Re: [asterisk-users] calling specific 1800-number not going through.
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] calling specific 1800-number not going through.
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: ;tag=1c1457828994
> To:
> Call-ID: 1457828497512012183855@10.0.0.110
> CSeq: 1 OPTIONS
> Contact:
> 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: ;tag=1c1457828994
> To: ;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" ;tag=as64f6417c
> To:
> Contact:
> 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: ;tag=4fc8ac12
> From: "asterisk";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: ;tag=1c1472330741
> To:
> Call-ID: 809487713120129287@10.0.0.110
> CSeq: 245 REGISTER
> Contact: ;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: ;tag=1c1472330741
> To: ;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: ;tag=1c1472330741
> To:
> 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: ;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 l
[asterisk-users] calling specific 1800-number not going through.
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: ;tag=1c1457828994
To:
Call-ID: 1457828497512012183855@10.0.0.110
CSeq: 1 OPTIONS
Contact:
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: ;tag=1c1457828994
To: ;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" ;tag=as64f6417c
To:
Contact:
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: ;tag=4fc8ac12
From: "asterisk";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: ;tag=1c1472330741
To:
Call-ID: 809487713120129287@10.0.0.110
CSeq: 245 REGISTER
Contact: ;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: ;tag=1c1472330741
To: ;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: ;tag=1c1472330741
To:
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: ;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: ;tag=1c1472330741
To: ;tag=as21c548bd
Call-ID: 809487713120129287@10.0.0.110
CSeq: 246 REGISTER
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: ;expires=3600
Date: Fri, 06 Jan 2012 01:39:11 GMT
Content-Length: 0
<>
Scheduling destruction of SIP dia
