Re: [asterisk-users] Call routing based on CID
Hi,
I've been trying to route incoming calls based on CID to a trunk but the
calls are not getting though. I am trying to use a wild card prefix based
on countries so I can point the call to the appropriate trunk.
I am running Asterisk 1.8 with FreePBX.
Here is a sample of my configuration in extentions_custom.conf
exten = _00336123412xx/44XX.,1,Set(RINGTIME=90,g)
exten = _00336123412xx/44XX.,n,Answer(10) exten =
_00336123412xx/44XX.,n,Dial(SIP/trunk01/${EXTEN},${RINGTIME})
exten = _00336123412xx/44XX.,n,Hangup()
Any kind suggestions is appreciated. Thanks.
Best Regards,
Geoffrey Yeoh
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Re: [asterisk-users] Call routing based on CID
Try: exten = _00336123412xx/_44XX.,1,Set(RINGTIME=90,g)
Notice the _ on your callerid pattern
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoffrey Yeoh
Sent: Thursday, October 11, 2012 1:15 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Call routing based on CID
Hi,
I've been trying to route incoming calls based on CID to a trunk but the calls
are not getting though. I am trying to use a wild card prefix based on
countries so I can point the call to the appropriate trunk.
I am running Asterisk 1.8 with FreePBX.
Here is a sample of my configuration in extentions_custom.conf
exten = _00336123412xx/44XX.,1,Set(RINGTIME=90,g)
exten = _00336123412xx/44XX.,n,Answer(10) exten =
_00336123412xx/44XX.,n,Dial(SIP/trunk01/${EXTEN},${RINGTIME})
exten = _00336123412xx/44XX.,n,Hangup()
Any kind suggestions is appreciated. Thanks.
Best Regards,
Geoffrey Yeoh
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[asterisk-users] Call routing based on CID
Thanks Eric. That works.
--
Try: exten = _00336123412xx/_44XX.,1,Set(RINGTIME=90,g)
Notice the _ on your callerid pattern
-Original Message-
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Geoffrey
Yeoh
Sent: Thursday, October 11, 2012 1:15 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Call routing based on CID
Hi,
I've been trying to route incoming calls based on CID to a trunk but the
calls are not getting though. I am trying to use a wild card prefix based
on countries so I can point the call to the appropriate trunk.
I am running Asterisk 1.8 with FreePBX.
Here is a sample of my configuration in extentions_custom.conf
exten = _00336123412xx/44XX.,1,Set(RINGTIME=90,g)
exten = _00336123412xx/44XX.,n,Answer(10) exten =
_00336123412xx/44XX.,n,Dial(SIP/trunk01/${EXTEN},${RINGTIME})
exten = _00336123412xx/44XX.,n,Hangup()
Any kind suggestions is appreciated. Thanks.
Best Regards,
Geoffrey Yeoh
--
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[asterisk-users] Call routing between two Asterisk boxes using SIP not working ...
Hello there! I need some help to configure two Asterix boxes to route calls using SIP. I followed the instructions present at this site: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html;, but I couldn't get it working so far. The only difference, besides the names that I've used, is that I'm using realtime to retrieve all information. Both boxes registrate on the other perfectly. The problem happens when one call gets routed. It seems that realtime on destination box is trying to find locally a SIP user 1001 that is the originator of the call and is a user of the original box. It finally ends with a: chan_sip.c:14780 handle_request_invite: Failed to authenticate user 1001 sip:1...@10.10.100.158;tag=as1e79b629 on destination box. Wireshark present on destination box indicates all the following steps: 1- Wengo client registered with user 1001 starts the call to number 2001 with Box 1 (at 10.10.100.158); 2- Box 1 makes the challenge; 3- Wengo replies the challenge; 4- Box 1 send an successfull ack to Wengo client and sends the INVITE to Box 2 (at 10.10.100.156) that holds user 2001; 5- Box 2 makes the challenge; 6- Box 1 replies the challenge; 7- Box 2 sends a 403 Forbidden; Has anyone had this problem ? Can anyone help me out on that ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call routing between two Asterisk boxes using SIP not working ...
