Re: [asterisk-users] Dial Plan Issues
the following two lines exist in the extensions_additional.conf [from-max]exten = _X,1,Answerexten = _X,n,Queue(8000,tr,,) and it DOES exist in the output of the 'show dialplan' [ Context 'from-max' created by 'pbx_config' ] '_X' = 1. Answer() [pbx_config]2. Queue(8000|tr||) [pbx_config] yet my system doesn't use it to route regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sun, 28 Sep 2008 23:31:46 +0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial Plan Issues On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote: this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! What is the dialplan? ls -ld /etc/asterisk /etc/asterisk/extensions.conf And what is the contents of extensions.conf ? What is the output of 'dialplan show' from the CLI? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 _ Want to do more with Windows Live? Learn “10 hidden secrets” from Jamie. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Dial Plan Issues
On Mon, 2008-09-29 at 14:51 +, Tariq .. wrote: the following two lines exist in the extensions_additional.conf [from-max] exten = _X,1,Answer exten = _X,n,Queue(8000,tr,,) and it DOES exist in the output of the 'show dialplan' [ Context 'from-max' created by 'pbx_config' ] '_X' = 1. Answer() [pbx_config] 2. Queue(8000|tr||) [pbx_config] yet my system doesn't use it to route regards Tariq-- Maybe I missed a message or something, but I don't see a response to Tzafrir's request to see /etc/asterisk/extensions.conf. extensions_additional.conf is not extensions.conf; and unless extensions.conf includes it, it will never be a part of your dialplan. You did mention that you were using trixbox in your original question, so we referred you to a trixbox mailing list, because rumors have it that trixbox does complicated things in their dialplan to accomplish their goals, and most folks in this mailing list (but not all) don't play much with trixbox. But if you are not using trixbox, then you might look in your extensions.conf to answer these questions. Another resource you have to investigate the dialplan is in the CLI of asterisk; you can say dialplan show, or dialplan show from-max to see if the from-max context has been included. when the pbx_config module (module load pbx_config.so) loads, it will read in /etc/asterisk/extensions.conf; if it is not there, that module will not complete the loading process. If want us to evaluate why your dialplan is not working, show us the dialplan in extensions.conf. murf __ Date: Sun, 28 Sep 2008 23:31:46 +0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial Plan Issues On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote: this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! What is the dialplan? ls -ld /etc/asterisk /etc/asterisk/extensions.conf And what is the contents of extensions.conf ? What is the output of 'dialplan show' from the CLI? -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Dial Plan Issues
no.. it's directly connected to the internet.. it's not an issue of accepting
calls.. see.. the problem is the call gets to the server.. the server tries to
route it..
but as if the dial plan is not there.. it rejects the call because it doesn't
know what to do with it..
for example of my SIP.Conf
[5003]
type=peer
qualify=yes
port=5060
nat=yes
host=HOSTIP
allow=all
dial=SIP/5003
context=from-smarttech
canreinvite=no
call-limit=50
deny=0.0.0.0/0.0.0.0
permit=HOSTIP/255.255.255.255
Extensions.conf
[from-smarttech]
exten = fax,1,Goto(ext-fax,in_fax,1)
exten = s,n,Set(__FROM_DID=${EXTEN})
exten = s,n,Gosub(app-blacklist-check,s,1)
exten = s,n,GotoIf($[ ${CALLERID(name)} != ] ?cidok)
exten = s,n,Set(CALLERID(name)=${CALLERID(num)})
exten = s,n(cidok),Noop(CallerID is ${CALLERID(all)})
exten = s,n,Set(__CALLINGPRES_SV=${CALLINGPRES_${CALLINGPRES}})
exten = s,n,SetCallerPres(allowed_not_screened)
exten = s,n,Goto(ext-queues,8004,1)
let's say smarttech is a voip provider.. which forwards calls to my user on
their system .. now my server is supposed to route those calls according to the
dial plan..
the same exact settings worked like magic on another server.. but on this
server.. it just as if the context and the dial plan does not exist.!!!
any idea?
