asterisk users - Jan 2004 - ATA call

Is your ATA running SIP if so, what version (2.16?)

With SIP, then * extensions.conf and sip.conf files are configured
you should see the following

asterisk3*CLI> sip show peers
Name/username    Host                 Mask             Port     Status
3000/3000        10.0.0.30       (D)  255.255.255.255  5060     OK (15 ms)
9000/9000        10.0.0.90       (D)  255.255.255.255  5060     OK (47 ms)

ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960

to test an extension from the CLI
CLI>dial <ext. #>
you should hear your ATA ring

Doug

Osvaldo Mundim Junior wrote:
> Hey all!
>
> I'm having problems trying to set up an ATA 186 with my Asterisk box.
When I
> get the phone to place the call, I type the extension and I only get busy
> signal after 5 seconds. So I can't call my Asterisk box from my ATA and
> either call from my Asterisk to my ATA.
>
> Does anybody know what can be happing?
>
> Log is attached..
>
> tks
> regards
> Oz
>
>   ------------------------------------------------------------------------
>                   Name: ast_log.txt
>    ast_log.txt    Type: Plain Text (text/plain)
>               Encoding: quoted-printable
--
FREE Unlimited Worldwide Voip calling
set-up an account and start saving today!
http://www.voippages.com ext. 7000
http://www.pulver.com/fwd/ ext. 83740
free IP phone software @
http://www.xten.com/
http://iaxclient.sourceforge.net/iaxcomm/

Hey all!

I'm having problems trying to set up an ATA 186 with my Asterisk box. When I
get the phone to place the call, I type the extension and I only get busy
signal after 5 seconds. So I can't call my Asterisk box from my ATA and
either call from my Asterisk to my ATA.

Does anybody know what can be happing?

Log is attached..