Hi guys! The problem was solved by the use of same password for registration users of both boxes. Is there no way to indicate different password for registration user of Box1 and registration user of Box2 ? Thanks and best regards, Mauro. Mauro Sergio Ferreira Brasil escreveu: Hello there! I need some help to configure two Asterix boxes to route calls using SIP. I followed the instructions present at this site: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html;, but I couldn't get it working so far. The only difference, besides the names that I've used, is that I'm using realtime to retrieve all information. Both boxes registrate on the other perfectly. The problem happens when one call gets routed. It seems that realtime on destination box is trying to find locally a SIP user 1001 that is the originator of the call and is a user of the original box. It finally ends with a: chan_sip.c:14780 handle_request_invite: Failed to authenticate user 1001 sip:1...@10.10.100.158;tag=as1e79b629 on destination box. Wireshark present on destination box indicates all the following steps: 1- Wengo client registered with user 1001 starts the call to number 2001 with Box 1 (at 10.10.100.158); 2- Box 1 makes the challenge; 3- Wengo replies the challenge; 4- Box 1 send an successfull ack to Wengo client and sends the INVITE to Box 2 (at 10.10.100.156) that holds user 2001; 5- Box 2 makes the challenge; 6- Box 1 replies the challenge; 7- Box 2 sends a 403 Forbidden; Has anyone had this problem ? Can anyone help me out on that ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call routing in voicemail
Dear All, When one configures a standalone asterisk voicemail and attach to a legacy PBX, and the PBX transfers a busy, no-response or switchedoff extension to Asterisk. What would be the source and destination caller IDs that would get to asterisk voicemail? Thanks, ... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call routing in voicemail
Dear All, When one configures a standalone asterisk voicemail and attach to a legacy PBX, and the PBX transfers a busy, no-response or switchedoff extension to Asterisk. What would be the source and destination IDs that would get to asyerisk voicemail? Thanks, ... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Routing
HI I am able to setup the Dundi and works fine in locating the phone number's and extensions in branch office's. Only problem is unable to route the call if we receive it on serverA from PSTN and some one enter the extension number which reside in ServerB, it doesn't route the call. But if I dial the extension on ServerB from phone on serverA it works fine. Ali Arshad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Routing Time Issue
This was orignally posted on The Asterisk Blog Forums. See the original post here.Pete101 says: I am having issues with all inbound calls coming into the system. It is taking like 10 seconds for it to decide where to route the call. It applies for both PSTN calls and VoiP calls. Does anyone have any idea where I should start?-- www.AsteriskBlog.comYour home for easy to learn Asterisk stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Routing based on Caller-Id
Hello, How can I build extensions.conf so that Asterisk routes calls based on the ANI, not the number dialed. Example: All calls coming down a PRI are going to the same number. I would like to route them to a new number based on the Calling-Station-Id. I.E. All calls from 413-773- go to 413-773-1234 -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Routing based on Caller-Id
Matthew Crocker wrote: Hello, How can I build extensions.conf so that Asterisk routes calls based on the ANI, not the number dialed. Example: All calls coming down a PRI are going to the same number. I would like to route them to a new number based on the Calling-Station-Id. I.E. All calls from 413-773- go to 413-773-1234 Use the 'ex-girlfriend' option in extensions.conf. See this http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf BTW, does anybody know the history of why ANI-based routing is called 'ex-girlfriend' option? AFAIK, it has always been referenced that way, both in the sample extensions.conf and docs, since the pre 1.0 days (if my memory serves me right - as far back as 0.3). Cheers. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Routing based on Caller-Id
http://72.14.203.104/search?q=cache:ggbL-X3kx_4J:www.loligo.com/asterisk/misc/Presentations/Asterisk-overview.v1.0.ppt+ex-girlfriend+ANI+asteriskhl=engl=usct=clnkcd=1 Redirection based on ANI * You can match against calling number instead of called number. * This is a.k.a. "The ex-girlfriend filter" by the inventor of the routines * This pattern matches against called number (1410…) and also against calling numer (301…) The ex-girlfriend filter = filtering unwanted calls based on number, vs. number dialed -brandonOn 8/2/06, Leo Ann Boon [EMAIL PROTECTED] wrote:Matthew Crocker wrote: Hello, How can I build extensions.conf so that Asterisk routes calls based on the ANI, not the number dialed. Example: All calls coming down a PRI are going to the same number.I would like to route them to a new number based on the Calling-Station-Id. I.E. All calls from 413-773- go to 413-773-1234Use the 'ex-girlfriend' option in extensions.conf. See this http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.confBTW, does anybody know the history of why ANI-based routing is called'ex-girlfriend' option? AFAIK, it has always been referenced that way, both in the sample extensions.conf and docs, since the pre 1.0 days (ifmy memory serves me right - as far back as 0.3).Cheers.Leo___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brandon GalbraithEmail: [EMAIL PROTECTED]AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Routing from GnuGK to Asterisk
Hi,
im using debian with asterisk instaled via apt-get (1.0.7 + oh323)
my asterisk is not a Gk, but a GW.
