[asterisk-users] calling specific 1800-number not going through.

2012-01-05 Thread Joseph 

I have a strange problem.
I'm using the same dialplan to call 1800-number:

[toll-free]
;second 7 audiocodes strips
exten = _71800XXX,1,Dial(SIP/7${EXTEN:1}@pstn-5665,60,tr)

When I call this number (through pstn-5665) 18005000347 the phone always rings 
busy.
When I call any other 1800-number the calls goes through.

When I call the same phone number 18005000347 through a different line the 
calls goes through every time.

Here is call (busy) trace to that 18005000347 with sip debug ON:

Can anybody decipher why I'm getting busy signal to that particular 1800-number 
but not others?


--- SIP read from UDP:10.0.0.110:5060 ---
OPTIONS sip:gateway@10.0.0.110 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1457834404
Max-Forwards: 70
From: sip:gateway@10.0.0.110:5060;tag=1c1457828994
To: sip:gateway@10.0.0.110
Call-ID: 1457828497512012183855@10.0.0.110
CSeq: 1 OPTIONS
Contact: sip:gateway@10.0.0.110:5060
Allow: 
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

-
--- (12 headers 0 lines) ---
Looking for gateway in default (domain 10.0.0.110)

--- Transmitting (NAT) to 10.0.0.110:5060 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 
10.0.0.110;branch=z9hG4bKac1457834404;received=10.0.0.110;rport=5060
From: sip:gateway@10.0.0.110:5060;tag=1c1457828994
To: sip:gateway@10.0.0.110;tag=as7091ae01
Call-ID: 1457828497512012183855@10.0.0.110
CSeq: 1 OPTIONS
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0



Scheduling destruction of SIP dialog '1457828497512012183855@10.0.0.110' in 
32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to 81.15.150.20:5060:
OPTIONS sip:sip.actio.pl SIP/2.0
Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK03484db5
Max-Forwards: 70
From: asterisk sip:asterisk@10.0.0.100;tag=as64f6417c
To: sip:sip.actio.pl
Contact: sip:asterisk@10.0.0.100:5060
Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
CSeq: 102 OPTIONS
User-Agent: Centrala
Date: Fri, 06 Jan 2012 01:39:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Length: 0


---

--- SIP read from UDP:81.15.150.20:5060 ---
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP 
10.0.0.100:5060;branch=z9hG4bK03484db5;received=68.148.245.78;rport=48715
To: sip:sip.actio.pl;tag=4fc8ac12
From: asterisksip:asterisk@10.0.0.100;tag=as64f6417c
Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
CSeq: 102 OPTIONS
Content-Length: 0

-
--- (7 headers 0 lines) ---
Really destroying SIP dialog '66070317301f64861df62d20769ba385@10.0.0.100:5060' 
Method: OPTIONS
-- Accepted AUTHENTICATED TBD call from 10.0.0.108

--- SIP read from UDP:10.0.0.110:5060 ---
REGISTER sip:10.0.0.100 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360
Max-Forwards: 70
From: sip:11@10.0.0.110;tag=1c1472330741
To: sip:11@10.0.0.110
Call-ID: 809487713120129287@10.0.0.110
CSeq: 245 REGISTER
Contact: sip:11@10.0.0.110:5060;expires=3600
Allow: 
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 3600
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Content-Length: 0

-
--- (12 headers 0 lines) ---
Sending to 10.0.0.110:5060 (NAT)

--- Transmitting (no NAT) to 10.0.0.110:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360;received=10.0.0.110
From: sip:11@10.0.0.110;tag=1c1472330741
To: sip:11@10.0.0.110;tag=as21c548bd
Call-ID: 809487713120129287@10.0.0.110
CSeq: 245 REGISTER
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=3a451a5b
Content-Length: 0



Scheduling destruction of SIP dialog '809487713120129287@10.0.0.110' in 32000 
ms (Method: REGISTER)

--- SIP read from UDP:10.0.0.110:5060 ---
REGISTER sip:10.0.0.100 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472407428
Max-Forwards: 70
From: sip:11@10.0.0.110;tag=1c1472330741
To: sip:11@10.0.0.110
Call-ID: 809487713120129287@10.0.0.110
CSeq: 246 REGISTER
Authorization: Digest 
username=11,realm=asterisk,nonce=3a451a5b,uri=sip:10.0.0.100,algorithm=MD5,response=5dd6df18064f3d23cb86ca306820e596
Contact: sip:11@10.0.0.110:5060;expires=3600
Allow: 
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 3600
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Content-Length: 0

-
--- (13 headers 0 lines) ---
Sending to 10.0.0.110:5060 (no NAT)

--- Transmitting (no NAT) to 10.0.0.110:5060 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472407428;received=10.0.0.110
From:

Re: [asterisk-users] calling specific 1800-number not going through.

2012-01-05 Thread Jim Dickenson 
It took 36 seconds for that number to answer when I called it and it looks like 
the call hung up after 32000 ms when you dialed via asterisk.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jan 5, 2012, at 5:45 PM, Joseph wrote:

 I have a strange problem.
 I'm using the same dialplan to call 1800-number:
 
 [toll-free]
 ;second 7 audiocodes strips
 exten = _71800XXX,1,Dial(SIP/7${EXTEN:1}@pstn-5665,60,tr)
 
 When I call this number (through pstn-5665) 18005000347 the phone always 
 rings busy.
 When I call any other 1800-number the calls goes through.
 
