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I just read: Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: /t/, ''T", "h", "H", "w", "W" or "L" (with multiple arguments). Probably there are more. I had in my memory that "r", "R", "m" would also prevent a

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Hi, I am attempting to setup an asterisk server, current specs: CentOS release 5 (Final) Asterisk 1.4.11 Asterisk-gui checked out from SVN last week I started with a fairly basic setup involving one VOIP provider who provided one dial in number, and a couple of handsets. Config files are below. It was pretty much totally built by Asterisk-gui, except for the fact I had to add

Hello All, I have a small * server in my home office with several IP phones. The system is not fully in service yet as I'm still hunting for a cost effective FXO adapter that I can rely upon for my two primary PSTNs. That said, I'd like to move it into service for another application...which brings up a question. I'd really like to stop making international calls from my cell phone

[0004f2xxxxxx](poly650) defaultuser=0004f2xxxxxx callerid="Front Desk" <1600> mailbox=1600 *setvar=callidnum=1234561600* and from extensions.conf: [outgoing] ; Outbound unrestricted domestic calls exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN} on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.) *exten =>

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Thanks all, I did contact Callcentric about it and their tech support helped meget those headers established. They even helped to troubleshoot Asterisk dialplan. A the end all works as it should. Thank you,Ivan Message: 2 Date: Mon, 15 Oct 2018 23:39:31 +0200 From: Daniel Tryba <daniel at tryba.nl> To: Asterisk Users Mailing List - Non-Commercial Discussion     <asterisk-users at

OK. My basic asterisk install seems to be working. I can get caller ID. My dialplan says: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten => s/9185415897,1,Set(CALLERID(name)=Michael Sullivan) exten => s/9185415897,1,HANGUP(1) exten => s,1,Set(CALLERID(name)=Someone Else) This is for testing. It's supposed to check the caller ID

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Hi, I've configured a new Asterisk 1.6.1 box to replace an old Bristuffed 1.2 Asterisk. Since then, it happens that forwarded calls are not presented the way they used to be. It seems that now, some endpoints are displaying the original caller id (that's what I'm trying to achive), while some are displaying the redirecting number : so if A calls B, B forwards to C depending on where

Update on this: I found that by changing insecure = very to insecure = invite, adding the second trunk no longer stopped calls working. I've read the documentation on this switch and still don't see how it applies/is meant to get used. Anyway, with this change in place, the following may help: asterisk*CLI> sip show registry Host Username

Hello List! We have problems setting the right caller id on outgoing calls. The Asterisk Pbx is located in Bucarest(Romania), our Telco provider is rcs-rds.ro. We have the local telefon number 40787 00-99, associated to our PRI E1 Line. Where 00-99 are the DID numbers available. The telco is aspecting a 3 digit long Callerid from us, for example like "710", for the extension 10.

> -----Original Message----- > From: Richard Lyman [mailto:pchammer@dynx.net] > Sent: Wednesday, December 20, 2006 4:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > Douglas Garstang wrote: > >> -----Original Message----- > >> From: David

Hello! Is there any easy way to use the caller ID "display info" (CALLERID(name) in Asterisk) in dialplan just as we could use the number in: exten => _X./67803287, 1, <action> I have a SIP GSM device, and when a call comes in, it passes me the caller ID like so: -- Sip message Header: From: "67803287" <sip:gsm@192.168.10.1>;tag=... -- Asterisk variables:

correct for a transfer. In the traditional Telco World the src (or A Number) field tends to be both the callerid of the customer and an identifier that ties the CDR to the customer for billing purposes. With Asterisk and a lot of other modern day softswitches there's usually a field called accountcode or similar which can be used to tie a CDR to a customer. The src field is then only

> -----Original Message----- > From: Richard Lyman [mailto:pchammer@dynx.net] > Sent: Wednesday, December 20, 2006 4:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > Douglas Garstang wrote: > >> -----Original Message----- > >> From: David

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> <title></title> </head> <body bgcolor="#ffffff" text="#000000"> Hello everybody,<br> <br> currently I'm implementing redirection

this purpose. I have been reading the voip-info pages and have set up AJAM and can get results from doing http requests to the asterisk server, however, this is in the form of an action, such as login, rather than subscribing to an event. I have been looking here for information http://www.voip-info.org/wiki/view/asterisk+manager+events Does anyone know if subscriptions is possible with AJAM?

Hi everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happens, it appears that the call still goes through as I can see the caller still navigating