Try this just dial 0 to dial out and then dial a internal number at you
company
What are the xxxx between 916 and 2128 ?
exten => _0.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN},70,Tt)-----Oorspronkelijk bericht----- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Thczv F. Thczv Verzonden: donderdag 15 september 2005 5:55 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Starting From Scratch Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed [EMAIL PROTECTED] It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to understand how all this works, I have stripped my extensions.conf down to almost nothing. I am building it up piece by piece. This is the entirety of my extensions.conf file: [globals] OUTBOUNDTRUNK=SIP/mysipprovider.com [from-internal] exten => 105,1,Answer() exten => 105,2,Playback(abandon-all-hope) exten => 105,3,Hangup() exten => 106,1,Dial(${OUTBOUNDTRUNK}/916xxx6000) exten => 107,1,Dial(${OUTBOUNDTRUNK}/916xxx2128) This is all just testing. When I dial 105 from either of my softphones, it plays the recording fine. My thought for the 106 and 107 extensions was to sort of hard code it so that if I dialed either of those extensions the call would automatically get routed over my outbound sip trunk to the appropriate offsite PSTN dialable number. Once I know that the calls will go through, I will create a proper dial plan. The 6000 number is my home PSTN phone. The 2128 number is my office desk phone (a SIP phone). Here is where I get stumped: When I dial 107 from my SIP softphone, the call goes out fine, and rings my SIP office phone just fine. In other words, my asterisk box passes the call to my company's server, and it goes through. The guys at work tell me that outside PSTN calls (like to my home PSTN phone) should work exactly the same way (no special dialing needed, just the standard 10 digit telephone number). But for some reason when I dial 106 from my softphone it doesn't work. I get a recording that tells me "the person you are trying to reach is unavailable." As near as I can tell (as someone unexperienced with this) from looking at the text that gets spit out when I run "SIP debug," both calls go through the same. The debug info from when I dialed 106 is included below. I know I am a dummy about this stuff. But I am trying to learn how it works. Have mercy on me you experts. Can any of you see what might be wrong? My guesses include two possibilities: 1. For some reason my PSTN onramp (my own company) isn't really passing the call to the PSTN as it should. 2. Even for this really basic hardcoding experiment, my extensions.conf file is too short. For example, the connection takes longer than asterisk expects, and so I need to tell it to keep waiting before playing the recording. Or something like that. Any ideas? Thanks, Dave ************ INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK032524f5 From: "102" <sip:[EMAIL PROTECTED]>;tag=as0dbb6283 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 15 Sep 2005 03:41:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 240 v=0 o=root 1541 1541 IN IP4 192.168.1.200 s=session c=IN IP4 192.168.1.200 t=0 0 m=audio 17464 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 - - - - (no NAT) to 66.81.0.87:5060 -- Called smf-reg.sip.o1.com/916xxx6000 asterisk1*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK032524f5;received=24.23.48.16 From: "102" <sip:[EMAIL PROTECTED]>;tag=as0dbb6283 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 66.81.0.87:5060 "Noisy feedback tells: pid=22068 req_src_ip=24.23.48.16 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri=sip:[EMAIL PROTECTED] via_cnt==1" 9 headers, 0 lines asterisk1*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.200:5060;received=24.23.48.16;branch=z9hG4bK032524f5 Record-Route: <sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on> Record-Route: <sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on> From: "102" <sip:[EMAIL PROTECTED]>;tag=as0dbb6283 To: <sip:[EMAIL PROTECTED]>;tag=as5f7d7868 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk SIPv2 (http://www.asterisk.org CVS-HEAD-03/02/05-12:13:56 ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 210 v=0 o=root 9338 9338 IN IP4 66.81.0.97 s=session c=IN IP4 66.81.0.97 t=0 0 m=audio 16488 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 13 headers, 10 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 66.81.0.97:16488 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: <sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on> list_route: hop: <sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on> list_route: hop: <sip:[EMAIL PROTECTED]> set_destination: Parsing <sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on> for address/port to send to set_destination: set destination to 66.81.0.87, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK137a266c Route: <sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on>,<sip:[EMAIL PROTECTED] 2.168.4.97> From: "102" <sip:[EMAIL PROTECTED]>;tag=as0dbb6283 To: <sip:[EMAIL PROTECTED]>;tag=as5f7d7868 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 66.81.0.87:5060 -- SIP/mysipprovider.com-9208 answered SIP/102-eb1a We're at 192.168.1.200 port 16880 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK4731E35A23434B6A9D2A94DFB9D2448E From: 102 <sip:[EMAIL PROTECTED]>;tag=787222251 To: <sip:[EMAIL PROTECTED]>;tag=as4f23aa40 Call-ID: [EMAIL PROTECTED] CSeq: 36288 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 240 v=0 o=root 1541 1541 IN IP4 192.168.1.200 s=session c=IN IP4 192.168.1.200 t=0 0 m=audio 16880 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 - - - - to 192.168.1.110:5060 -- Attempting native bridge of SIP/102-eb1a and SIP/smf-reg.sip.o1.com-9208 asterisk1*CLI> Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:5060;rport;branch=z9hG4bKCC576F589D574F5FB4D8F9553C08E91A From: 102 <sip:[EMAIL PROTECTED]>;tag=787222251 To: <sip:[EMAIL PROTECTED]>;tag=as4f23aa40 Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 36288 ACK Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines asterisk1*CLI> Sip read: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:5060;rport;branch=z9hG4bK702E81C5C4E74BD593ABA699EB591494 From: 102 <sip:[EMAIL PROTECTED]>;tag=787222251 To: <sip:[EMAIL PROTECTED]>;tag=as4f23aa40 Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 36289 BYE Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 10 headers, 0 lines Sending to 192.168.1.110 : 5060 (non-NAT) Transmitting (no NAT): IP/2.0 200 OK> Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK702E81C5C4E74BD593ABA699EB591494 From: 102 <sip:[EMAIL PROTECTED]>;tag=787222251 To: <sip:[EMAIL PROTECTED]>;tag=as4f23aa40 Call-ID: [EMAIL PROTECTED] CSeq: 36289 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.1.110:5060 set_destination: Parsing <sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on> for address/port to send to set_destination: set destination to 66.81.0.87, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK13e08827 Route: <sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on>,<sip:[EMAIL PROTECTED] 2.168.4.97> From: "102" <sip:[EMAIL PROTECTED]>;tag=as0dbb6283 To: <sip:[EMAIL PROTECTED]>;tag=as5f7d7868 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 66.81.0.87:5060 == Spawn extension (from-internal, 106, 1) exited non-zero on 'SIP/102-eb1a' asterisk1*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.200:5060;received=24.23.48.16;branch=z9hG4bK13e08827 Record-Route: <sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on> Record-Route: <sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on> From: "102" <sip:[EMAIL PROTECTED]>;tag=as0dbb6283 To: <sip:[EMAIL PROTECTED]>;tag=as5f7d7868 Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk SIPv2 (http://www.asterisk.org CVS-HEAD-03/02/05-12:13:56 ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 12 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' asterisk1*CLI> _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
