Hello When a user makes a call to an Asterisk system, He dials a number . We need to know that dialed number. We can get the dialed number by using CALLERID(dnid) and we can get the CLI information using CALLERID(num). I am facing problem while getting the number dialed. if the user is using SIP phone then we can get the number dialed. but if it using PSTN then we are unable to get the number dialed using CALLERID(dnid). Do any other way exists to find out? Or this is some thing wrong with the PSTN. -- Kind Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091123/cb1cbf6e/attachment.htm
ABBAS SHAKEEL wrote:> I am facing problem while getting the number dialed. if the user is > using SIP phone then we can get the number dialed. but if it using PSTN > then we are unable to get the number dialed using CALLERID(dnid). Do any > other way exists to find out? Or this is some thing wrong with the PSTN.DNID breakage is a long-standing Asterisk problem. If this is taking place in the context of the dial plan, why not just use ${EXTEN}? -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671
Thanks Alex,
suppose this is the context
[abc]
exten => s,1,Answer();
exten => s,n,Noop(${EXTEN});
exten => s,n,Noop(${CALLERID(dnid)});
I get the following out put
Answer("DAHDI/2-1", "")
NoOp("DAHDI/2-1", "s") in new stack
NoOp("DAHDI/2-1", "")
But i need the number that the user is dialing not the "s";
On Mon, Nov 23, 2009 at 3:24 PM, Alex Balashov <abalashov at
evaristesys.com>wrote:
> ABBAS SHAKEEL wrote:
>
> > I am facing problem while getting the number dialed. if the user is
> > using SIP phone then we can get the number dialed. but if it using
PSTN
> > then we are unable to get the number dialed using CALLERID(dnid). Do
any
> > other way exists to find out? Or this is some thing wrong with the
PSTN.
>
> DNID breakage is a long-standing Asterisk problem.
>
> If this is taking place in the context of the dial plan, why not just
> use ${EXTEN}?
>
> --
> Alex Balashov - Principal
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Kind Regards
Shakeel Abbas
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I am curious what happens if you do the following instead:
[abc]
exten => _.,1,Answer
exten => _.,n,NoOp(${EXTEN})
ABBAS SHAKEEL wrote:
> Thanks Alex,
>
> suppose this is the context
>
> [abc]
> exten => s,1,Answer();
> exten => s,n,Noop(${EXTEN});
> exten => s,n,Noop(${CALLERID(dnid)});
>
> I get the following out put
>
>
> Answer("DAHDI/2-1", "")
> NoOp("DAHDI/2-1", "s") in new stack
> NoOp("DAHDI/2-1", "")
>
> But i need the number that the user is dialing not the "s";
>
>
>
>
> On Mon, Nov 23, 2009 at 3:24 PM, Alex Balashov
> <abalashov at evaristesys.com <mailto:abalashov at
evaristesys.com>> wrote:
>
> ABBAS SHAKEEL wrote:
>
> > I am facing problem while getting the number dialed. if the user
is
> > using SIP phone then we can get the number dialed. but if it
> using PSTN
> > then we are unable to get the number dialed using CALLERID(dnid).
> Do any
> > other way exists to find out? Or this is some thing wrong with
> the PSTN.
>
> DNID breakage is a long-standing Asterisk problem.
>
> If this is taking place in the context of the dial plan, why not just
> use ${EXTEN}?
