I have a small system, server, client and 2 phones. Phones are polycom 501's. In general all is working fine. I can call the two phones, speak etc... I can have the server call each phone and play a wave file.
However, when trying to setup a direct dial number of 1044 that calls another machine running asterisk - something ODD is happening. ; This is not working.... [smvoice-sip] exten => 1044,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044) exten => 1044,n,Hangup ; changing 1044 to 10 works find. [smvoice-sip] exten => 10,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044) exten => 10,n,Hangup I am running 1.4.22 and DAHDI 2.0.0 complete. Why is it picking up "10" when trying to dial "1044". How can I determine what is going on here. Thanks, Jerry This is the SIP debug for the 1044 case that does not work. ----------------- Use 'exit' when done Asterisk 1.4.22, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <[EMAIL PROTECTED]> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Found [0;37;40m[1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found [0mConnected to Asterisk 1.4.22 currently running on devcentos5x64 (pid = 3127) devcentos5x64*CLI> Verbosity is at least 5 [Kdevcentos5x64*CLI> <--- SIP read from 192.168.1.89:5060 ---> INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bKb872214aDBECCC5D From: "404" <sip:[EMAIL PROTECTED]>;tag=25AB8538-7BACFE71 To: <sip:[EMAIL PROTECTED];user=phone> CSeq: 1 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Supported: 1?00rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1226069152 1226069152 IN IP4 192.168.1.89 s=Polycom IP Phone c=IN IP4 192.168.1.89 t=0 0 m=audio 2244 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 11 lines) --- Sending to 192.168.1.89 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] <--- Reliably Transmitting (no NAT) to 192.168.1.89:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bKb872214aDBECCC5D;received=192.168.1.89 From: "404" <sip:[EMAIL PROTECTED]>;tag=25AB8538-7BACFE71 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5a3d998e Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces? Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f1b706f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) Found user '404' ?? <--- SIP read from 192.168.1.89:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bKb872214aDBECCC5D From: "404" <sip:[EMAIL PROTECTED]>;tag=25AB8538-7BACFE71 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5a3d998e CSeq: 1 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Max-Forwards: ?70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- ? [Kdevcentos5x64*CLI> <--- SIP read from 192.168.1.89:5060 ---> INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bK9c456f3360D09552 From: "404" <sip:[EMAIL PROTECTED]>;tag=25AB8538-7BACFE71 To: <sip:[EMAIL PROTECTED];user=phone> CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Supported: 1?00rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="404", realm="asterisk", nonce="1f1b706f", uri="sip:[EMAIL PROTECTED];user=phone", response="c6e14f94fa0bbe3d742b6f570982ed79", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1226069152 1226069152 IN IP4 192.168.1.89 s=Polycom IP Phone c=IN IP4 192.168.1.89 t=0 0 m=audio 2244 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (15 headers 11 lines) --- Sending to 192.168.1.89 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found user '404' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.89:2244 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for???????????? ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.89:2244 Looking for 10 in smvoice-sip (domain 192.168.1.8) <--- Reliably Transmitting (no NAT) to 192.168.1.89:5060 ---> SIP/2.0 484 Address Incomplete Via:?????? SIP/2.0/UDP 192.168.1.89;branch=z9hG4bK9c456f3360D09552;received=192.168.1.89 From: "404" <sip:[EMAIL PROTECTED]>;tag=25AB8538-7BACFE71 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5a3d998e Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INV?ITE) [Kdevcentos5x64*CLI> <--- SIP read from 192.168.1.89:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bK9c456f3360D09552 From: "404" <sip:[EMAIL PROTECTED]>;tag=25AB8538-7BACFE71 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5a3d998e CSeq: 2 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Proxy-Authoriz?ation: Digest username="404", realm="asterisk", nonce="1f1b706f", uri="sip:[EMAIL PROTECTED];user=phone", response="c6e14f94fa0bbe3d742b6f570982ed79", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- ? [Kdevcentos5x64*CLI> <--- SIP read from 192.168.1.89:5060 ---> INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bK51dc2d00EB13255B From: "404" <sip:[EMAIL PROTECTED]>;tag=25AB8538-7BACFE71 To: <sip:[EMAIL PROTECTED];user=phone> CSeq: 3 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Supporte?d: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="404", realm="asterisk", nonce="1f1b706f", uri="sip:[EMAIL PROTECTED];user=phone", response="75209201fe7d6f0854ecb918879e6049", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1226069152 1226069153 IN IP4 192.168.1.89 s=Polycom IP Phone c=IN IP4 192.168.1.89 t=0 0 m=audio 2244 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (15 headers 11 lines) --- Sending to 192.168.1.89 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found user '404' [Nov 7 09:48:02] NOTICE[3145]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user "404" <sip:[EMAIL PROTECTED]>;tag=25AB8538-7BACFE71 <--- Reliably Transmitting (no NAT) to 192.168.1.89:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4b?????K51dc2d00EB13255B;received=192.168.1.89 From: "404" <sip:[EMAIL PROTECTED]>;tag=25AB8538-7BACFE71 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5a3d998e Call-ID: [EMAIL PROTECTED] CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) ? [Kdevcentos5x64*CLI> <--- SIP read from 192.168.1.89:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bK51dc2d00EB13255B From: "404" <sip:[EMAIL PROTECTED]>;tag=25AB8538-7BACFE71 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5a3d998e CSeq: 3 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Proxy-Auth?orization: Digest username="404", realm="asterisk", nonce="1f1b706f", uri="sip:[EMAIL PROTECTED];user=phone", response="c6e14f94fa0bbe3d742b6f570982ed79", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- ? [Kdevcentos5x64*CLI> quit Executing last minute cleanups Asterisk cleanly ending (0). [0m _______________________________________________ -- 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