 When I call the same phone number 18005000347 through a different line the 
 calls goes through every time.
 
 Here is call (busy) trace to that 18005000347 with sip debug ON:
 
 Can anybody decipher why I'm getting busy signal to that particular 
 1800-number but not others?
 
 
 --- SIP read from UDP:10.0.0.110:5060 ---
 OPTIONS sip:gateway@10.0.0.110 SIP/2.0
 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1457834404
 Max-Forwards: 70
 From: sip:gateway@10.0.0.110:5060;tag=1c1457828994
 To: sip:gateway@10.0.0.110
 Call-ID: 1457828497512012183855@10.0.0.110
 CSeq: 1 OPTIONS
 Contact: sip:gateway@10.0.0.110:5060
 Allow: 
 REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
 User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
 Accept: application/sdp, application/simple-message-summary, message/sipfrag
 Content-Length: 0
 
 -
 --- (12 headers 0 lines) ---
 Looking for gateway in default (domain 10.0.0.110)
 
 --- Transmitting (NAT) to 10.0.0.110:5060 ---
 SIP/2.0 404 Not Found
 Via: SIP/2.0/UDP 
 10.0.0.110;branch=z9hG4bKac1457834404;received=10.0.0.110;rport=5060
 From: sip:gateway@10.0.0.110:5060;tag=1c1457828994
 To: sip:gateway@10.0.0.110;tag=as7091ae01
 Call-ID: 1457828497512012183855@10.0.0.110
 CSeq: 1 OPTIONS
 Server: Centrala
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
 PUBLISH
 Supported: replaces, timer
 Accept: application/sdp
 Content-Length: 0
 
 
 
 Scheduling destruction of SIP dialog '1457828497512012183855@10.0.0.110' in 
 32000 ms (Method: OPTIONS)
 Reliably Transmitting (no NAT) to 81.15.150.20:5060:
 OPTIONS sip:sip.actio.pl SIP/2.0
 Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK03484db5
 Max-Forwards: 70
 From: asterisk sip:asterisk@10.0.0.100;tag=as64f6417c
 To: sip:sip.actio.pl
 Contact: sip:asterisk@10.0.0.100:5060
 Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
 CSeq: 102 OPTIONS
 User-Agent: Centrala
 Date: Fri, 06 Jan 2012 01:39:07 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
 PUBLISH
 Supported: replaces, timer
 Content-Length: 0
 
 
 ---
 
 --- SIP read from UDP:81.15.150.20:5060 ---
 SIP/2.0 501 Unsupported Method
 Via: SIP/2.0/UDP 
 10.0.0.100:5060;branch=z9hG4bK03484db5;received=68.148.245.78;rport=48715
 To: sip:sip.actio.pl;tag=4fc8ac12
 From: asterisksip:asterisk@10.0.0.100;tag=as64f6417c
 Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
 CSeq: 102 OPTIONS
 Content-Length: 0
 
 -
 --- (7 headers 0 lines) ---
 Really destroying SIP dialog 
 '66070317301f64861df62d20769ba385@10.0.0.100:5060' Method: OPTIONS
-- Accepted AUTHENTICATED TBD call from 10.0.0.108
 
 --- SIP read from UDP:10.0.0.110:5060 ---
 REGISTER sip:10.0.0.100 SIP/2.0
 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360
 Max-Forwards: 70
 From: sip:11@10.0.0.110;tag=1c1472330741
 To: sip:11@10.0.0.110
 Call-ID: 809487713120129287@10.0.0.110
 CSeq: 245 REGISTER
 Contact: sip:11@10.0.0.110:5060;expires=3600
 Allow: 
 REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
 Expires: 3600
 User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
 Content-Length: 0
 
 -
 --- (12 headers 0 lines) ---
 Sending to 10.0.0.110:5060 (NAT)
 
 --- Transmitting (no NAT) to 10.0.0.110:5060 ---
 SIP/2.0 401 Unauthorized
 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360;received=10.0.0.110
 From: sip:11@10.0.0.110;tag=1c1472330741
 To: sip:11@10.0.0.110;tag=as21c548bd
 Call-ID: 809487713120129287@10.0.0.110
 CSeq: 245 REGISTER
 Server: Centrala
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
 PUBLISH
 Supported: replaces, timer
 WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=3a451a5b
 Content-Length: 0
 
 
 
 Scheduling destruction of SIP dialog '809487713120129287@10.0.0.110' in 32000 
 ms (Method: REGISTER)
 
 --- SIP read from UDP:10.0.0.110:5060 ---
 REGISTER sip:10.0.0.100 SIP/2.0
 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472407428
 Max-Forwards: 70
 From: sip:11@10.0.0.110;tag=1c1472330741
 To: sip:11@10.0.0.110
 Call-ID: 809487713120129287@10.0.0.110
 CSeq: 246 REGISTER
 Authorization: Digest 
 username=11,realm=asterisk,nonce=3a451a5b,uri=sip:10.0.0.100,algorithm=MD5,response=5dd6df18064f3d23cb86ca306820e596
 Contact: sip:11@10.0.0.110:5060;expires=3600
 Allow:

Re: [asterisk-users] calling specific 1800-number not going through.

2012-01-05 Thread Joseph 

Solved,
It seems to me the vendor is blocking the 1800 number in Western Canada.
Our second line I'm not sure where it is terminated: Toronto or USA or it could 
be they are blocking the 1800 line to home users and not for business lines.

--
Joseph

--
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