Look this conf
/etc/asterisk/oh323.conf
[register]
alias=asterisk
gwprefix=9
gwprefix=8
gwprefix=7
gwprefix=6
gwprefix=5
gwprefix=4
gwprefix=3
gwprefix=2
gwprefix=1
gwprefix=0
/etc/gatekeeper.ini
[RasSrv::GWPrefixes]
127.0.0.1
asterisk=0,1,2,3,4,5,6,7,8,9
in this way i force all calls to asterisk
i use some extensions with tech-prefix to route calls
like this
exten = _130X.,1,SetCallerID(130)
exten = _130X.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED],tr
Rafael Marconi
Em 09/12/2005, às 18:03, Code Lover escreveu:
Hi all,
I installed asterisk-oh323 and GnuGK on the linux box, and i wanted to
route all gnugk registered endpoint's call to oh323 GW (Asterisk)
which is already registered with GnuGK as Gatekeepr.
here is my gnugk.ini configuation.
[Gatekeeper::Main]
Fourtytwo=42
TimeToLive=750
Name=gnugk
[RoutedMode]
GKRouted=1
H245Routed=0
CallSignalPort=1720
CallSignalHandlerNumber=1
RemoveH245AddressOnTunneling=0
AcceptNeighborsCalls=1
AcceptUnregisteredCalls=1
SupportNATedEndpoints=1
DropCallsByReleaseComplete=1
[RasSrv::ARQFeatures]
ArjReasonRouteCallToSCN=0
ArjReasonRouteCallToGatekeeper=1
RoundRobinGateways=1
[RoutingPolicy]
default=neighbor
[RasSrv::Neighbors]
GK1=asterisk
[Neighbor::GK1]
GatekeeperIdentifier=GK1
Host=212.xxx.xxx.xxx
SendPrefixes=0
AcceptPrefixes=*
ForwardLRQ=always
Here is my error what i am getting in gnugk log.
admissionReject {
requestSeqNum = 8191
rejectReason = calledPartyNotRegistered null
}
Please help me to solve this problem
--
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Thank You,
Code Lover
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Re: [Asterisk-Users] Call Routing from GnuGK to Asterisk
Hi Rafael Marconi,
I follow your configuration but it does not seems to work. and i am
getting some error.
oh323.conf
-
GKID
GnuGk
gatekeeper=asteriskserveronsamemachine.com
gatekeeperTTL=600
userInputMode=TONE
amaFlags=default
accountCode=H323
language=en
musiconhold=default
[register]
alias=asterisk
gwprefix=9
gwprefix=8
gwprefix=7
gwprefix=6
gwprefix=5
gwprefix=4
gwprefix=3
gwprefix=2
gwprefix=1
gwprefix=0
context=from-oh323
my.ini
-
[RasSrv::GWPrefixes]
127.0.0.1
asterisk=0,1,2,3,4,5,6,7,8,9
It is not sending any request on asterisk server.
disengageReject {
requestSeqNum = 1
rejectReason = requestToDropOther null
}
--
Thank You,
Code Lover
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[Asterisk-Users] Call Routing from GnuGK to Asterisk
Hi all,
I installed asterisk-oh323 and GnuGK on the linux box, and i wanted to
route all gnugk registered endpoint's call to oh323 GW (Asterisk)
which is already registered with GnuGK as Gatekeepr.
here is my gnugk.ini configuation.
[Gatekeeper::Main]
Fourtytwo=42
TimeToLive=750
Name=gnugk
[RoutedMode]
GKRouted=1
H245Routed=0
CallSignalPort=1720
CallSignalHandlerNumber=1
RemoveH245AddressOnTunneling=0
AcceptNeighborsCalls=1
AcceptUnregisteredCalls=1
SupportNATedEndpoints=1
DropCallsByReleaseComplete=1
[RasSrv::ARQFeatures]
ArjReasonRouteCallToSCN=0
ArjReasonRouteCallToGatekeeper=1
RoundRobinGateways=1
[RoutingPolicy]
default=neighbor
[RasSrv::Neighbors]
GK1=asterisk
[Neighbor::GK1]
GatekeeperIdentifier=GK1
Host=212.xxx.xxx.xxx
SendPrefixes=0
AcceptPrefixes=*
ForwardLRQ=always
Here is my error what i am getting in gnugk log.