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
Date: Fri, 26 Sep 2008 11:55:45 -0500From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: Re: [asterisk-users] Dial Plan Issues
Steve Murphy wrote:
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote:
Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of
my servers is not allowing the other to send calls while it is possible in the
opposit direction..
i have the same exact settings for the extensions.conf
i tried with another friend of mine.. and connected to his server.. and it
didn't allow him to send me calls..
so my question is..
is it possible that my server is not accepting any context ? it only runs the
ones that come default with Trixbix.. like chanspy, ext-local, from-trunk...
and so on..
what can i do to avoide this problem?? i can't rebuild a new box this one is a
production server and i wasn't making tests.. i was connecting two of my
employer's servers with each other..
regards
Tariq--
You might try a trixbox users mailing list.
There might be a few trixbox users hanging around in
this group who might be able to help, but your
chances are much better in that list.
murf
The server that is not accepting calls is not behind a NAT firewall by any
chance is it?
_
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Re: [asterisk-users] Dial Plan Issues
This is a better question asked on a Fonality list. Maybe they have a manual. Thanks, Steve Totaro On Thu, Sep 25, 2008 at 10:21 AM, Tariq .. [EMAIL PROTECTED] wrote: Greetings, i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox.. i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. i have the same exact settings for the extensions.conf i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. so my question is.. is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other.. regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ Want to do more with Windows Live? Learn 10 hidden secrets from Jamie. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns%21550F681DAD532637%215295.entry?ocid=TXT_TAGLM_WL_domore_092008 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Dial Plan Issues
this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sun, 28 Sep 2008 10:00:56 -0400From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [asterisk-users] Dial Plan Issues This is a better question asked on a Fonality list. Maybe they have a manual.Thanks,Steve Totaro On Thu, Sep 25, 2008 at 10:21 AM, Tariq .. [EMAIL PROTECTED] wrote: Greetings,i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction..i have the same exact settings for the extensions.confi tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls..so my question is..is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on..what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other..regardsAHD Tarek SawahIntegrated Digital SystemsCCNA, MCSE, RHCE, VoIPSyria: +963 944 618286USA: +1 347 562 2308_Want to do more with Windows Live? Learn 10 hidden secrets from Jamie.http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008___-- Bandwidth and Colocation Provided by http://www.api-digital.com --AstriCon 2008 - September 22 - 25 Phoenix, ArizonaRegister Now: http://www.astricon.netasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Thanks,Steve Totaro1.888.777.18881.240.938.1212 (cell) _ See how Windows Mobile brings your life together—at home, work, or on the go. http://clk.atdmt.com/MRT/go/msnnkwxp1020093182mrt/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Dial Plan Issues
On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote: this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! What is the dialplan? ls -ld /etc/asterisk /etc/asterisk/extensions.conf And what is the contents of extensions.conf ? What is the output of 'dialplan show' from the CLI? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Dial Plan Issues
Steve Murphy wrote: On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote: Greetings, i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox.. i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. i have the same exact settings for the extensions.conf i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. so my question is.. is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other.. regards Tariq-- You might try a trixbox users mailing list. There might be a few trixbox users hanging around in this group who might be able to help, but your chances are much better in that list. murf The server that is not accepting calls is not behind a NAT firewall by any chance is it? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Dial Plan Issues
Greetings, i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox.. i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. i have the same exact settings for the extensions.conf i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. so my question is.. is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other.. regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ Want to do more with Windows Live? Learn “10 hidden secrets” from Jamie. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Dial Plan Issues
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote: Greetings, i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox.. i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. i have the same exact settings for the extensions.conf i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. so my question is.. is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other.. regards Tariq-- You might try a trixbox users mailing list. There might be a few trixbox users hanging around in this group who might be able to help, but your chances are much better in that list. murf AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
