similar to: Pattern-matching in the dial-plan

Hey all, I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming into it from a Meridian-switch. The incoming numbers on this PRI all start with the same digit and the last part of the dialled number is signalled to Asterisk digit by digit, until Asterisk signals that the number is complete and the call rings. All works well, unless I have 2 or more numbers which start with the same

Hi all! I am in the same situation here in Sweden, but I just got the X100 card and don t get it to work with DTMF caller ID. Is the only way to solve this to buy a TDM400 card w FXO modules? Is it just a driver question or a HW issue with the X100 card? Where to find dok about the new keywords in conf files like "cidstart=polarity" in zapata.conf? rgds Gunnar >>>

Hi i am using pap2 phone adaptors as clients to connect to asterisk server i am able to make calls but i cannot access voice mail using phone or start recording while call is in progress and when i place a call to local sip extension there is a long pause ( 15 sec ) before the call gets dialled i assume that the problem would be due to the dial plan in PAP2 if so please help me changing it

Hi, I have a Linksys PAP2-NA connectd to my asterisk. I would like the device to add 2 characters in front of the dialled number always when it send the call to my asterisk. I dont know how to do that. I will summarise my requirement. My friend dials 1-210-1234345, i want the asterisk to get 55-1-210-1234345. Can someone help me to add this dialplan. Thanks in advance Dan -------------- next

I've got two patterns I want to match on making an outgoing call... (one day - to do Least Cost Routing for Cell/Mobile calls) Firstly - I prefer '0' rather than '9' to get an outside line... Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084) or its just another number to dial... I added the following... the playback just advises me which 'route' is

Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including

dear all, was hoping someone could give me instruction on the syntax of extension pattern matching for letters the proposed 'dial plan' is one where any letter in the dialled digits causes the pbx to assume we are dilaling a sip url and as such forward to the appropraite sip service provider was hoping to avoid the plan in john todd's example that assumes anything prefixed with 3 is

Can someone please explain how to match a + character in a dial plan (so that I can swap it for the "00" country escape code). In Europe at least the + is a common shortcut for the international prefix (which is "00" in my country). However, my trunk chokes on the + character and all my speed-dials are setup with a + at the start of them... Trying to fix the phone rather

I'm trying to find a way in extensions.conf to match ANYTHING dialled, including characters such as *. The following works for numbers... exten => _X.,1,AGI(script) but doesn't catch when someone dialls * first. I tried this: exten => _.,1,AGI(script) which catches when someone dials say, *123 for example, but after the AGI script terminates, Asterisk executes it again with

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

Hi, We have been migrating our PBX system from Asterisk 1.8 and chan_sip to Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have stumbled on a behaviour difference I don't like. With chan_pjsip when a phone went unexpectedly offline (Ethernet cable disconnected) Asterisk would detect this quickly (through the 'qualify' pings), mark the phone as 'Unavailable' and

Yo all, The PSTN access-number for the Dutch IAXTel <-> PSTN-gateway has changed. The new number is: +31 20 3987 567. Calling from IAXTel to Dutch toll-free PSTN-numbers is still done in the same way, by calling "31800<rest of number>". Mark: Could you please update your web-sites to reflect this change? The old number is mentioned on "http://www.gnophone.com/",

Hey all, I recently noticed that DTMF caller-ID was implemented in CVS, so I requested the service from my telco (the Dutch KPN) and tried to get it going in Asterisk (current CVS), without success so far. This system has 1 X100P, 2 TDM400P's with 4 FXS-modules each and 2 HFC-PCI ISDN-cards (zaphfc-driver) in it. The analog line I'm trying to get caller-ID working on is obviously on the

Using FWD and accessing it via this extension: exten => _*8.,1,Dial(SIP/${EXTEN:2}@fwd.pulver.com) This works *perfectly* with SIP phones. However with a regular phone plugged into an FXS card (PhoneJack PCI in my case) the '.' traps the first number dialled after *8 and tries calling that. I've tried setting a digit timeout but it doesn't seem to help. Changing that to

Hello This is a fun one for the list... Twice now, the Police have contacted us to say they have had a silent call then hangup from our landline number to the 999 service. As a matter of course, they follow up these calls in case someone is in distress. Nobody here was in distress - well, no more than normal! The Police aren't hugely happy when we tell them it must be a mistake. Thing

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Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves,

I have the following dialplan: ; calls prefix by '8' are recorded exten = _8[01]./_251,1,Set(something=shortened) exten = _8[01]./_251,n,Set(WAV=filename) exten = _8[01]./_251,n,Monitor(wav,${WAV},mb) exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g) exten = _8[01]./_251,n,System(send-recorded-conversation ${WAV}.wav ${EXTEN:1} emailaddr) exten = _8[01]./_251,n,Hangup() The idea is that

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden

Hey all, As there seem to be some problems with DTMF-signalling between chan_sip and several clients, due to which many could not properly dial a number at the dial-tone of the XS4ALL-gateway at FWD-number "42442", I've now arranged for a prefix on FWD for this gateway.