admissionReject {
requestSeqNum = 8191
rejectReason = calledPartyNotRegistered null
}
Please help me to solve this problem
--
--
Thank You,
Code Lover
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RE: [Asterisk-Users] Call Routing based on number dialed (using S IP)
Is this even possible or am I better off getting a voip number for each of the existing numbers I want to forward. Thanks! -Original Message- From: Geoff Manning [mailto:[EMAIL PROTECTED] Sent: Friday, June 03, 2005 4:53 PM To: Asterisk Users (E-mail) Subject: [Asterisk-Users] Call Routing based on number dialed (using SIP) Is it possible to route calls based on the number called when the inbound call is SIP based? Here is what we are trying to do: 1) Someone dials one of the companies 5 long standing, published phone numbers which have been forwarded to ONE Voip telephone number by the telco. 2) The SER server where that Voip number terminates is passing it to our Asterisk server 3) Is there a way to determine what the original number dialed was? We want to avoid needing a Voip number for every forwarded number. Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Routing based on number dialed (using S IP)
Geoff Manning wrote: Is this even possible or am I better off getting a voip number for each of the existing numbers I want to forward. Thanks! -Original Message- From: Geoff Manning [mailto:[EMAIL PROTECTED] Sent: Friday, June 03, 2005 4:53 PM To: Asterisk Users (E-mail) Subject: [Asterisk-Users] Call Routing based on number dialed (using SIP) Is it possible to route calls based on the number called when the inbound call is SIP based? Here is what we are trying to do: 1) Someone dials one of the companies 5 long standing, published phone numbers which have been forwarded to ONE Voip telephone number by the telco. 2) The SER server where that Voip number terminates is passing it to our Asterisk server 3) Is there a way to determine what the original number dialed was? We want to avoid needing a Voip number for every forwarded number. Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can use the sipgetheader() application. If you pass a call to asterisk, the field To in the SIP header stay as originally dialed. So, with sipgetheader(or_To=To) Cut(or_To,or_To,:,2) Cut(or_To,or_To,@,1) in your dialplan, you can get the original dialed number. with the cut function you can cut the sip: and the @domain.asd substrings. Mirko ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Routing based on number dialed (using S IP)
sipgetheader(or_To=To) Cut(or_To,or_To,:,2) Cut(or_To,or_To,@,1) That works! Thanks! Correction to the cut command below, replaced , with = : sipgetheader(or_To=To) Cut(or_To=or_To,:,2) Cut(or_To=or_To,@,1) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Routing based on number dialed (using S IP)
The thing with this is what he said: forward to ONE Voip telephone number by the telco. Asterisk will not know which number was dialed, because to it - it's just another call going to that single telephone number... Just call forwarding! NOW in an extreme case depending on the hardware and agreements, you could get the original number that was dialed sent as another SIP header along with other information... But that's likely not going to happen. - Joshua Colp. (file in #asterisk on Freenode) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mirko Marghitola Sent: Tuesday, June 07, 2005 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Routing based on number dialed (using S IP) Geoff Manning wrote: Is this even possible or am I better off getting a voip number for each of the existing numbers I want to forward. Thanks! -Original Message- From: Geoff Manning [mailto:[EMAIL PROTECTED] Sent: Friday, June 03, 2005 4:53 PM To: Asterisk Users (E-mail) Subject: [Asterisk-Users] Call Routing based on number dialed (using SIP) Is it possible to route calls based on the number called when the inbound call is SIP based? Here is what we are trying to do: 1) Someone dials one of the companies 5 long standing, published phone numbers which have been forwarded to ONE Voip telephone number by the telco. 2) The SER server where that Voip number terminates is passing it to our Asterisk server 3) Is there a way to determine what the original number dialed was? We want to avoid needing a Voip number for every forwarded number. Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can use the sipgetheader() application. If you pass a call to asterisk, the field To in the SIP header stay as originally dialed. So, with sipgetheader(or_To=To) Cut(or_To,or_To,:,2) Cut(or_To,or_To,@,1) in your dialplan, you can get the original dialed number. with the cut function you can cut the sip: and the @domain.asd substrings. Mirko ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Routing based on number dialed (using SIP)
Is it possible to route calls based on the number called when the inbound call is SIP based? Here is what we are trying to do: 1) Someone dials one of the companies 5 long standing, published phone numbers which have been forwarded to ONE Voip telephone number by the telco. 2) The SER server where that Voip number terminates is passing it to our Asterisk server 3) Is there a way to determine what the original number dialed was? We want to avoid needing a Voip number for every forwarded number. Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call routing
I have two asterisk boxes connected using IAX. There are two T1s on each box. I have all my dialing rules in one of the asterisk boxes and all of my agents register on the same box where I have all the dialing rules. See diagram below: Asterisk_1 --2xT1-- PSTN || || Asterisk_2 --2xT1-- PSTN || || SIP_Agents I'm wondering how can I configure extensions.conf in Asterisk_1 so that EVERY incoming call (regardless of DID or CallerID or whatever) received from PSTN in Asterisk_1 is routed via IAX to Asterisk_2? Basically, anything that arrives from Zap/g1 or Zap/g2 in Asterisk_1 should be automatically routed to Asterisk_2 preserving all call features, such as DID, CallerID, etc. Any ideas? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call routing question
Hi Cameron,
Thanks for the suggestions. I think this is precisely what I was
looking for, unfortunately neither of those variables appears to be set
on my incoming calls. This is probably because I'm doing remote call
forwarding which is done by the phone company rather than regular call
forwarding. I guess I'll just have to get different numbers from my
VOIP provider in order to route my Verizon numbers to different extensions.
