One way Audio switch to IAX2 did not fix problem on asterisk

...they both give me the same problems!

ONE WAY CALLING! I can recieve calls but not place any I get a busy signal on IAX2 or a all circuits are busy now on SIP.

Here is a same of the debug

IAX..................................................>

Code:

[Apr  4 05:19:25]    FORMAT          : 4 [Apr  4 05:19:25] [Apr  4 05:19:25] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX     Subclass: ACK [Apr  4 05:19:25]    Timestamp: 00010ms  SCall: 00001  DCall: 00342 [64.34.164.254:4569] [Apr  4 05:19:35] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX     Subclass: LAGRQ [Apr  4 05:19:35]    Timestamp: 10003ms  SCall: 00001  DCall: 00342 [64.34.164.254:4569] [Apr  4 05:19:35] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass: LAGRP [Apr  4 05:19:35]    Timestamp: 10003ms  SCall: 00342  DCall: 00001 [64.34.164.254:4569] [Apr  4 05:19:35] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass: ACK [Apr  4 05:19:35]    Timestamp: 10003ms  SCall: 00001  DCall: 00342 [64.34.164.254:4569] [Apr  4 05:19:35] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass: LAGRQ [Apr  4 05:19:35]    Timestamp: 10010ms  SCall: 00342  DCall: 00001 [64.34.164.254:4569] [Apr  4 05:19:35] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 003 Type: IAX     Subclass: LAGRP [Apr  4 05:19:35]    Timestamp: 10010ms  SCall: 00001  DCall: 00342 [64.34.164.254:4569] [Apr  4 05:19:35] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX     Subclass: ACK [Apr  4 05:19:35]    Timestamp: 10010ms  SCall: 00342  DCall: 00001 [64.34.164.254:4569] [Apr  4 05:19:36] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX     Subclass: HANGUP [Apr  4 05:19:36]    Timestamp: 10553ms  SCall: 00342  DCall: 00001 [64.34.164.254:4569] [Apr  4 05:19:36]    CAUSE CODE      : 17 [Apr  4 05:19:36] [Apr  4 05:19:36] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX     Subclass: ACK [Apr  4 05:19:36]    Timestamp: 10553ms  SCall: 00001  DCall: 00342 [64.34.164.254:4569] [Apr  4 05:19:47] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: REGREQ [Apr  4 05:19:47]    Timestamp: 00006ms  SCall: 00002  DCall: 00000 [64.34.164.254:4569] [Apr  4 05:19:47]    USERNAME        : 1153195240 [Apr  4 05:19:47]    REFRESH         : 60 </code> For sip.....................................> <code> --- Retransmitting #3 (NAT) to 216.211.27.163:5060: REGISTER sip:<HOSTNAME OF SIP PROVIDER> SIP/2.0 Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK176d4ff5;rport From: <sip:DID_Phone_1@<HOSTNAME OF SIP PROVIDER>>;tag=as225e6cbd To: <sip:DID_Phone_1@<HOSTNAME OF SIP PROVIDER>> Call-ID: 0ac3c85c1181b12a316b078c04edddcc@<WAN IP OF ROUTER> CSeq: 187 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:DID_Phone_1@<Internal IP address of ASTERISK Server>> Event: registration Content-Length: 0 --- Reliably Transmitting (NAT) to <Internal IP of VOIP ADAPER_1>:5060: OPTIONS sip:133@<Internal IP of VOIP ADAPER_1>:5060 SIP/2.0 Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK651a5e19;rport From: "Unknown" <sip:Unknown@<WAN IP OF ROUTER>>;tag=as42c741d5 To: <sip:133@<Internal IP of VOIP ADAPER_1>:5060> Contact: <sip:Unknown@<Internal IP address of ASTERISK Server>> Call-ID: 3ab44aab6fe9740d2b79def64b39d13d@<WAN IP OF ROUTER> CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 Apr 2008 01:58:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Reliably Transmitting (NAT) to <Internal IP of VOIP ADAPER_1>:5061: OPTIONS sip:601@<Internal IP of VOIP ADAPER_1>:5061 SIP/2.0 Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK6b77ec9a;rport From: "Unknown" <sip:Unknown@<WAN IP OF ROUTER>>;tag=as2134deae To: <sip:601@<Internal IP of VOIP ADAPER_1>:5061> Contact: <sip:Unknown@<Internal IP address of ASTERISK Server>> Call-ID: 47cc68a1068406434a73def83d94e5e0@<WAN IP OF ROUTER> CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 Apr 2008 01:58:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Reliably Transmitting (NAT) to <Internal IP address of Voip Adapter_2>:5060: OPTIONS sip:121@<Internal IP address of Voip Adapter_2>:5060 SIP/2.0 Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK58d88a9e;rport From: "Unknown" <sip:Unknown@<WAN IP OF ROUTER>>;tag=as36bc5498 To: <sip:121@<Internal IP address of Voip Adapter_2>:5060> Contact: <sip:Unknown@<Internal IP address of ASTERISK Server>> Call-ID: 625a4fdc6f57d76d553f4c2f04de8e2b@<WAN IP OF ROUTER> CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 Apr 2008 01:58:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Reliably Transmitting (NAT) to <Internal IP address of Voip Adapter_2>:5060: OPTIONS sip:113@<Internal IP address of Voip Adapter_2>:5060 SIP/2.0 Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK525380b3;rport From: "Unknown" <sip:Unknown@<WAN IP OF ROUTER>>;tag=as71653d66 To: <sip:113@<Internal IP address of Voip Adapter_2>:5060> Contact: <sip:Unknown@<Internal IP address of ASTERISK Server>> Call-ID: 28424aba449648442fed01670bd3c63c@<WAN IP OF