tks
regards
Oz
-------------- next part --------------
> 8 headers, 0 lines
> Retransmitting #1 (NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219
> From: sip:porto@200.170.156.77;tag=3346186142
> To: <sip:4057@200.170.156.77;user=phone>;tag=as36ac1b92
> Call-ID: 471622025@192.168.0.150
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact:
> Proxy-Authenticate: Digest realm="asterisk",
nonce="4bd7a841"
> Content-Length: 0
> 
> 290?
>  to 200.167.103.219:1025
> Sip read: LI>
> INVITE sip:4057@200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060
> From: sip:porto@200.170.156.77;tag=3346186142
> To: <sip:4057@200.170.156.77;user=phone>
> Call-ID: 471622025@192.168.0.150
> CSeq: 2 INVITE
> Contact: <sip:porto@192.168.0.150:5060;transport=udp>
> User-Agent: Cisco ATA 186  v2.16.1 ata18x (030709a)
> Proxy-Authorization: Digest
>
username="porto",realm="asterisk",nonce="4bd7a841",uri="sip:4057@200.170.156
> .77",response="1ecb99d4d5e23be179a9eb55eb33c62a"
> Expires: 300
> Content-Length: 250
> Content-Type: application/sdp
> 
> v=0
> o=porto 3642 3642 IN IP4 192.168.0.150
> s=ATA186 Call
> c=IN IP4 192.168.0.150
> t=0 0
> m=audio 16384 RTP/AVP 18 8 0 101
> a=rtpmap:18 G729/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> 
> 12 headers, 11 lines
> Using latest request as basis request
> Sending to 192.168.0.150 : 5060 (NAT)
> Found audio format UNKN
> Found audio format ALAW
> Found audio format UNKN
> Found audio format UNKN
> Found description format G729
> Found description format PCMA
> Found description format PCMU
> Found description format telephone-event
> Capabilities: us - 256, them - 268/0, combined - 256
> Non-codec capabilities: us - 1, them - 1, combined - 1
> 10 headers, 0 lines
> Reliably Transmitting:
> OPTIONS sip:200.167.103.219:1025 SIP/2.0
> Via: SIP/2.0/UDP 200.170.156.77:5060;branch=z9hG4bK1937468f
> From: "asterisk"
<sip:asterisk@200.170.156.77>;tag=as5566fcc8
> To: <sip:200.167.103.219:1025>
> Contact: <sip:asterisk@200.170.156.77>
> Call-ID: 2d2e8097757b385a3444a1ab239ab86c@200.170.156.77
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Length: 0
> 
>  (no NAT) to 200.167.103.219:1025
> Sip read: LI>
> ACK sip:4057@200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219
> From: sip:porto@200.170.156.77;tag=3346186142
> To: <sip:4057@200.170.156.77;user=phone>;tag=as36ac1b92
> Call-ID: 471622025@192.168.0.150
> CSeq: 1 ACK
> User-Agent: Cisco ATA 186  v2.16.1 ata18x (030709a)
> Content-Length: 0
> 
> 
> 8 headers, 0 lines
> Sip read: LI>
> INVITE sip:4057@200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060
> From: sip:porto@200.170.156.77;tag=3346186142
> To: <sip:4057@200.170.156.77;user=phone>
> Call-ID: 471622025@192.168.0.150
> CSeq: 2 INVITE
> Contact: <sip:porto@192.168.0.150:5060;transport=udp>
> User-Agent: Cisco ATA 186  v2.16.1 ata18x (030709a)
> Proxy-Authorization: Digest
>
username="porto",realm="asterisk",nonce="514a024a",uri="sip:4057@200.170.156
> .77",response="adb7da64c3f557d1db20b699c04f6d84"
> Expires: 300
> Content-Length: 250
> Content-Type: application/sdp
> 
> v=0
> o=porto 3692 3692 IN IP4 192.168.0.150
> s=ATA186 Call
> c=IN IP4 192.168.0.150
> t=0 0
> m=audio 16384 RTP/AVP 18 8 0 101
> a=rtpmap:18 G729/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> 
> 12 headers, 11 lines
> Using latest request as basis request
> Sending to 192.168.0.150 : 5060 (non-NAT)
> Found audio format UNKN
> Found audio format ALAW
> Found audio format UNKN
> Found audio format UNKN
> Found description format G729
> Found description format PCMA
> Found description format PCMU
> Found description format telephone-event
> Capabilities: us - 256, them - 268/0, combined - 256
> Non-codec capabilities: us - 1, them - 1, combined - 1
> Reliably Transmitting (NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219
> From: sip:porto@200.170.156.77;tag=3346186142
> To: <sip:4057@200.170.156.77;user=phone>;tag=as046b1041
> Call-ID: 471622025@192.168.0.150
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact:
> Proxy-Authenticate: Digest realm="asterisk",
nonce="6512ffab"
> Content-Length: 0
> 
> 
>  to 200.167.103.219:1025