Thanks again for the help,
-Herman
Cameron Beattie wrote:
Try using the special identifiers ${DNID} or ${RDNIS}. Refer to http://www.voip-info.org/tiki-index.php?page=Asterisk%20variables for more info.
Regards
Cameron
Original message
--
Date: Tue, 08 Mar 2005 10:09:37 -0500
From: Herman Sheremetyev [EMAIL PROTECTED]
Subject: [Asterisk-Users] call routing question
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi All,
I have a question about call routing. I currently have a phone number provided by Voicepulse that connects directly to my Asterisk box and another phone number provided by Verizon that I have Remote Call
Forwarded to the Voicepulse number. What I'm wondering is if the
information about which number is actually dialed available for me to route the
calls to different extensions? Thanks for the help and I apologize if this has
already been discussed.
Thanks,
-Herman
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[Asterisk-Users] call routing question
Hi All, I have a question about call routing. I currently have a phone number provided by Voicepulse that connects directly to my Asterisk box and another phone number provided by Verizon that I have Remote Call Forwarded to the Voicepulse number. What I'm wondering is if the information about which number is actually dialed available for me to route the calls to different extensions? Thanks for the help and I apologize if this has already been discussed. Thanks, -Herman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call routing question
Try using the special identifiers ${DNID} or ${RDNIS}. Refer to http://www.voip-info.org/tiki-index.php?page=Asterisk%20variablesfor
more info.
Regards
Cameron
Original message
--
Date: Tue, 08 Mar 2005 10:09:37 -0500
From: Herman Sheremetyev [EMAIL PROTECTED]
Subject: [Asterisk-Users] call routing question
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi All,
I have a question about call routing. I currently have a phone number
provided by Voicepulse that connects directly to my Asterisk box and another
phone number provided by Verizon that I have Remote Call
Forwarded to the Voicepulse number. What I'm wondering is if the
information about which number is actually dialed available for me to route
the calls to different extensions? Thanks for the help and I apologize if this
has already been discussed.
Thanks,
-Herman
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[Asterisk-Users] call routing on free trunk ??
Hello: I would like to know if it's possible to route a call based on a free trunk. Let's say we got 5 o 6 locations with 4 analog lines and asterisk and Digium hardware on each. We got 10 sip phones ( or IAX ) on each location. We got an sdsl 400 Kbps with the same Internet provider, with good perfomance on their own network. We have done all the IP QOS stuff (we got internet cafe too) And when the user dial a number we do an LCR preasignation (adding 1050 e.g. in front of the string dialed) depending on prefix (LCDial e.g.). At this moment how can I route the call to a free analog on a remote asterisk line if I got the 4 full lines at this location ?? Is it by dialplan ?? Do I need to register all trunks on all servers or can I do centralized ?? If the call is transfered. where will be the CDRs produced ?? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call routing based on remote ip address.
While setting up my first dial plan, I find that notions like remote ip, network, or incoming network interface seem to be totally lacking regarding calling parties, where * still seems to fully rely on the easily spoofable caller id. Especially, allowing only certain ips or networks to enter a specific context in the dial plan is apparently not possible, at least in the h323 world. Don't know yet about sip or aix, but I guess it's the same since the extension syntax xyz = extension/somevariable limits routing decisions to built in variables, where ip related info is simply missing, at least as far as I can see (you are wholeheartedly invited to prove me wrong). Question hence: did anybody tackle those issues anyway, maybe on code level (patch/extra module)? Are plans underway to fix that stuff? I just can't believe that, if my above statements were right, anyone would expose an * server to the internet and still feel secure, especially if that server allows connections to billable services (like even bandwidth usually is) ... Any info highly appreciated. Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call routing based on remote ip address.