ROUTER> CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 Apr 2008 01:58:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- voip*CLI> <--- SIP read from <Internal IP of VOIP ADAPER_1>:5060 ---> SIP/2.0 200 OK To: <sip:133@<Internal IP of VOIP ADAPER_1>:5060>;tag=8ea407de1526a2d6i0 From: "Unknown" <sip:Unknown@<WAN IP OF ROUTER>>;tag=as42c741d5 Call-ID: 3ab44aab6fe9740d2b79def64b39d13d@<WAN IP OF ROUTER> CSeq: 102 OPTIONS Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK651a5e19 Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura <-------------> --- (10 headers 0 lines) --- voip*CLI> <--- SIP read from <Internal IP of VOIP ADAPER_1>:5061 ---> SIP/2.0 200 OK To: <sip:601@<Internal IP of VOIP ADAPER_1>:5061>;tag=aa4f9f9e2cc9ee16i1 From: "Unknown" <sip:Unknown@<WAN IP OF ROUTER>>;tag=as2134deae Call-ID: 47cc68a1068406434a73def83d94e5e0@<WAN IP OF ROUTER> CSeq: 102 OPTIONS Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK6b77ec9a Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura <-------------> --- (10 headers 0 lines) --- voip*CLI> <--- SIP read from <Internal IP address of Voip Adapter_2>:5060 ---> SIP/2.0 200 OK To: <sip:121@<Internal IP address of Voip Adapter_2>:5060>;tag=66f1e13a96b53913i1 From: "Unknown" <sip:Unknown@<WAN IP OF ROUTER>>;tag=as36bc5498 Call-ID: 625a4fdc6f57d76d553f4c2f04de8e2b@<WAN IP OF ROUTER> CSeq: 102 OPTIONS Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK58d88a9e Server: 192.168.4.1 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '625a4fdc6f57d76d553f4c2f04de8e2b@<WAN IP OF ROUTER>' Method: OPTIONS Really destroying SIP dialog '47cc68a1068406434a73def83d94e5e0@<WAN IP OF ROUTER>' Method: OPTIONS Really destroying SIP dialog '3ab44aab6fe9740d2b79def64b39d13d@<WAN IP OF ROUTER>' Method: OPTIONS voip*CLI> <--- SIP read from <Internal IP address of Voip Adapter_2>:5060 ---> SIP/2.0 200 OK To: <sip:113@<Internal IP address of Voip Adapter_2>:5060>;tag=df9cfaf263322abbi0 From: "Unknown" <sip:Unknown@<WAN IP OF ROUTER>>;tag=as71653d66 Call-ID: 28424aba449648442fed01670bd3c63c@<WAN IP OF ROUTER> CSeq: 102 OPTIONS Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK525380b3 Server: 192.168.4.1 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '28424aba449648442fed01670bd3c63c@<WAN IP OF ROUTER>' Method: OPTIONS Retransmitting #5 (NAT) to 209.197.130.18:5060: REGISTER sip:<Sip provider_2> SIP/2.0 Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK78f92b1e;rport From: <sip:<DID_number_3>@<Sip provider_2>>;tag=as6518f6d9 To: <sip:<DID_number_3>@<Sip provider_2>> Call-ID: 074ef09a4f625b581652d3fa3e7a1b21@<WAN IP OF ROUTER> CSeq: 187 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_number_3>@<Internal IP address of ASTERISK Server>> Event: registration Content-Length: 0 --- Retransmitting #4 (NAT) to 216.211.27.163:5060: REGISTER sip:<HOSTNAME OF SIP PROVIDER> SIP/2.0 Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK45f1d139;rport From: <sip:<DID_number_2>@<HOSTNAME OF SIP PROVIDER>>;tag=as0d020145 To: <sip:<DID_number_2>@<HOSTNAME OF SIP PROVIDER>> Call-ID: 0c553d677136a1616b9fa58811079b49@<WAN IP OF ROUTER> CSeq: 187 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_number_2>@<Internal IP address of ASTERISK Server>> Event: registration Content-Length: 0 --- Retransmitting #4 (NAT) to 216.211.27.163:5060: REGISTER sip:<HOSTNAME OF SIP PROVIDER> SIP/2.0 Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK176d4ff5;rport From: <sip:DID_Phone_1@<HOSTNAME OF SIP PROVIDER>>;tag=as225e6cbd To: <sip:DID_Phone_1@<HOSTNAME OF SIP PROVIDER>> Call-ID: 0ac3c85c1181b12a316b078c04edddcc@<WAN IP OF ROUTER> CSeq: 187 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:DID_Phone_1@<Internal IP address of ASTERISK Server>> Event: registration Content-Length: 0 --- Retransmitting #6 (NAT) to 209.197.130.18:5060: REGISTER sip:<Sip provider_2> SIP/2.0 Via: SIP/2.0/UDP <Internal IP address of ASTERISK Server>:5060;branch=z9hG4bK78f92b1e;rport From: <sip:<DID_number_3>@<Sip provider_2>>;tag=as6518f6d9 To: <sip:<DID_number_3>@<Sip provider_2>> Call-ID: 074ef09a4f625b581652d3fa3e7a1b21@<WAN IP OF ROUTER> CSeq: 187 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_number_3>@<Internal IP address of ASTERISK Server>> Event: registration Content-Length: 0 </code> Another sample of Debug code during my journey... <code> --- Retransmitting #6 (NAT) to 209.197.130.18:5060: REGISTER sip:<Sip provider_1> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK3f46d5b6;rport From: <sip:<DID_1>@<Sip provider_1>>;tag=as72b0b741 To: <sip:<DID_1>@<Sip provider_1>> Call-ID: 074ef09a4f625b581652d3fa3e7a1b21@<WAN IP OF ROUTER> CSeq: 151 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_1>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #6 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK28a8b80a;rport From: <sip:<DID_2>@<Sip Provider_2>>;tag=as7d6f5686 To: <sip:<DID_2>@<Sip Provider_2>> Call-ID: 0c553d677136a1616b9fa58811079b49@<WAN IP OF ROUTER> CSeq: 151 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_2>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #6 