646022 at sip.net-voz.com SIP/2.0 > Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport > From: "901" <sip:5515816168 at sip.net-voz.com>;tag=as3c6dfee5 > To: <sip:xxxxxx6022 at sip.net-voz.com> > Contact: <sip:xxxxx6168 at 192.168.8.3> > Call-ID: 5fc995c93d10f2a73186133377cafc88 at sip.net-voz.com > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch", > algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com", > 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.comDoes 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
Hi Steve,> 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. >Indeed it tries seven times. But I think this is the Asterisk default. The same account configured in my Snom Phone works without problem, - from same network to same network. Thanks al lot Enrique Cartagena - Colombia http://www.sipcolombia.com
On Tue, 2008-02-26 at 12:31 -0500, Dirk Enrique Seiffert wrote:> 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:I'll take a stab at it.> First of all: Registry works!Registering to another host doesn't mean anything when it comes to sending them a call. Registration only tells them your IP address and port so that they can send calls *to you*.> 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:5756646022 at sip.net-voz.com SIP/2.0 > Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport > From: "901" <sip:5515816168 at sip.net-voz.com>;tag=as3c6dfee5 > To: <sip:5756646022 at sip.net-voz.com> > Contact: <sip:5515816168 at 192.168.8.3> > Call-ID: 5fc995c93d10f2a73186133377cafc88 at sip.net-voz.com > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch", > algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com", > 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 > > ---Notice how the Contact Header and the SDP all have the IP address of 192.168.8.3? If your firewall isn't masquerading (rewriting) those addresses as the SIP traffic goes through it, then the device on the other end is going to try to contact 192.168.8.3, and I'm guessing it's going to have a hard time doing that. (This would also explain why you're seeing outbound traffic only in your tcpdump traces.) -- Jared Smith Community Relations Manager Digium, Inc.