Bruno Hertz wrote: Especially, allowing only certain ips or networks to enter a specific context in the dial plan is apparently not possible, at least in the h323 world. Don't know yet about sip or aix, but I guess it's the same since the extension syntax xyz = extension/somevariable limits routing decisions to built in variables, where ip related info is simply missing, at least as far as I can see (you are wholeheartedly invited to prove me wrong). I don't know about chan_h323 because I don't use it, but certainly the other IP channel drivers allow you to control access to your type=user entries via many means: IP, password, RSA key, etc. There is no way for any user to get to the _dialplan_ if they can't authenticate as a user, so there is no need for this level of access control in the dialplan itself. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call routing based on remote ip address.
On Tue, 2004-12-21 at 11:23 -0700, Kevin P. Fleming wrote: I don't know about chan_h323 because I don't use it, but certainly the other IP channel drivers allow you to control access to your type=user entries via many means: IP, password, RSA key, etc. There is no way for any user to get to the _dialplan_ if they can't authenticate as a user, so there is no need for this level of access control in the dialplan itself. Many thanks Kevin, so it's done via channel config and not dial plan routing, I then presume due to protocol specific auth schemes. Very good to know that. In the meantime, I digged further and found that at least with oh323 ip address based routing can be done with gotoif and the oh323_raddr variable. This thankfully invalidates my original complaint, too :) But I must say I'd still very much like if an identity concept related to ip networking was available at dial plan level. Hopefully it's gonna happen sometime in the future. Anyway, consider this as settled as far as I am concerned, and again thanks very much. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call routing based upon callerID
Hello,
There is an agi script for this, but I use goto's in the extensions.conf. Its
not terribly efficient, but it gets the job done.
I tried the blacklist but it only payed attention to the callerid. The number
was completely ignored. I could only put in one WIRELESS CALLER, and there
are a lot.. so I googled, irc'd and moved on...
[globals]
MYFAMILY=413999 || 413999 || 617999;the 9's are real
numbers in mine
GIRLFAM=415999 ;the || = or
GIRLWORK=6509XX || 41599X ;the 9's should be real again, X's are
wild
BROTHERWORK=415XXX
CELL=1415999
[macro-callfwd]
exten = s,1,SetCallerID(${FWDCID});this no workey yet
exten = s,2,SetCIDName(${FOO}})
exten = s,3,Dial(IAX2/[EMAIL PROTECTED]/${ARG1},8,tr) ;call me
exten = s,4,Hangup
[incoming]
exten = s,1,Wait(0)
exten = s,2,GotoIf($[${CALLERIDNUM} = ${FOO}]?9:3) ;necessary
exten = s,3,GotoIf($[${CALLERIDNUM} = ${TESTVAR}]?103:4) ; test variable
exten = s,4,GotoIf($[${CALLERIDNUM} = ${MYFRIENDS}]?103:5)
exten = s,5,GotoIf($[${CALLERIDNUM} = ${MYFAMILY}]?103:6)
exten = s,6,GotoIf($[${CALLERIDNUM} = ${GIRLFAM}]?103:7)
exten = s,7,GotoIf($[${CALLERIDNUM} = ${BROTHERWORK}]?103:8)
exten = s,8,GotoIf($[${CALLERIDNUM} = ${GIRLWORK}]?103:9)
exten = s,9,Dial(Zap/g2,20,tr) ;either I don't know the caller, or its a
business call
exten = s,10,Macro(callfwd,${CELL},10,tr) ;one fxo so forward them to cell
across the net
exten = s,11,Answer
exten = s,12,Playback(business/showmemoney) ; custom voicemail
exten = s,13,VoiceMail2(s494)
exten = s,14,Hangup
exten = s,103,Dial(Zap/g2,20,tr) ; I should use a distinctive ring
exten = s,104,Dial(IAX2/[EMAIL PROTECTED]/${CELL},10,tr) ; different syntax
than the macro
exten = s,105,Answer ; same outcome
as the macro
exten = s,106,Playback(personal/ola)
exten = s,107,Voicemail2(s344)
exten = s,108,Hangup
Hope I helped, It took me a while.