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK5822d0e9;rport From: <sip:<DID_3>@<Sip Provider_2>>;tag=as53385fd3 To: <sip:<DID_3>@<Sip Provider_2>> Call-ID: 0ac3c85c1181b12a316b078c04edddcc@<WAN IP OF ROUTER> CSeq: 151 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_3>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- REGISTER 12 headers, 0 lines Reliably Transmitting (NAT) to 209.197.130.18:5060: REGISTER sip:<Sip provider_1> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK3fc15c21;rport From: <sip:<DID_1>@<Sip provider_1>>;tag=as60ab06be To: <sip:<DID_1>@<Sip provider_1>> Call-ID: 074ef09a4f625b581652d3fa3e7a1b21@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_1>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Really destroying SIP dialog '074ef09a4f625b581652d3fa3e7a1b21@<WAN IP OF ROUTER>' Method: REGISTER REGISTER 12 headers, 0 lines Reliably Transmitting (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK152285ad;rport From: <sip:<DID_2>@<Sip Provider_2>>;tag=as10122dfe To: <sip:<DID_2>@<Sip Provider_2>> Call-ID: 0c553d677136a1616b9fa58811079b49@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_2>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Really destroying SIP dialog '0c553d677136a1616b9fa58811079b49@<WAN IP OF ROUTER>' Method: REGISTER REGISTER 12 headers, 0 lines Reliably Transmitting (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK10018e69;rport From: <sip:<DID_3>@<Sip Provider_2>>;tag=as19d1fe3f To: <sip:<DID_3>@<Sip Provider_2>> Call-ID: 0ac3c85c1181b12a316b078c04edddcc@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_3>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Really destroying SIP dialog '0ac3c85c1181b12a316b078c04edddcc@<WAN IP OF ROUTER>' Method: REGISTER Retransmitting #1 (NAT) to 209.197.130.18:5060: REGISTER sip:<Sip provider_1> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK3fc15c21;rport From: <sip:<DID_1>@<Sip provider_1>>;tag=as60ab06be To: <sip:<DID_1>@<Sip provider_1>> Call-ID: 074ef09a4f625b581652d3fa3e7a1b21@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_1>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #1 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK152285ad;rport From: <sip:<DID_2>@<Sip Provider_2>>;tag=as10122dfe To: <sip:<DID_2>@<Sip Provider_2>> Call-ID: 0c553d677136a1616b9fa58811079b49@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_2>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #1 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK10018e69;rport From: <sip:<DID_3>@<Sip Provider_2>>;tag=as19d1fe3f To: <sip:<DID_3>@<Sip Provider_2>> Call-ID: 0ac3c85c1181b12a316b078c04edddcc@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_3>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- voip*CLI> <--- SIP read from 64.34.164.254:5060 ---> INVITE sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP> SIP/2.0 Via: SIP/2.0/UDP 64.34.164.254:5060;branch=z9hG4bK0ffa7ea0;rport From: "8070010000" <sip:8070010000@64.34.164.254>;tag=as4016c124 To: <sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP>> Contact: <sip:8070010000@64.34.164.254> Call-ID: 429c00574457880a2010341b066c2ce6 [!at] 64.34.164.254 (replace the [!at] with a @) CSeq: 102 INVITE User-Agent: Voice Network Inc 1b Max-Forwards: 70 Remote-Party-ID: "8070010000" <sip:8070010000@64.34.164.254>;privacy=off;screen=no Date: Fri, 04 Apr 2008 01:46:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 343 v=0 o=root 17640 17640 IN IP4 64.34.164.254 s=session c=IN IP4 64.34.164.254 t=0 0 m=audio 17590 RTP/AVP 0 18 111 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 16 lines) --- Sending to 64.34.164.254 : 5060 (NAT) Using INVITE request as basis request - 429c00574457880a2010341b066c2ce6 [!at] 64.34.164.254 (replace the [!at] with a @) Found peer '<USER PROVIDER 2>' <--- Reliably Transmitting (NAT) to 64.34.164.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.34.164.254:5060;branch=z9hG4bK0ffa7ea0;received=64.34.164.254;rport=5060 From: "8070010000" <sip:8070010000@64.34.164.254>;tag=as4016c124 To: <sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP>>;tag=as4d298fc7 Call-ID: 429c00574457880a2010341b066c2ce6 [!at] 64.34.164.254 (replace the [!at] with a @) CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="595feed7" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '429c00574457880a2010341b066c2ce6@64.34.164.254' in 6656 ms (Method: INVITE) voip*CLI> <--- SIP read from 64.34.164.254:5060 ---> ACK sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP> SIP/2.0 Via: SIP/2.0/UDP 64.34.164.254:5060;branch=z9hG4bK0ffa7ea0;rport From: "8070010000" <sip:8070010000@64.34.164.254>;tag=as4016c124 To: <sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP>>;tag=as4d298fc7 Contact: <sip:8070010000@64.34.164.254> Call-ID: 429c00574457880a2010341b066c2ce6 [!at] 64.34.164.254 (replace the [!at] with a @) CSeq: 102 ACK User-Agent: Voice Network Inc 1b Max-Forwards: 70 Remote-Party-ID: "8070010000" <sip:8070010000@64.34.164.254>;privacy=off;screen=no Content-Length: 0 <-------------> --- (11 headers 0 lines) --- voip*CLI> <--- SIP read from 64.34.164.254:5060 ---> INVITE sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP> SIP/2.0 Via: SIP/2.0/UDP 64.34.164.254:5060;branch=z9hG4bK35b90849;rport From: "8070010000" <sip:8070010000@64.34.164.254>;tag=as4016c124 To: <sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP>> Contact: <sip:8070010000@64.34.164.254> Call-ID: 429c00574457880a2010341b066c2ce6 [!at] 64.34.164.254 (replace the [!at] with a @) CSeq: 103 INVITE User-Agent: Voice Network Inc 1b Max-Forwards: 70 Remote-Party-ID: "8070010000" <sip:8070010000@64.34.164.254>;privacy=off;screen=no Proxy-Authorization: Digest username="<USER PROVIDER 2>", realm="asterisk", algorithm=MD5, uri="sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP>", nonce="595feed7", response="bb3f4b24014a5d246630e9db3603615f", opaque="" Date: Fri, 04 Apr 2008 01:46:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 343 v=0 o=root 17640 17641 IN IP4 64.34.164.254 s=session c=IN IP4 64.34.164.254 t=0 0 m=audio 17590 RTP/AVP 0 18 111 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 16 lines) --- Sending to 64.34.164.254 : 5060 (NAT) Using INVITE request as basis request - 429c00574457880a2010341b066c2ce6 [!at] 64.34.164.254 (replace the [!at] with a @) Found peer '<USER PROVIDER 2>' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 111 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 64.34.164.254:17590 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 111 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x906 (gsm|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 64.34.164.254:17590 Looking for <DID_INCOMMING_MAIN> in from-trunk (domain <WAN ROUTER IP>) list_route: hop: <sip:8070010000@64.34.164.254> <--- Transmitting (NAT) to 64.34.164.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.34.164.254:5060;branch=z9hG4bK35b90849;received=64.34.164.254;rport=5060 From: "8070010000" <sip:8070010000@64.34.164.254>;tag=as4016c124 To: <sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP>> Call-ID: 429c00574457880a2010341b066c2ce6 [!at] 64.34.164.254 (replace the [!at] with a @) CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:<DID_INCOMMING_MAIN>@<internal IP of asterisk server>> Content-Length: 0 <------------>     -- Executing [<DID_INCOMMING_MAIN>@from-trunk:1] NoOp("SIP/<USER PROVIDER 2>-09491ea0", "Catch-All DID Match - Found <DID_INCOMMING_MAIN> - You probably want a DID for this.") in new stack     -- Executing [<DID_INCOMMING_MAIN>@from-trunk:2] Goto("SIP/<USER PROVIDER 2>-09491ea0", "ext-did|s|1") in new stack     -- Goto (ext-did,s,1)     -- Executing [s@ext-did:1] Set("SIP/<USER PROVIDER 2>-09491ea0", "__FROM_DID=s") in new stack     -- Executing [s@ext-did:2] Gosub("SIP/<USER PROVIDER 2>-09491ea0", "app-blacklist-check|s|1") in new stack     -- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/<USER PROVIDER 2>-09491ea0", "") in new stack     -- Executing [s@app-blacklist-check:2] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "0?blacklisted") in new stack     -- Executing [s@app-blacklist-check:3] Return("SIP/<USER PROVIDER 2>-09491ea0", "") in new stack     -- Executing [s@ext-did:3] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "1 ?cidok") in new stack     -- Goto (ext-did,s,5)     -- Executing [s@ext-did:5] NoOp("SIP/<USER PROVIDER 2>-09491ea0", "CallerID is "8070010000" <8070010000>") in new stack     -- Executing [s@ext-did:6] Ringing("SIP/<USER PROVIDER 2>-09491ea0", "") in new stack voip*CLI> <--- Transmitting (NAT) to 64.34.164.254:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 64.34.164.254:5060;branch=z9hG4bK35b90849;received=64.34.164.254;rport=5060 From: "8070010000" <sip:8070010000@64.34.164.254>;tag=as4016c124 To: <sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP>>;tag=as464393e0 Call-ID: 429c00574457880a2010341b066c2ce6 [!at] 64.34.164.254 (replace the [!at] with a @) CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:<DID_INCOMMING_MAIN>@<internal IP of asterisk server>> Content-Length: 0 <------------>     -- Executing [s@ext-did:7] Set("SIP/<USER PROVIDER 2>-09491ea0", "FAX_RX_EMAIL=eric@tbayrentals.com") in new stack     -- Executing [s@ext-did:8] Answer("SIP/<USER PROVIDER 2>-09491ea0", "") in new stack Audio is at <internal IP of asterisk server> port 15574 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP voip*CLI> <--- Reliably Transmitting (NAT) to 64.34.164.