similar to: Redirecting incoming PRI to PSTN

Hello, I'm trying to receive a fax to my inbound number from voipbuster. Asterisk receives the call and starts the rxfax application successful, but then nothing happens. The calling party is still hearing a ringing tone, or sometimes nothing. Voicecalls are working correct and without problems. For testing I've add a local number (300) to the dialplan. When I call this number

Is there a way to log all PRI events to a logfile? Cheers, Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com <http://www.griffin.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041123/d983141f/attachment.htm

Good morning, I have a Bell Canada PRI here (switchtype=national) and I am trying to perform a call-forward-unconditional on one of the DIDs. The idea is that when DID 5551234 receives a call, Asterisk redirects it back out the same PRI to some external number. This is simple enough to do with something along these lines: [PRI] exten => 5551234,1,Set(CALLERID(RDNIS)=${EXTEN}) exten =>

> Hi Damon and others, > > Your example is still doing what I tried already, so eventually the > dial command ends like: > Dial(zap/4/*21*) > or > Dial(zap/4/*31*) > I prefer to use Dial(zap/4/*21*<thenumber>) > or Dial(zap/4/*31*<thenumber>) > > But whatever I try, the error message as in my first post shows up and > the line hangs up before the

Dear all I am going to install 2 port pri card on asterisk but i dont know how to incomming call goes in to IVR and how to route call outside base on pattern match means if some one call on mobile phone then use PRI 1 and if call on landline phon call route through pri 2 how to make dission base on pattern number Rgds satish patel

Hi, I have an odd problem where when a call is made to any landline phone the callerid number shows up on the caller id device with no problems. However when I call a cell phone from Sprint or Verizon it always shows up as unavaliable. I've tried multiple cell phones from each carrier. Though today I called a t-mobile phone and the callerid came through fine. Is this an * level problem

Dear all, In my Asterisk 1.2.17 architecture different levels of permissions are established using different contexts that hierarchically include more permissive contexts until default context is reached. In default context there are only local extensions, only in more restricted contexts there are the PSTN access. So, if some user dials some number, Asterisk looks which context that user

Hello Surprisingly, Google didn't return any thread in this ng about the Obi110 device, which is a new alternative to the 3102: www.nerdvittles.com/?p=720 I'd like some feedback from Asterisk users who have tried the Obi110 to connect it to a landline. Thank you.

Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it hangs up soon as the other party picks up. I have used different IP phones, VSP's and etc.

hello, i've an asterisk box which is connected to an E1/PRI via a TE110P card. incoming calls from mobile phones where the number is transfered as a whole block work fine, but when dialing from an analog or ISDN line to the asterisk box there is a timeout of about 3-5 seconds. originally my incoming context looked like: exten => _X.,1,Dial(SIP/${EXTEN}@domain.tld) so i assumed that the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I am trying to redirect to a local extension the incoming calls that I receive to an account which I have in iptel.org, but when receiving I'm obtaining this error: alderamin*CLI> -- Executing [300 at from-internal:1] Dial("SIP/danib-089f8820", "SIP/300|30|tTrm") in new stack [Feb 19 19:22:50] WARNING[19254]:

Hello, I'm trying to figure out how to get Asterisk to dial an extension when a call comes from the outside and contains the extension already. (Somebody wants to call a user of Asterisk with extension "111" from the outside) For example: I've hooked Asterisk to sipgate.de and received a landline phone number (say 0781205237). Now if you dial 0781205237 and and an extension

All We have a PRI line setup on an asterisk box using TE110P. Both outbound and inbound are working fine BUT the provider claims that all our numbers come prefixed with a '0' (in India a 0 prefix indicates long distance) and that could become an issue with local calls. National Numbering Plan for Landline in India is typically 0+Area Code + phone number. If it's a local number, you

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

hi all, i'd like to know if it would be possible with asterisk (and which hardware would i need) to implement the following (or is it not possible with asterisk - but possible with ...) I'd like to set up something like a "Mobile to Conventionel Network Gateway" - so that users (with there Mobile Phone) which are registered (known Call Number) can Call a Conventionel Network

Hey folks, I have a T100P card with a PRI; when doing outbound dialing over the PRI, I can use SetCIDNum(2024561414|a) to force caller ID to display as "The White House" on a land line. This is apparently done as a reverse lookup by Verizon, as I do not hand the PRI the words "The White House" -- 202-456-1414 is an actual White House number. Using SetCIDName("Flying

Hi all, I spent most of the last weekend testing and trying to diagnose some mostly incoming call issues. I'll start with the easy one in the hopes it might have a positive impact on the others. First- I have an account with IAXTel. I can place calls to other IAXTel subscribers and also through IAXTel to landline toll free numbers and all works great. iax2 show registry shows I am

Has anyone tried integrating Asterisk and Rhetorical's rVoice software? We're evaluating different approaches to system announcements via T2S. Has anyone gone down this route that could give some advice? I've installed festival and wasn't too impressed, the demo one the website seems far better quality and clarity then the defaults in the source package. However I must admit

We're trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number. UK Landline->voipfone->asterisk 1.4->voipfone->UK landline About 1 in 3 times the call at the final landline is silent and we see "RTP Read too short" scrolling on the console log. Where do we

Hello, I am running Asterisk 13.6.0 in an AWS instance, and I set it up with Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the calls actually "reach" the PBX, but for some reason, they are not caught by any of my extensions context. Here's what I observe when I test this from a non-PBX connected E164 number (a landline), say 555-666-1212. My Twilio number is