[Asterisk-Users] Help Please!!!!

Thu Feb 17 10:04:41 MST 2005

Thanks, I will begin my testing

Erick
----- Original Message ----- 
From: "Race Vanderdecken" <asteriskusers at codetyrant.com>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
<asterisk-users at 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
> asteriskusers at codetyrant.com
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Erick
> Weber V.
> Sent: Wednesday, February 16, 2005 2:34 PM
> To: asterisk-users at 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:1088 at 201.133.170.82 SIP/2.0
> Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport
> From: "Weber Automundo" <sip:404 at XXX.XXX.XXX.XXX>;tag=as4da46cda
> To: <sip:1088 at 201.133.170.82>
> Contact: <sip:404 at XXX.XXX.XXX.XXX>
> Call-ID: 00b325641a0f0d680014aad165ce6d4c at XXX.XXX.XXX.XXX
> 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:1088 at 192.168.1.2:5060;user=phone;transport=udp>
> set_destination: Parsing
> <sip:1088 at 192.168.1.2:5060;user=phone;transport=udp> for address/port to
>
> send to
> set_destination: set destination to 192.168.1.2, port 5060
> Transmitting:
> ACK sip:1088 at 192.168.1.2:5060 SIP/2.0
> Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport
> From: "Weber Automundo" <sip:404 at XXX.XXX.XXX.XXX>;tag=as4da46cda
> To: <sip:1088 at 201.133.170.82>;tag=939809556
> Contact: <sip:404 at XXX.XXX.XXX.XXX>
> Call-ID: 00b325641a0f0d680014aad165ce6d4c at XXX.XXX.XXX.XXX
> 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:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
> To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as601a996c
> Call-ID: 30194281 at 192.168.1.2
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:1088 at XXX.XXX.XXX.XXX>
> 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:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
> To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as601a996c
> Call-ID: 30194281 at 192.168.1.2
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:1088 at XXX.XXX.XXX.XXX>
> WWW-Authenticate: Digest realm="asterisk", nonce="0711b1d6"
> Content-Length: 0
>
>
> to 201.133.170.82:5060
> Scheduling destruction of call '30194281 at 192.168.1.2' 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:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
> To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as601a996c
> Call-ID: 30194281 at 192.168.1.2
> CSeq: 2 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:1088 at XXX.XXX.XXX.XXX>
> 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:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
> To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as601a996c
> Call-ID: 30194281 at 192.168.1.2
> CSeq: 2 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Expires: 120
> Contact: <sip:1088 at XXX.XXX.XXX.XXX>;expires=120
> Date: Wed, 16 Feb 2005 00:43:46 GMT
> Content-Length: 0
>
>
> to 201.133.170.82:5060
> Scheduling destruction of call '30194281 at 192.168.1.2' 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:asterisk at XXX.XXX.XXX.XXX>;tag=as59adf4c2
> To: <sip:201.133.170.82>
> Contact: <sip:asterisk at XXX.XXX.XXX.XXX>
> Call-ID: 00373ed80ed1caa4513efc600569cc96 at XXX.XXX.XXX.XXX
> 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 '00373ed80ed1caa4513efc600569cc96 at XXX.XXX.XXX.XXX'
> set_destination: Parsing
> <sip:1088 at 192.168.1.2:5060;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:1088 at 192.168.1.2:5060 SIP/2.0
> Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK2bdff4fa;rport
> From: "Weber Automundo" <sip:404 at XXX.XXX.XXX.XXX>;tag=as4da46cda
> To: <sip:1088 at 201.133.170.82>;tag=939809556
> Contact: <sip:404 at XXX.XXX.XXX.XXX>
> Call-ID: 00b325641a0f0d680014aad165ce6d4c at XXX.XXX.XXX.XXX
> 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 '00b325641a0f0d680014aad165ce6d4c at XXX.XXX.XXX.XXX'
> Destroying call '30194281 at 192.168.1.2'
> 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:asterisk at XXX.XXX.XXX.XXX>;tag=as370254a4
> To: <sip:201.133.170.82>
> Contact: <sip:asterisk at XXX.XXX.XXX.XXX>
> Call-ID: 78dea4a01158e0154b01eba37450a08b at XXX.XXX.XXX.XXX
> 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 '78dea4a01158e0154b01eba37450a08b at XXX.XXX.XXX.XXX'
> 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:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
> To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as13999c1a
> Call-ID: 30194281 at 192.168.1.2
> CSeq: 3 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:1088 at XXX.XXX.XXX.XXX>
> 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:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
> To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as13999c1a
> Call-ID: 30194281 at 192.168.1.2
> CSeq: 3 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:1088 at XXX.XXX.XXX.XXX>
> WWW-Authenticate: Digest realm="asterisk", nonce="33e2f5df"
> Content-Length: 0
>
>
> to 201.133.170.82:5060
> Scheduling destruction of call '30194281 at 192.168.1.2' 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:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
> To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as13999c1a
> Call-ID: 30194281 at 192.168.1.2
> CSeq: 4 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:1088 at XXX.XXX.XXX.XXX>
> 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:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
> To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as13999c1a
> Call-ID: 30194281 at 192.168.1.2
> CSeq: 4 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Expires: 120
> Contact: <sip:1088 at XXX.XXX.XXX.XXX>;expires=120
> Date: Wed, 16 Feb 2005 00:45:30 GMT
> Content-Length: 0
>
>
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