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.34.164.254:5060;branch=z9hG4bK35b90849;received=64.34.164.254;rport=5060 From: "8070010000" <sip:8070010000@64.34.164.254>;tag=as4016c124 To: <sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP>>;tag=as464393e0 Call-ID: 429c00574457880a2010341b066c2ce6 [!at] 64.34.164.254 (replace the [!at] with a @) CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:<DID_INCOMMING_MAIN>@<internal IP of asterisk server>> Content-Type: application/sdp Content-Length: 240 v=0 o=root 2618 2618 IN IP4 <internal IP of asterisk server> s=session c=IN IP4 <internal IP of asterisk server> t=0 0 m=audio 15574 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------>     -- Executing [s@ext-did:9] PlayTones("SIP/<USER PROVIDER 2>-09491ea0", "ring") in new stack     -- Executing [s@ext-did:10] NVFaxDetect("SIP/<USER PROVIDER 2>-09491ea0", "3") in new stack voip*CLI> <--- SIP read from 64.34.164.254:5060 ---> ACK sip:<DID_INCOMMING_MAIN>@<internal IP of asterisk server>:5060 SIP/2.0 Via: SIP/2.0/UDP 64.34.164.254:5060;branch=z9hG4bK09d90b46;rport From: "8070010000" <sip:8070010000@64.34.164.254>;tag=as4016c124 To: <sip:<DID_INCOMMING_MAIN>@<WAN ROUTER IP>>;tag=as464393e0 Contact: <sip:8070010000@64.34.164.254> Call-ID: 429c00574457880a2010341b066c2ce6 [!at] 64.34.164.254 (replace the [!at] with a @) CSeq: 103 ACK User-Agent: Voice Network Inc 1b Max-Forwards: 70 Remote-Party-ID: "8070010000" <sip:8070010000@64.34.164.254>;privacy=off;screen=no Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Retransmitting #2 (NAT) to 209.197.130.18:5060: REGISTER sip:<Sip provider_1> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK3fc15c21;rport From: <sip:<DID_1>@<Sip provider_1>>;tag=as60ab06be To: <sip:<DID_1>@<Sip provider_1>> Call-ID: 074ef09a4f625b581652d3fa3e7a1b21@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_1>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #2 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK152285ad;rport From: <sip:<DID_2>@<Sip Provider_2>>;tag=as10122dfe To: <sip:<DID_2>@<Sip Provider_2>> Call-ID: 0c553d677136a1616b9fa58811079b49@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_2>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #2 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK10018e69;rport From: <sip:<DID_3>@<Sip Provider_2>>;tag=as19d1fe3f To: <sip:<DID_3>@<Sip Provider_2>> Call-ID: 0ac3c85c1181b12a316b078c04edddcc@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_3>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- voip*CLI> <--- SIP read from 67.193.1.231:63855 ---> NOTIFY sip:voip.tbayrentals.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.20:5060;branch=z9hG4bK-e9d4e2e8 From: Madej <sip:905@voip.tbayrentals.com>;tag=9acf01767bb236d5o1 To: <sip:voip.tbayrentals.com> Call-ID: 3b4de902-fd3f94b9 [!at] 10.0.0.20 (replace the [!at] with a @) CSeq: 6884 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/PAP2-3.1.22(LS) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- Transmitting (NAT) to 67.193.1.231:63855 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 10.0.0.20:5060;branch=z9hG4bK-e9d4e2e8;received=67.193.1.231 From: Madej <sip:905@voip.tbayrentals.com>;tag=9acf01767bb236d5o1 To: <sip:voip.tbayrentals.com>;tag=as0140dafe Call-ID: 3b4de902-fd3f94b9 [!at] 10.0.0.20 (replace the [!at] with a @) CSeq: 6884 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------>     -- Executing [s@ext-did:11] Goto("SIP/<USER PROVIDER 2>-09491ea0", "ivr-2|s|1") in new stack     -- Goto (ivr-2,s,1)     -- Executing [s@ivr-2:1] Set("SIP/<USER PROVIDER 2>-09491ea0", "LOOPCOUNT=0") in new stack     -- Executing [s@ivr-2:2] Set("SIP/<USER PROVIDER 2>-09491ea0", "__DIR-CONTEXT=default") in new stack     -- Executing [s@ivr-2:3] Set("SIP/<USER PROVIDER 2>-09491ea0", "_IVR_CONTEXT_ivr-2=") in new stack     -- Executing [s@ivr-2:4] Set("SIP/<USER PROVIDER 2>-09491ea0", "_IVR_CONTEXT=ivr-2") in new stack     -- Executing [s@ivr-2:5] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "1?begin") in new stack     -- Goto (ivr-2,s,8)     -- Executing [s@ivr-2:8] Set("SIP/<USER PROVIDER 2>-09491ea0", "TIMEOUT(digit)=3") in new stack     -- Digit timeout set to 3     -- Executing [s@ivr-2:9] Set("SIP/<USER PROVIDER 2>-09491ea0", "TIMEOUT(response)=5") in new stack     -- Response timeout set to 5     -- Executing [s@ivr-2:10] BackGround("SIP/<USER PROVIDER 2>-09491ea0", "custom/001-08-F-Initial") in new stack     -- <SIP/<USER PROVIDER 2>-09491ea0> Playing 'custom/001-08-F-Initial' (language 'en') Retransmitting #3 (NAT) to 209.197.130.18:5060: REGISTER sip:<Sip provider_1> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK3fc15c21;rport From: <sip:<DID_1>@<Sip provider_1>>;tag=as60ab06be To: <sip:<DID_1>@<Sip provider_1>> Call-ID: 074ef09a4f625b581652d3fa3e7a1b21@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_1>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #3 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK152285ad;rport From: <sip:<DID_2>@<Sip Provider_2>>;tag=as10122dfe To: <sip:<DID_2>@<Sip Provider_2>> Call-ID: 0c553d677136a1616b9fa58811079b49@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_2>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #3 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK10018e69;rport From: <sip:<DID_3>@<Sip Provider_2>>;tag=as19d1fe3f To: <sip:<DID_3>@<Sip Provider_2>> Call-ID: 0ac3c85c1181b12a316b078c04edddcc@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_3>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #4 (NAT) to 209.197.130.18:5060: REGISTER sip:<Sip provider_1> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK3fc15c21;rport From: <sip:<DID_1>@<Sip provider_1>>;tag=as60ab06be To: <sip:<DID_1>@<Sip provider_1>> Call-ID: 074ef09a4f625b581652d3fa3e7a1b21@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_1>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #4 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK152285ad;rport From: <sip:<DID_2>@<Sip Provider_2>>;tag=as10122dfe To: <sip:<DID_2>@<Sip Provider_2>> Call-ID: 0c553d677136a1616b9fa58811079b49@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_2>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #4 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK10018e69;rport From: <sip:<DID_3>@<Sip Provider_2>>;tag=as19d1fe3f To: <sip:<DID_3>@<Sip Provider_2>> Call-ID: 0ac3c85c1181b12a316b078c04edddcc@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_3>@<internal IP of asterisk server>> Event: registration Content-Length: 0 ---   == CDR updated on SIP/<USER PROVIDER 2>-09491ea0     -- Executing [601@ivr-2:1] ExecIf("SIP/<USER PROVIDER 2>-09491ea0", "0|dbDel|") in new stack     -- Executing [601@ivr-2:2] Set("SIP/<USER PROVIDER 2>-09491ea0", "__NODEST=") in new stack     -- Executing [601@ivr-2:3] Goto("SIP/<USER PROVIDER 2>-09491ea0", "from-did-direct|601|1") in new stack     -- Goto (from-did-direct,601,1)     -- Executing [601@from-did-direct:1] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "0?ext-local|601|1") in new stack     -- Executing [601@from-did-direct:2] Macro("SIP/<USER PROVIDER 2>-09491ea0", "user-callerid|") in new stack     -- Executing [s@macro-user-callerid:1] NoOp("SIP/<USER PROVIDER 2>-09491ea0", "user-callerid: 8070010000 8070010000") in new stack     -- Executing [s@macro-user-callerid:2] Set("SIP/<USER PROVIDER 2>-09491ea0", "AMPUSER=8070010000") in new stack     -- Executing [s@macro-user-callerid:3] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "0?report") in new stack     -- Executing [s@macro-user-callerid:4] ExecIf("SIP/<USER PROVIDER 2>-09491ea0", "1|Set|REALCALLERIDNUM=8070010000") in new stack     -- Executing [s@macro-user-callerid:5] NoOp("SIP/<USER PROVIDER 2>-09491ea0", "REALCALLERIDNUM is 8070010000") in new stack     -- Executing [s@macro-user-callerid:6] Set("SIP/<USER PROVIDER 2>-09491ea0", "AMPUSER=") in new stack     -- Executing [s@macro-user-callerid:7] Set("SIP/<USER PROVIDER 2>-09491ea0", "AMPUSERCIDNAME=") in new stack     -- Executing [s@macro-user-callerid:8] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "1?report") in new stack     -- Goto (macro-user-callerid,s,13)     -- Executing [s@macro-user-callerid:13] NoOp("SIP/<USER PROVIDER 2>-09491ea0", "TTL:  ARG1: ") in new stack     -- Executing [s@macro-user-callerid:14] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "0?continue") in new stack     -- Executing [s@macro-user-callerid:15] Set("SIP/<USER PROVIDER 2>-09491ea0", "__TTL=64") in new stack     -- Executing [s@macro-user-callerid:16] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "1?continue") in new stack     -- Goto (macro-user-callerid,s,23)     -- Executing [s@macro-user-callerid:23] NoOp("SIP/<USER PROVIDER 2>-09491ea0", "Using CallerID "8070010000" <8070010000>") in new stack     -- Executing [601@from-did-direct:3] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "1?skipdb") in new stack     -- Goto (from-did-direct,601,5)     -- Executing [601@from-did-direct:5] Set("SIP/<USER PROVIDER 2>-09491ea0", "__NODEST=") in new stack     -- Executing [601@from-did-direct:6] Set("SIP/<USER PROVIDER 2>-09491ea0", "__BLKVM_OVERRIDE=BLKVM/601/SIP/<USER PROVIDER 2>-09491ea0") in new stack     -- Executing [601@from-did-direct:7] Set("SIP/<USER PROVIDER 2>-09491ea0", "__BLKVM_BASE=601") in new stack     -- Executing [601@from-did-direct:8] Set("SIP/<USER PROVIDER 2>-09491ea0", "DB(BLKVM/601/SIP/<USER PROVIDER 2>-09491ea0)=TRUE") in new stack     -- Executing [601@from-did-direct:9] Set("SIP/<USER PROVIDER 2>-09491ea0", "RRNODEST=") in new stack     -- Executing [601@from-did-direct:10] Set("SIP/<USER PROVIDER 2>-09491ea0", "__NODEST=601") in new stack     -- Executing [601@from-did-direct:11] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "1?REPCID") in new stack     -- Goto (from-did-direct,601,16)     -- Executing [601@from-did-direct:16] NoOp("SIP/<USER PROVIDER 2>-09491ea0", "CALLERID(name) is 8070010000") in new stack     -- Executing [601@from-did-direct:17] Set("SIP/<USER PROVIDER 2>-09491ea0", "_RGPREFIX=Sales") in new stack     -- Executing [601@from-did-direct:18] Set("SIP/<USER PROVIDER 2>-09491ea0", "CALLERID(name)=Sales8070010000") in new stack     -- Executing [601@from-did-direct:19] Set("SIP/<USER PROVIDER 2>-09491ea0", "RecordMethod=Group") in new stack     -- Executing [601@from-did-direct:20] Macro("SIP/<USER PROVIDER 2>-09491ea0", "record-enable|601-8072515614#|Group") in new stack     -- Executing [s@macro-record-enable:1] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "0?2:4") in