similar to: CallerID Name not available.

Hi, I'm trying to compile an llvm program which makes use of exception handling. While compiling the code with llc i get the following assertion failure llc: FunctionLoweringInfo.cpp:163: void llvm::FunctionLoweringInfo::clear(): Assertion `CatchInfoFound.size() == CatchInfoLost.size() && "Not all catch info was assigned to a landing pad!"' failed. 0 llc

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the

Hi Shivram, > I'm trying to compile an llvm program which makes use of exception > handling. While compiling the code with llc i get the following > assertion failure an unfortunate weakness of the code generators is that they require calls to eh.selector to be in a landing pad (landing pad = where an invoke unwinds to). They can also handle some simple cases when the selector call

The host I am working with has two accounts from the same DID provider. Incoming calls work correctly and dial the appropriate extensions. This also allows incoming calls to be billed appropriately to the individual DID accounts. Outgoing calls from either extension default to the first DID, i.e. calls from either extension have the same callerID. How can an extension specify separate outgoing

I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. Here is the extension.conf [default] ; ; By default we include the demo. In a production system,

Hi guys, I need your help ... I have a couple of DIDs that reach my Asterisk box .... But I'd like to set my DIDs automatically via Macro or other routine based on the number called by my agent ... Ex: My agent called 954-111-1111 ... So I'd like to set the callerid as 954-222-2222 (That is my DID) Thanks in advance, Marcelo

Hi I have an ISDN connection with 100 DIDs assigned to it... What I'm trying to achieve is set the proper outgoing callerID while showing the local caller's extension in the CDR. There is a behaviour that I just can't explain. the callerid field in sip.conf is set as : callerid="Jean-Yves/E" <300> the callerid in iax.conf is set a: callerid="Jean-Yves/E"

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Adams, Gavin Is there any additional information I could provide to start tracking this down? I was thinking about looking into the various applications source to see how they access the data elements for callerid. I know where the values are pulled

Hi, I have two asterisk servers with the same version of 1.4.29.1. The first server named it as MYE1. MYE1 is an incoming server that can accept incoming calls from PSTN(ZAP E1). The second server is a pbx functions server and named it as MYPBX(SIP). The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default

This depends on what you mean by ?not involving the service provider.? If you are literally forwarding calls that come in on the PRI back out on the PRI, the most efficient way is with Two B-Channel Transfer (TBCT). Check it out in the wiki. You need to make sure your carrier supports the feature. When you want to do a ?transfer,? you have an incoming call alerting or answered, you

I've searched around and found a few similar situations where the phone will call out when using a Asterisk server but not receive inbound calls. My issue is a little stranger. If I call out from the phone then the phone will receive the next inbound call. The phone will not receive another inbound call until a call out again from it first. Any ideas? I am using SIP and am using the latest

My, how embarrassing. I of course meant that as a personal message to Don. But if anyone else knows the answer, I'm interested! lol Cheers, j On 03/18/2015 10:02 AM, Jeff LaCoursiere wrote: > > Hey Don, > > How are you? I may be heading your way in the next month or so. Have > to meet with a guy in Eden Prairie, and stop off at my > brother/sisterm-in-law's as

Hey all, We've upgraded our PRI trunk to support what BellSouth calls "enhanced caller id name delivery". The weird part is, I'm only capable of seeing the name portion on incoming calls within voicemail2's email delivery. For example, on an incoming call, asterisk is reporting this: Context from extensions.conf (BS delivers 7-digit DIDs): exten => 9133727,1,Answer

I am pretty sure that there are no IAX providers that offer CallerID name but wanted to double check with the list in case something has changed recently. Is anyone aware of an IAX provider that offers incoming CallerID name? Is there a technical limitation within IAX which is preventing IAX providers from offering CallerID Name? Why is no one offering this? -------------- next part

Forgot to reply all. Duncan suggested that I put the selector call before the store but that didn't help. I get the same assertion failure. On Thu, Jun 10, 2010 at 2:50 PM, Shivram Khandeparker <shivramk at gmail.com>wrote: > HI Duncan, > > On Thu, Jun 10, 2010 at 1:58 PM, Duncan Sands <baldrick at free.fr> wrote: > >> an unfortunate weakness of the code

Does anybody have callerid name coming in on a Cisco PRI via a Cisco gateway via SIP to *? I've seen a few people ask and a few people that say it should work, but I've never seen an actual working config. I do a debug on our Cisco gateway and I can see the callerid name, however none of the features that should send it via SIP seem to work. Cisco docs say to use the following:

Hi, I have small confusion in my configuration which is I had some DID's like 044578900-04457999. I was configured dial plan below mention. exten => _0XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)}) exten => _0XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) exten => _0XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident}) exten =>

Hey Don, How are you? I may be heading your way in the next month or so. Have to meet with a guy in Eden Prairie, and stop off at my brother/sisterm-in-law's as well. Got a question for you - with TBCT, who pays for the call once it is transferred? Still me as the owner of the trunk? Lets say I take a call that was dialled locally (caller believes this is "free"), and I do a

Hi, I'm putting together a fax solution for my company that utilizes T.38. I wanted to throw out my plan and get some feedback if anyone is doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by