Adrian
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RE: [Asterisk-Users] Call routing based upon callerID
Well, it is what he asked for, perhaps it was because I didn't do all of it for him, since I wanted him to learn rather than just copy... Let me explain: John : The scenario is that I want all calls originating from number x to be routed to a particular extension exten = s/12345678,1,congestion This means that any call with callerid of 12345678 will execute the congestion application. The application at the end can be anything at all, a Goto, a dial whatever... John: those from yy to another exten = s/24681012,1,Dial(SIP/phone2) the same as above but for 24681012 and that it run the dial application.. John: and anything else to a third. exten = s,1,Dial(SIP/phone1,30) neither of the above 2 were met, (no callerid or callerid not matching) so dial a different phone... If I've suddenly become unable to understand English then let me knowif not then hopefully this explains how to use the feature... Andy On 29/03/2004 at 20:42 Matthew B Marlowe wrote: I don't think this is what he was trying to do - And if it was, well then I'm trying to do something else. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Powell Sent: Monday, March 29, 2004 7:16 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call routing based upon callerID John, This is referenced as the anti ex-girlfriend feature... example: exten = s/12345678,1,congestion exten = s/24681012,1,Dial(SIP/phone2) exten = s,1,Dial(SIP/phone1,30) also check page 31 of the handbook... hth Andy *** REPLY SEPARATOR *** On 29/03/2004 at 20:34 John F. Baird wrote: Hi, I've search and though I've found a few references I have not been able to find any concrete examples of * routing a call based upon the caller ID. The scenario is that I want all calls originating from number x to be routed to a particular extension, those from yy to another and anything else to a third. Can someone please provide a reference to samples or tell me if it cannot be done easily? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call routing based upon callerID
Thanks to all those who replied. The anti ex-girlfriend facility seems to be doing just what I was after. Maybe I just didn't have enough ex-girlfriends; or maybe just not enough that turned into stalkers. Regards, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call routing based upon callerID
On Tuesday 30 March 2004 06:25, John F. Baird wrote: Thanks to all those who replied. The anti ex-girlfriend facility seems to be doing just what I was after. Maybe I just didn't have enough ex-girlfriends; or maybe just not enough that turned into stalkers. The anti-ex-girlfriend facility is actually more humorous than the example provided: the original usage was to forward the call to the ex-gf's cell when the home number called and to the home number when the cell number called. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call routing based upon callerID
Hi, I've search and though I've found a few references I have not been able to find any concrete examples of * routing a call based upon the caller ID. The scenario is that I want all calls originating from number x to be routed to a particular extension, those from yy to another and anything else to a third. Can someone please provide a reference to samples or tell me if it cannot be done easily? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call routing based upon callerID
There are specific examples of call routing based on CallerID in the handbook. You can read the handbook at this URL: http://www.digium.com/handbook-draft.pdf. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John F. Baird Sent: Monday, March 29, 2004 4:34 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call routing based upon callerID Hi, I've search and though I've found a few references I have not been able to find any concrete examples of * routing a call based upon the caller ID. The scenario is that I want all calls originating from number x to be routed to a particular extension, those from yy to another and anything else to a third. Can someone please provide a reference to samples or tell me if it cannot be done easily? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.645 / Virus Database: 413 - Release Date: 3/28/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.645 / Virus Database: 413 - Release Date: 3/28/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call routing based upon callerID
John, This is referenced as the anti ex-girlfriend feature... example: exten = s/12345678,1,congestion exten = s/24681012,1,Dial(SIP/phone2) exten = s,1,Dial(SIP/phone1,30) also check page 31 of the handbook... hth Andy *** REPLY SEPARATOR *** On 29/03/2004 at 20:34 John F. Baird wrote: Hi, I've search and though I've found a few references I have not been able to find any concrete examples of * routing a call based upon the caller ID. The scenario is that I want all calls originating from number x to be routed to a particular extension, those from yy to another and anything else to a third. Can someone please provide a reference to samples or tell me if it cannot be done easily? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call routing based upon callerID
John, It can be done. I have not successfully done it yet though, although I have not researched it too much I would like to do it as well so if you find any information on it and get it working. Let me know. :) I know for a fact * does support it though. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John F. Baird Sent: Monday, March 29, 2004 5:34 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call routing based upon callerID Hi, I've search and though I've found a few references I have not been able to find any concrete examples of * routing a call based upon the caller ID. The scenario is that I want all calls originating from number x to be routed to a particular extension, those from yy to another and anything else to a third. Can someone please provide a reference to samples or tell me if it cannot be done easily? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call routing based upon callerID
I don't think this is what he was trying to do - And if it was, well then I'm trying to do something else. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Powell Sent: Monday, March 29, 2004 7:16 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call routing based upon callerID John, This is referenced as the anti ex-girlfriend feature... example: exten = s/12345678,1,congestion exten = s/24681012,1,Dial(SIP/phone2) exten = s,1,Dial(SIP/phone1,30) also check page 31 of the handbook... hth Andy *** REPLY SEPARATOR *** On 29/03/2004 at 20:34 John F. Baird wrote: Hi, I've search and though I've found a few references I have not been able to find any concrete examples of * routing a call based upon the caller ID. The scenario is that I want all calls originating from number x to be routed to a particular extension, those from yy to another and anything else to a third. Can someone please provide a reference to samples or tell me if it cannot be done easily? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call routing based on dnis
Hi, Is it to possible to route incomming call using dnis information to specific extension or section in the extensions.conf ?? like my users dial my asterisk box using different toll free numbers, but i can't offer them unique services. Plz suggest that if it is possible ? if possible then any example ... Regards Apna Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo.