new stack     -- Goto (macro-record-enable,s,4)     -- Executing [s@macro-record-enable:4] AGI("SIP/<USER PROVIDER 2>-09491ea0", "recordingcheck|20080403-214616|1207273567.5") in new stack     -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck   recordingcheck|20080403-214616|1207273567.5: Recording enable for 601   recordingcheck|20080403-214616|1207273567.5: CALLFILENAME=g601-20080403-214616-1207273567.5     -- AGI Script recordingcheck completed, returning 0     -- Executing [s@macro-record-enable:999] MixMonitor("SIP/<USER PROVIDER 2>-09491ea0", "g601-20080403-214616-1207273567.5.wav") in new stack     -- Executing [601@from-did-direct:21] Set("SIP/<USER PROVIDER 2>-09491ea0", "RingGroupMethod=ringallv2") in new stack     -- Executing [601@from-did-direct:22] Set("SIP/<USER PROVIDER 2>-09491ea0", "_FMGRP=601") in new stack     -- Executing [601@from-did-direct:23] GotoIf("SIP/<USER PROVIDER 2>-09491ea0", "0?DIALGRP") in new stack     -- Executing [601@from-did-direct:24] Answer("SIP/<USER PROVIDER 2>-09491ea0", "") in new stack     -- Executing [601@from-did-direct:25] Wait("SIP/<USER PROVIDER 2>-09491ea0", "1") in new stack   == Begin MixMonitor Recording SIP/<USER PROVIDER 2>-09491ea0     -- Executing [601@from-did-direct:26] Playback("SIP/<USER PROVIDER 2>-09491ea0", "custom/warn_legal") in new stack     -- <SIP/<USER PROVIDER 2>-09491ea0> Playing 'custom/warn_legal' (language 'en') Retransmitting #5 (NAT) to 209.197.130.18:5060: REGISTER sip:<Sip provider_1> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK3fc15c21;rport From: <sip:<DID_1>@<Sip provider_1>>;tag=as60ab06be To: <sip:<DID_1>@<Sip provider_1>> Call-ID: 074ef09a4f625b581652d3fa3e7a1b21@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_1>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #5 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK152285ad;rport From: <sip:<DID_2>@<Sip Provider_2>>;tag=as10122dfe To: <sip:<DID_2>@<Sip Provider_2>> Call-ID: 0c553d677136a1616b9fa58811079b49@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_2>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- Retransmitting #5 (NAT) to 216.211.27.163:5060: REGISTER sip:<Sip Provider_2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK10018e69;rport From: <sip:<DID_3>@<Sip Provider_2>>;tag=as19d1fe3f To: <sip:<DID_3>@<Sip Provider_2>> Call-ID: 0ac3c85c1181b12a316b078c04edddcc@<WAN IP OF ROUTER> CSeq: 152 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:<DID_3>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- REGISTER 13 headers, 0 lines Reliably Transmitting (NAT) to 64.34.164.254:5060: REGISTER sip:<SIP Provider 2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK75def67a;rport From: <sip:<USER PROVIDER 2>@<SIP Provider 2>>;tag=as7418e56e To: <sip:<USER PROVIDER 2>@<SIP Provider 2>> Call-ID: 0ed4ddfb5a70db5a7fb1f1b251b06ac0@<WAN IP OF ROUTER> CSeq: 143 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="<USER PROVIDER 2>", realm="voicenetwork.ca", algorithm=MD5, uri="sip:<SIP Provider 2>", nonce="0437de7d", response="a04f5346678541bc5567e0c2ad271cd6", opaque="" Expires: 120 Contact: <sip:<USER PROVIDER 2>@<internal IP of asterisk server>> Event: registration Content-Length: 0 --- voip*CLI> <--- SIP read from 64.34.164.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK75def67a;received=<WAN ROUTER IP>;rport=5060 From: <sip:<USER PROVIDER 2>@<SIP Provider 2>>;tag=as7418e56e To: <sip:<USER PROVIDER 2>@<SIP Provider 2>> Call-ID: 0ed4ddfb5a70db5a7fb1f1b251b06ac0@<WAN IP OF ROUTER> CSeq: 143 REGISTER User-Agent: Voice Network Inc 1b Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:<USER PROVIDER 2>@64.34.164.254> Content-Length: 0 <-------------> --- (10 headers 0 lines) --- voip*CLI> <--- SIP read from 64.34.164.254:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK75def67a;received=<WAN ROUTER IP>;rport=5060 From: <sip:<USER PROVIDER 2>@<SIP Provider 2>>;tag=as7418e56e To: <sip:<USER PROVIDER 2>@<SIP Provider 2>>;tag=as4e56d0f4 Call-ID: 0ed4ddfb5a70db5a7fb1f1b251b06ac0@<WAN IP OF ROUTER> CSeq: 143 REGISTER User-Agent: Voice Network Inc 1b Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="voicenetwork.ca", nonce="2a68d7b8" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Responding to challenge, registration to domain/host name <SIP Provider 2> REGISTER 13 headers, 0 lines Reliably Transmitting (NAT) to 64.34.164.254:5060: REGISTER sip:<SIP Provider 2> SIP/2.0 Via: SIP/2.0/UDP <internal IP of asterisk server>:5060;branch=z9hG4bK4128105c;rport From: <sip:<USER PROVIDER 2>@<SIP Provider 2>>;tag=as1a4d4b7d To: <sip:<USER PROVIDER 2>@<SIP Provider 2>> Call-ID: 0ed4ddfb5a70db5a7fb1f1b251b06ac0@<WAN IP OF ROUTER> CSeq: 144 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="<USER PROVIDER 2>", realm="voicenetwork.ca", algorithm=MD5, uri="sip:<SIP Provider 2>", nonce="2a68d7b8", response="d0c1f33d1352a2fe21917be0658b047e", opaque="" Expires: 120 Contact: <sip:<USER PROVIDER 2>@<internal IP of asterisk server>> Event: registration Content-Length: 0

