similar to: Dial plan Question

Is what I am trying to do in this context possible. That is changing the incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not preceeded by a "1" I want to add a "1". Often calls come in without the preceeding "1" and this plays havoc with my redial if the 3 digit area code matches a local 3 digit extension. All my outside calls are 10 digits or 1+10

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

Does it matter if you use upper or lowercase rules - I.E. - "x" vs. "X" or mix them? Not that I would do that as a rule but sometimes you make mistakes! Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

Ok asked this earlier with no response so I will phrase it a different way. I am sure someone had to deal with this and there is a "best way." I want to let Asterisk make the decision on best path based on local exchange - xxx-yyy - where xxx is one of my local area codes and xxx is exchange designator. The problem is that the list is rather large. Maybe 50-100. The idea is that I can

On systems even key systems it is customary to have an 'internal' dial tone. Since Asterisk simply ignores the 9 and keeps the tone going it is hard to tell for some 'new users' if they can make a call. My first idea was to change the generated dial tone via source. Then if the user presses 9 go to a different context where I would record about 30 seconds of the normal dial

Did not know how to make up a subject line for this. I have a dial plan that allows a caller can try my cell phone. And that's fine. If the call cannot be made, it sends caller back to voice menu. However, I'd like a way for the caller to elect to go back to the voice menu, if they end up getting the cell phone voice mail. Is that possible? joe a.

I'm experiencing a strange problem. My inbound calls are hanging up right after Background() message even though response timeout is set to 10 sec. [voicepulseincoming] exten=>_X.,1,Answer exte=>_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(9:00-15:00|fri|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(*|*|*|*?after-business-hours,s,1)

Does anyone know the maximum number of digits for an international phone number? Doing some searching, it looks like 16 numbers including the "011" is the maximum number, because 17 is just not found: OK: 1234567890123456 http://www.google.com/search?q=011XXXXXXXXXXXXX Not OK: 12345678901234567

Hallo Group, I have basically set up a small asterisk system, which ahs 4 peers: * registers at 2 Sipgates * 2 hardware phones connected to it Both Hardware phones can phone outwards(cheaper sipgate is selected with dialplan) Calls from both sipgates make my hardware phones ring But here comes the challenges: Is it possible to configure asterisk in such a way that in the phone: * there are

Hi All; I'm a newbie so please be gentle. I'm a new * user and am using it to control the 3 IP phones in my house. I'm using asterisk because I enjoy the flexibility and I'm sort of a tinkerer. Here's my question: Everyone has used the "dial by directory" function where you dial the user's name to connect to that extension. Instead of an inward dial,

Hi Guys, I Have this peers on my sip.conf [provider-302333-3000] type=friend context=provider secret=xpto username=3023333000 host=sip.provider.com fromuser=3023333000 insecure=very canreinvite=no [provider-302222-3001] type=friend context=provider secret=xpto username=3022223001 host=sip.provider.com fromuser=3022223001 insecure=very canreinvite=no I Have in my sip.conf two extension 3000

Can anyone suggest a source for a free robot dialer or examples? I need to do some non-commercial auto dialing using Asterisk. Simple phone numbers in a file, line by line format. I found one called AstAutoDiaker but I was not able to get it to work and it appears to not be supported - no email response from author. Doug

My SIP phones can dial out through Sipura SPA3k to POTS for local and 911 calls _but_ incoming POTS calls are being swallowup somehow. Am I on the right track with the code snippit below? sip.conf: --------- In sip.conf the following code is _supposed_ to ring the SIP phones when a POTS line call comes in through Sipuara to Asterisk. [spa3k-pstn-in] ; Pots-line-in from Sipura ; If

Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at

http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns says: ======== ! wildcard, matches zero or more characters immediately (only Asterisk 1.2 and later, see note) Note: The exclamation mark wildcard, which is available only in Asterisk 1.2 and later, behaves specially - it will match as soon as can without waiting for the dialing to complete, but

Hello All, I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not PRI) which is all happily coexisting and all lights are green. The T-1 comes in from the world into a "Shark Box" which splits the T into 384K data and 6 channels voice. The data side is working great. The voice side, not so great. It was originally broken out to 6 pots line and Verizon came back

Hi all, I'm using asterisk in real time mode...My extensions.conf table contains: [default] switch => Realtime/default at extensions I have added the following to extensions.conf table; context:micho exten: _X. priority: 1 app:Dial appdata: SIP/00XXXXXX at PSTN GAteway Asterisk server is connected succeffully to database...As soon as i make a call i got the following error message:

Hello ppl, Any way to specify alphabetical exten patterns in the dialplans on Asterisk? All my users would have alpha/numerical ids. I don't want to add a line for every user in my dialplans. I searched around, but couldn't get anything useful. Any way to get around this? Thanks in advance - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

On Thu, Jun 4, 2015 at 5:33 PM, Reid Kleckner <rnk at google.com> wrote: > On Thu, Jun 4, 2015 at 5:17 PM, Teresa Johnson <tejohnson at google.com> wrote: >> >> Agreed. Although I assume you mean invoke the new pass under a >> ThinLTO-only option so that avail extern are not dropped in the >> compile pass before the LTO link? > > > No, this pass