similar to: Setting caller ID on no Caller ID

I have looked into this callerid problem now for a few hours. 1) Caller id on a sipura-2000 now shows: cidname 2000 Where cidname is the new outputted formate from the cid_rewrite agi script and 2000 is the exten number. In looking at the Dial() application, option "o" 'o' -- Original (inbound) Caller*ID should be placed on the outbound leg of the call

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

Greetings, 2 nights ago I upgraded one of my remote servers to the latest CVS Stable, Asterisk CVS-v1-0-02/07/05-16:13:48, and my inbound and outbound caller ID stopped working. At first I thought it was my carrier, so I've been yelling at them for the past day or so. I then upgraded my core network (about 8 asterisk servers) to the latest stable as of 3am last night, Asterisk

Before I "reinvent-the-wheel", does any one know of PHP based interface to the CDR table? If not, I will get started writing one. Thanks, Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030624/d4cb6911/attachment.htm

Is there a bug in Firefly (3rdparty) wherein it does not show caller ID? I am using SetCIDNum(12345) before I dial my firefly (IAX2) phone... no caller ID. CallerID is passed properly to other clients. -A.

I have an issue where someone will park a call, and then it will ring back to them, but because the caller-id looks like a regular inbound call, they don't know how to answer the call (these are the receptionists). I've tried to make an extention that I can transfer to that will set the caller-id, and that works, but I'm having issues. For instance.... if I do a blind transfer to an

Hello all, I would like to set a different caller-id for the second leg of a call when doing an originate. For example: Action: Originate Channel: sip/1234 Context: mycontext Exten: 1 Priority: 1 Callerid: "123 <123>" Async: true This sets the caller-id correctly when dialing sip/1234, but I would like to set the caller-id for the second leg of the call (the one that goes to 1 at

Just to bug you all (feel free to rant at me), a client wants to set his caller*ID number for outbound calls though us to PSTN. the client is using SIP to us, he can set the caller*ID name fine. if he sets his caller*ID number to anything other than his account number (8440101), the call is dropped into the default context (and then hung up by our dial plan). To get around this i

Greetings folks, I'm currently dealing with a company to let me set Caller-ID-Name on outbound calls. So far pretty happy with their services. The basic service works like this: * CLEC sets Point Code to point to this company * CLEC has to sign LOA saying they give me permission to set the Caller-ID-Name through this company. * I go into web interface and set name. However, the CLEC is

Hi Most of our customers have generic names like "Hospital", so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as "Reading Hospital" so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you.

> > To get around this i updated CVS HEAD and changed the sip entity from > > type=user to type=peer (yes peer!) (type=friend works too but im making > > a point) the client now must register to set his outbound caller*ID > Number. > > Yes, that is normal. SIP has difficulty separating the remote party > identification from the authentication identification

I am trying to make a data channel using ISDN and i need to set the caller id num field. Can any body tell me how i can set the caller id field since i notice in chan_dahdi.conf callerid field doesn't work with channel originate. Thanks, Pawel -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! I would like to get connected with a VoIP provider for home. At some point, I'm sure I will be connecting to it via an Asterisk box, but for now, I will be using whatever hardware they provide. What recomendations do you in the Asterisk community have for a reliable VoIP service that will hopefully interoperate with Asterisk? A company that is actually willing to work with an

On Thu, Jun 18, 2015 at 1:26 PM, Matt Riddell <lists at venturevoip.com> wrote: > Did you buy the number from your carrier? Maybe it?s set on their side > for the trunk. > That's what I think too, but they are denying this. I think what's happening is they have a customer service guy interpreting logs (probably incorrectly). When I had a Century Link POTS line, I had a

I cannot find any means of passing my own Callerid using Voxee. It always comes across as NO ID, or nothing, or unknown. I could not find anything on their website about setting your own caller id in the system either. (their web account pages). Is anyone here using their own Callerid information through Voxee? thanks

Hello list, I'm observing what I believe to be inconsistent behaviour regarding "Newstate" AMI events for the "Ringing" state. As such I come to you asking for experience or advice: am I wrong or should I file a bug ? I present you a short introduction which I feel is relevant; however, if you want to go straight to my technical question, please scroll

Admittedly this is probably pretty stupid of me, but there are just some things I can't understand by reading documentation. Any suggestions or recommendations about how to handle my problem are greatly appreciated. I'm trying to achieve the same functionality as my Nortel PBX, without rewriting much 'C' code. In my dialplan I'd like to compare two variables as a means of

Hello list, I'm sending calls to a queue in the attended way, that is, *1.* the original call is put on hold, *2.* a second line is open to call the queue, *3.*after an agent is connected the original call is transfered to its final destination. 1. Zap/1-1 <--> SIP/agentA-tag1 2. SIP/agentA-tag2 <--> SIP/agentB-tag 3.

> On 18Jun, 2015, at 13:44, Greg Woods <greg at gregandeva.net> wrote: > > I am certain that the old number that is showing up as my caller ID is not present in any of my config files (that includes sip.conf and iax.conf, everything in the /etc/asterisk directory has been checked). Did you buy the number from your carrier? Maybe it?s set on their side for the trunk. -- Cheers,

> At any rate, if I can figure out the right way to set the caller ID explicitly, and assuming they honor it if I do, then none of this will matter. Ok, so just do exten => s,n,Set(CALLERID(all)=?Greg? <5551234>) https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID -- Cheers, Matt Riddell _______________________________________________ http://www.venturevoip.com/news.php