similar to: PASS CORRECT CALLER ID INFO TO OFFSITE TRANSFER

I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling

I doubt it's possible but I'll ask just in case there's a "legal" way to do that. I have an asterisk server setup at work. When someone call from a PSTN line and enter an extension it rings for a few seconds on the SIP phones of that extension and then if there's no answer it transfer the call to the cellphone of the person having this extension using hook

I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller

I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system

Hello, Avaya has a nice feature that allows you to a) ring both a cellphone and a desktop phone at the same time b) transfer calls (and access other PBX features) from the cellphone that recieved the call, as long as the call is bridged through the PBX c) while talking on the cellphone, pick up the handset on your desktop phone and the call is automatically moved there, hanging up the cellphone

Greetings list, Have there been any further developments recently regarding presenting the original caller's caller ID to SIP devices after an attended transfer? I've googled around on the topic, but most of the threads I've found (some from this very list) are all dated back in mid-2006 and I wondered if there have been developments on the issue. To recap, the desired behaviour

Hello, I'm wondering why asterisk is not transferring the callerid to the sip device. Scenario as follows: sangoma <---> zaptel <---> asterisk <---> sip <---> SIP-Device zaptel is reporting the callerid, but in the sip packages the sip-address shows unknown as user part, as this sip debug package shows: Executing Dial("Zap/62-1",

Dear rsync community, As Time Machine doesn't support offsite backups, I started developing an rsync helper script for synchronizations between Macs and network attached storage (NAS) devices. After running and refining my script for some time now, it works very well for my purposes. Lately, I made my script, that I called Space Machine, available on my blog: http://goo.gl/KhgF4 With Space

We have contracted with an outside call center to provide sales for a certain product. We want to be able to transfer people over to those dedicated sales agents using an attended transfer (so we can prepare them with as much information as we have), to a regular extension. So far, so good. All of this is working just great. We want the caller's information presented as the CallerID so

I have a Motorola Razr successfully connected to asterisk using a bluetooth dongle and chan_bluetooth. Here's some issues I've run across: - You have to ignore the instructions in bluetooth.conf, saying to run "sdptool search --bdaddr xx:xx:xx:xx:xx:xx 0x111F" to determine the correct channel to use for your phone. My phone reported Channel 7, but will not work with anything

As usual, this is most likely a easy question, but here it goes any way: How can I change the caller id on a transferred call so the called party knows the call has been transferred from a colleague and it's not coming directly from our outside lines? The story goes like this: 1) Client calls. All phones ring. 2) Someone picks up the phone. 3) The phone gets transferred to someone. 4) The

All, Having trouble getting answer from Verizon. I believe Asterisk will let me specify a name and number that is sent to the PSTN (Verizon) of outgoing calls. For instance, if I have a client, First Bank, and their toll free number is 888-555-1234, I could send that name and number. Verizon is telling me that they will forward the number I send them, but the name will be my company's

I've seen caller-id come through from carriers as: NPA-NXX-xxxx, 1-NPA-NXX-xxxx, and +1-NPA-NXX-xxxx My question is: what is the correct way to send Caller-ID by set standards? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100809/1866d18a/attachment.htm

Assuming that you are in the US (don't know about elsewhere), the Calling Party Name is obtained by a database lookup performed by the PSTN switch terminating the call. Calling Party Name is the name assigned to your line or location by the phone company. You can't "send" it. You can only receive it, if the phone company has it to transmit in the FACILITY message. Display Name

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.

I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few

I have a single DID number from sellvoip.net. This works well and has been mostly reliable (HEH). I am wondering if caller id info is likely be provided for calls to my DID #? If so, should they show up in the CLI(they don't)? Or how do I go about looking at them (the caller ID infos that is)? Thanks, Marty

Hi Asterisk Users, I was wondering why a call that is received from Asterisk shows a caller ID 'Unknown' . So here is the scenario, 'A' calls 'Asterisk'. 'A' joins a conference on 'Asterisk'. 'Asterisk' calls 'B'. 'B' gets joined to the same conference also. 'B' somehow receives the caller ID 'Unknown' and not the

Is it wise to use an outage to promote your business, not on the user's list and not multiple times? Put it in your signature or something ;-) Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Shane Breen > Sent: Wednesday,

I've used rsync (but probably not for the size you're referring to), it works and has enough features to meet most needs. I have had a single situation where corruption occurred during transfer (a few times, have no idea why), might want to independently confirm the integrity of the transfer. ________________________________ From: CentOS <centos-bounces at centos.org> on behalf of