Thanks, I will begin my testing
Erick
----- Original Message ----- From: "Race Vanderdecken" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>
Sent: Wednesday, February 16, 2005 8:18 PM
Subject: RE: [Asterisk-Users] Help Please!!!!
Greetings Mr. Weber,
Remember the rule in mathematics that is much easier to solve for one variable.
You stateed you are having a problem with the 1088 extension. If look like you are trying to make a call from the 404 extension to the 1088 extension.
1. If you have 6 ATA's running shut 5 of them off. Test each one separately. Then turn one on at a time and see the problem can be traced to one ATA
2. You are getting sent an authorization request from asterisk to the 1088 extension.
WWW-Authenticate: Digest realm="asterisk", nonce="0711b1d6"
Make sure you don't have any of the secret= or the md5secret= stuff set in the sip.conf, until you can get each phone to talk in the open. Then change, one, 1, uno, phone at a time.
3. If you have a SIP phone that is not an ATA then set it up and try to dial the 1088 and see if you get the same thing.
4. Do a sip show users to make sure the 1088 is registered with asterisk.
5. Do the normal, things don't work dance, by unplugging the phone and reconnecting a different phone to the ata. Change the power suplly with another ata. Change the RJ45 patch cable. Try a different port in the switch or wall. Swap one of the known working ATA and change it to the 1088 ata.
6. Go to lunch and have a beer. Find a new job and settle down with a good woman. Leave telecom and go into organic farming.
Race "The Tyrant" Vanderdecken [EMAIL PROTECTED]
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Wednesday, February 16, 2005 2:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help Please!!!! Importance: High
I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is that one of them is dropping calls an I can't figure out what is the problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem.
Any help will be appreciate
Thanks
Erick Weber
VoIP*CLI> sip debug peer 1088 SIP Debugging Enabled for IP: 201.133.170.82:5060 Peer RTP is at port 192.168.1.69:0 Peer RTP is at port 192.168.1.69:0 -- Executing Dial("SIP/404-cbc9", "SIP/1088|60|tr") in new stack We're at XXX.XXX.XXX.XXX port 17506 Answering/Requesting with root capability 256 12 headers, 8 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport From: "Weber Automundo" <sip:[EMAIL PROTECTED]>;tag=as4da46cda To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164
v=0 o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX s=session c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 17506 RTP/AVP 18 a=rtpmap:18 G729/8000 a=silenceSupp:off - - - - (NAT) to 201.133.170.82:5060 -- Called 1088 -- SIP/1088-ec82 is ringing Found RTP audio format 18 Found RTP audio format 101 Peer RTP is at port 192.168.1.2:0 Found description format G729 Found description format telephone-event Capabilities: us - 0x100(G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x100(G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> set_destination: Parsing <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> for address/port to
send to set_destination: set destination to 192.168.1.2, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport From: "Weber Automundo" <sip:[EMAIL PROTECTED]>;tag=as4da46cda To: <sip:[EMAIL PROTECTED]>;tag=939809556 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0
(NAT) to 201.133.170.82:5060 -- SIP/1088-ec82 answered SIP/404-cbc9 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 Using latest request as basis request Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0
to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> WWW-Authenticate: Digest realm="asterisk", nonce="0711b1d6" Content-Length: 0
to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Using latest request as basis request Sending to 192.168.1.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0
to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: <sip:[EMAIL PROTECTED]>;expires=120 Date: Wed, 16 Feb 2005 00:43:46 GMT Content-Length: 0
to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:201.133.170.82 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK0972cae7 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as59adf4c2 To: <sip:201.133.170.82> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0
(no NAT) to 201.133.170.82:5060 Destroying call '[EMAIL PROTECTED]' set_destination: Parsing <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> for address/port to
send to set_destination: set destination to 192.168.1.2, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK2bdff4fa;rport From: "Weber Automundo" <sip:[EMAIL PROTECTED]>;tag=as4da46cda To: <sip:[EMAIL PROTECTED]>;tag=939809556 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0
(NAT) to 201.133.170.82:5060 == Spawn extension (hi, 1088, 1) exited non-zero on 'SIP/404-cbc9' -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 192.168.1.2 Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:201.133.170.82 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK0689fc21 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as370254a4 To: <sip:201.133.170.82> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:44:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0
(no NAT) to 201.133.170.82:5060 Destroying call '[EMAIL PROTECTED]' Using latest request as basis request Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as13999c1a Call-ID: [EMAIL PROTECTED] CSeq: 3 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0
to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as13999c1a Call-ID: [EMAIL PROTECTED] CSeq: 3 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> WWW-Authenticate: Digest realm="asterisk", nonce="33e2f5df" Content-Length: 0
to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Using latest request as basis request Sending to 192.168.1.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as13999c1a Call-ID: [EMAIL PROTECTED] CSeq: 4 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0
to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as13999c1a Call-ID: [EMAIL PROTECTED] CSeq: 4 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: <sip:[EMAIL PROTECTED]>;expires=120 Date: Wed, 16 Feb 2005 00:45:30 GMT Content-Length: 0
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