Re: [Asterisk-Users] call routing based on dnis
azher: simple to do, assuming numbers are being passed through on dnis, in the relevant context (from zapata) put exten = 6601122,1,Hangup #users dialing here bye-bye exten = 5551122,1,Playback(beep) #users here (pun intended) beep - wasim On Sun, 17 Aug 2003, Azher Amin wrote: Is it to possible to route incomming call using dnis information to specific extension or section in the extensions.conf ?? like my users dial my asterisk box using different toll free numbers, but i can't offer them unique services. Plz suggest that if it is possible ? if possible then any example ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call routing question
I have a quick call routing question that I'm sure is simple, but for the
life of me I can't figure out.
For example, someone dials 94162384000 asterisk trys our first call route
(our sip trunk) as per the extension rule below:
exten = _9NX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
However, this call fails because 94162384000 is one of our phone lines and
our SIP gateway detects a loop and returns a SIP 503 message. Is there a
way to have asterisk stip the '9' and try it as a local extension call as if
the user didn't dial 9? I try this (see below) and it failed:
exten = _9NX,2,Dial(${EXTEN:1})
Thanks in advance, I'm sure it's a simple problem and I'm just missing
something...
Regards,
M. Gamble
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RE: [Asterisk-Users] Call routing question
Hi Matt! :)
You can use the Local channel driver:
exten = _9NXXNXX,1,Dial(Local/${EXTEN:[EMAIL PROTECTED])
Where ${CONTEXT} is set to the local context you want to use.
-wade
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matthew M. Gamble
Sent: Thursday, August 07, 2003 8:36 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Call routing question
I have a quick call routing question that I'm sure is simple, but for the
life of me I can't figure out.
For example, someone dials 94162384000 asterisk trys our first call route
(our sip trunk) as per the extension rule below:
exten = _9NX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
However, this call fails because 94162384000 is one of our phone lines and
our SIP gateway detects a loop and returns a SIP 503 message. Is there a
way to have asterisk stip the '9' and try it as a local extension call as
if
the user didn't dial 9? I try this (see below) and it failed:
exten = _9NX,2,Dial(${EXTEN:1})
Thanks in advance, I'm sure it's a simple problem and I'm just missing
something...
Regards,
M. Gamble
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RE: [Asterisk-Users] Call routing question
On Thu, 2003-08-07 at 19:55, Wade Weppler wrote:
Hi Matt! :)
You can use the Local channel driver:
exten = _9NXXNXX,1,Dial(Local/${EXTEN:[EMAIL PROTECTED])
Where ${CONTEXT} is set to the local context you want to use.
What would be wrong with just a simple Goto?
exten = _9NX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9NX,2,Goto(${EXTEN:1}|1)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matthew M. Gamble
Sent: Thursday, August 07, 2003 8:36 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Call routing question
I have a quick call routing question that I'm sure is simple, but for the
life of me I can't figure out.
For example, someone dials 94162384000 asterisk trys our first call route
(our sip trunk) as per the extension rule below:
exten = _9NX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
However, this call fails because 94162384000 is one of our phone lines and
our SIP gateway detects a loop and returns a SIP 503 message. Is there a
way to have asterisk stip the '9' and try it as a local extension call as
if
the user didn't dial 9? I try this (see below) and it failed:
exten = _9NX,2,Dial(${EXTEN:1})
Thanks in advance, I'm sure it's a simple problem and I'm just missing
something...
Regards,
M. Gamble
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--
Steven Critchfield [EMAIL PROTECTED]
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