Can anyone let in some insight? I'm starting to see why there are all those windows Voip commericals now lol.

The network setup is as follows:


<NET>------<router>---------<server>


I have port forwarding for 5060-5090 to the server IP
and a port forward for the IAX2 to the server IP

No firewall rules on the router or the server.

On the server the "route" is setup with only one default route the IP of the internal network's gateway

I tried Dail strings to make sure that wasn't a problem.


Sample of one DID Configuration...

Outgoing....

Trunk name is <user>
------------------------

Code:

username=<user> type=peer secret=<password> nat=yes host=<sip provider> fromuser=<user> insecure=very context=from-trunk qualify=yes

Incomming...

Code:

Incomming trunk name is: <user>-in username=<user> type=user secret=<password> nat=yes insecure=very fromuser=<user> context=from-trunk qualify=yes

Connect string...

<user>:<password>@<sip provider>/<user>


For the asterisk INFO is states....

Registration Sent.

Hi
For it's own reasons, Asterisk uses ports 10,000 up for its RTP streams, so you will need to open those ports as well.

IAX is totally different to SIP. You can't register to a SIP provider with it unless they specifically support it

I switched to IAXs for one of my DIDs because one of the providers push it more then sip. So they do offer it, and it wasn't blindly implemented. I'm using the provider supplied freepbx connection strings. I was trying to see if the one protocal would work but appearly not. The connection strings have worked in the past for SIP. I bet it's something very simple but I can't find it / see it. Rebooted several times flushed the router settings.

Still incomming calls work perfectly, no lag, cut outs, jitter. But outgoing is not working!