>
> --
> Alex Balashov - Principal
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> Kind Regards
> Shakeel Abbas
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Thanks Alex Thats just a test code>>exten => _.,1,Answer >>exten => _.,n,NoOp(${EXTEN})This also have the same output as that of previous. On Mon, Nov 23, 2009 at 3:46 PM, Alex Balashov <abalashov at evaristesys.com>wrote:> I am curious what happens if you do the following instead: > > [abc] > > exten => _.,1,Answer > exten => _.,n,NoOp(${EXTEN}) > > ABBAS SHAKEEL wrote: > > > Thanks Alex, > > > > suppose this is the context > > > > [abc] > > exten => s,1,Answer(); > > exten => s,n,Noop(${EXTEN}); > > exten => s,n,Noop(${CALLERID(dnid)}); > > > > I get the following out put > > > > > > Answer("DAHDI/2-1", "") > > NoOp("DAHDI/2-1", "s") in new stack > > NoOp("DAHDI/2-1", "") > > > > But i need the number that the user is dialing not the "s"; > > > > > > > > > > On Mon, Nov 23, 2009 at 3:24 PM, Alex Balashov > > <abalashov at evaristesys.com <mailto:abalashov at evaristesys.com>> wrote: > > > > ABBAS SHAKEEL wrote: > > > > > I am facing problem while getting the number dialed. if the user > is > > > using SIP phone then we can get the number dialed. but if it > > using PSTN > > > then we are unable to get the number dialed using CALLERID(dnid). > > Do any > > > other way exists to find out? Or this is some thing wrong with > > the PSTN. > > > > DNID breakage is a long-standing Asterisk problem. > > > > If this is taking place in the context of the dial plan, why not just > > use ${EXTEN}? > > > > -- > > Alex Balashov - Principal > > Evariste Systems > > Web : http://www.evaristesys.com/ > > Tel : (+1) (678) 954-0670 > > Direct : (+1) (678) 954-0671 > > > > _______________________________________________ > > -- 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 > > > > > > > > > > -- > > Kind Regards > > Shakeel Abbas > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- 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 > > > -- > Alex Balashov - Principal > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > > _______________________________________________ > -- 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 >-- Kind Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091123/25fb3321/attachment.htm
Xavier Mesquida
2009-Nov-23 11:14 UTC
[asterisk-users] Interconnect Asterisk with another PBX
Hi, I want
to interconnect a Alcatel OmniPCX PBX with SIP support with an Asterisk PBX. My
intention is Alcatel PBX manage all external calls and analog extensions and
Asterisk
manage all the SIP users (because I have to pay for every SIP license in
Alcatel PBX and I can?t edit configuration or password in that PBX)
What?s the
best way to interconnect the 2 PBX? With SIP, with a FXO interface or FXS? How
can I do that? Thanks
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Alex Balashov
2009-Nov-23 11:17 UTC
[asterisk-users] Interconnect Asterisk with another PBX
PRI is likely the simplest and most reliable. Xavier Mesquida wrote:> > Hi, I want to interconnect a Alcatel OmniPCX PBX with SIP support with > an Asterisk PBX. My intention is Alcatel PBX manage all external calls > and analog extensions and Asterisk manage all the SIP users (because I > have to pay for every SIP license in Alcatel PBX and I can?t edit > configuration or password in that PBX) > > What?s the best way to interconnect the 2 PBX? With SIP, with a FXO > interface or FXS? How can I do that? 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-- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671
Ryan Wagoner
2009-Nov-23 16:17 UTC
[asterisk-users] Interconnect Asterisk with another PBX
Either use SIP or PRIs to do the integration. FXO and FXS interfaces are a single port, where as a PRI will provide you with 23 channels. Use QSIG signaling over the PRI so Caller ID names will show between the systems. I just integrated a Toshiba CIX with Asterisk due to the cost for SIP licensing and the reliability of the Toshiba VOIP Phones. They were having hardware failures every few months. I went with Sangoma PRI cards using QSIG. Everything has been working great and I have rolled out 12 Snom 370 phones to work with the 150 Toshiba Digital phones. To the end users the experience is seamless as they can 4 digit dial any extension and the call will be routed to the correct system. This does take a bit of duplicate setup on the two systems, but was worth the hassle for the end result. Ryan On Mon, Nov 23, 2009 at 6:17 AM, Alex Balashov <abalashov at evaristesys.com> wrote:> PRI is likely the simplest and most reliable. > > Xavier Mesquida wrote: > >> >> Hi, I want to interconnect a Alcatel OmniPCX PBX with SIP support with >> an Asterisk PBX. My intention is Alcatel PBX manage all external calls >> and analog extensions and Asterisk manage all the SIP users (because I >> have to pay for every SIP license in Alcatel PBX and I can?t edit >> configuration or password in that PBX) >> >> What?s the best way to interconnect the 2 PBX? With SIP, with a FXO >> interface or FXS? How can I do that? 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 > > > -- > Alex Balashov - Principal > Evariste Systems > Web ? ? : http://www.evaristesys.com/ > Tel ? ? : (+1) (678) 954-0670 > Direct ?: (+1) (678) 954-0671 > > _______________________________________________ > -- 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