> Sip read: LI>
> ACK sip:4057@200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219
> From: sip:porto@200.170.156.77;tag=3346186142
> To: <sip:4057@200.170.156.77;user=phone>;tag=as36ac1b92
> Call-ID: 471622025@192.168.0.150
> CSeq: 1 ACK
> User-Agent: Cisco ATA 186  v2.16.1 ata18x (030709a)
> Content-Length: 0
> 
> 
> 8 headers, 0 lines
> Retransmitting #1 (no NAT):
> OPTIONS sip:200.167.103.219:1025 SIP/2.0
> Via: SIP/2.0/UDP 200.170.156.77:5060;branch=z9hG4bK1937468f
> From: "asterisk"
<sip:asterisk@200.170.156.77>;tag=as5566fcc8
> To: <sip:200.167.103.219:1025>
> Contact: <sip:asterisk@200.170.156.77>
> Call-ID: 2d2e8097757b385a3444a1ab239ab86c@200.170.156.77
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Length: 0
> 
> tent-
>  to 200.167.103.219:1025
> Sip read: LI>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 200.170.156.77:5060;branch=z9hG4bK1937468f
> From: "asterisk"
<sip:asterisk@200.170.156.77>;tag=as5566fcc8
> To: <sip:200.167.103.219:1025>;tag=3346186142
> Call-ID: 2d2e8097757b385a3444a1ab239ab86c@200.170.156.77
> CSeq: 102 OPTIONS
> Server: Cisco ATA 186  v2.16.1 ata18x (030709a)
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
> Content-Length: 250
> Content-Type: application/sdp
> 
> v=0
> o=porto 3779 3779 IN IP4 192.168.0.150
> s=ATA186 Call
> c=IN IP4 192.168.0.150
> t=0 0
> m=audio 16384 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:18 G729/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> 
> 10 headers, 11 lines
> Retransmitting #1 (NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219
> From: sip:porto@200.170.156.77;tag=3346186142
> To: <sip:4057@200.170.156.77;user=phone>;tag=as046b1041
> Call-ID: 471622025@192.168.0.150
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact:
> Proxy-Authenticate: Digest realm="asterisk",
nonce="6512ffab"
> Content-Length: 0
> 
> 
>  to 200.167.103.219:1025
> Sip read: LI>
> INVITE sip:4057@200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060
> From: sip:porto@200.170.156.77;tag=3346186142
> To: <sip:4057@200.170.156.77;user=phone>
> Call-ID: 471622025@192.168.0.150
> CSeq: 2 INVITE
> Contact: <sip:porto@192.168.0.150:5060;transport=udp>
> User-Agent: Cisco ATA 186  v2.16.1 ata18x (030709a)
> Proxy-Authorization: Digest
>
username="porto",realm="asterisk",nonce="4bd7a841",uri="sip:4057@200.170.156
> .77",response="1ecb99d4d5e23be179a9eb55eb33c62a"
> Expires: 300
> Content-Length: 250
> Content-Type: application/sdp
> 
> v=0
> o=porto 3792 3792 IN IP4 192.168.0.150
> s=ATA186 Call
> c=IN IP4 192.168.0.150
> t=0 0
> m=audio 16384 RTP/AVP 18 8 0 101
> a=rtpmap:18 G729/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> 
> 12 headers, 11 lines
> Ignoring this request
> Retransmitting #2 (NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219
> From: sip:porto@200.170.156.77;tag=3346186142
> To: <sip:4057@200.170.156.77;user=phone>;tag=as046b1041
> Call-ID: 471622025@192.168.0.150
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact:
> Proxy-Authenticate: Digest realm="asterisk",
nonce="6512ffab"
> Content-Length: 0
> 
> 
>  to 200.167.103.219:1025
> Sip read: LI>
> ACK sip:4057@200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219
> From: sip:porto@200.170.156.77;tag=3346186142
> To: <sip:4057@200.170.156.77;user=phone>;tag=as046b1041
> Call-ID: 471622025@192.168.0.150
> CSeq: 2 ACK
> User-Agent: Cisco ATA 186  v2.16.1 ata18x (030709a)
> Content-Length: 0
> 
> 
> 8 headers, 0 lines
> Sip read: LI>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 200.170.156.77:5060;branch=z9hG4bK1937468f
> From: "asterisk"
<sip:asterisk@200.170.156.77>;tag=as5566fcc8
> To: <sip:200.167.103.219:1025>;tag=3346186142
> Call-ID: 2d2e8097757b385a3444a1ab239ab86c@200.170.156.77
> CSeq: 102 OPTIONS
> Server: Cisco ATA 186  v2.16.1 ata18x (030709a)
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
> Content-Length: 250
> Content-Type: application/sdp
> 
> v=0
> o=porto 3885 3885 IN IP4 192.168.0.150
> s=ATA186 Call
> c=IN IP4 192.168.0.150
> t=0 0
> m=audio 16384 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:18 G729/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15