similar to: Set caller ID for outgoing PRI calls

I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to

Yes, I have the exact same problem. :( -----Original Message----- From: Tomislav Vojvodic [mailto:tomislav@vox-mundi.net] Sent: Thursday, May 11, 2006 5:48 AM To: xytek@hotmail.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted

There is hotel application weary popular in Croatia - Micros-Fidelio. Now I need to connect Asterisk with this application for purpose of billing. Thing is that hotel would like to give customer one bill for every service that he used while he was in hotel. Has anybody connected Asterisk with Micros-Fidelio? As I understand this isn't some local developed application, it's something

I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me

Hi everyone, I'm trying to update time on all Cisco 7905 phones in my company.. is there some way to do it from asterisk? Thanks, Tomislav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/ea0c7e02/attachment.htm

Hello R gurus, I just spent my first weekend wrestling with R, but so far have come up empty handed. I have a dataset that represents file downloads; it has 4 dimensions: date, filename, email, and country. (sample data below) My first goal is to get an idea of the frequency of repeated downloads. Let me explain that. Some people tend to download multiple times, e.g. if the download fails

What am I doing wrong here? Why is this happening? libpri is version 1.0.7-1 (debian package) asterisk is version 1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2 -- Executing Dial("SIP/739-5935", "Zap/g1/0916000739") in new stack -- Called g1/0916000739 -- Channel 0/1, span 1 got hangup Oct 10 13:14:45 WARNING[7544]: app_dial.c:412 wait_for_answer:

Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn't cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. Thank you.

Hi everyone! Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a

Hello, few days ago I returned from FreeBSD where I was using ZFS/GPT disks in raid-1 array back to (Arch)Linux and installed syslinux as bootloader for booting from single MBR disk. Now, after I finished with migration of my /home data, I'd like to deploy both of my 1TB disks using ext4 (btrfs is not stable enough) under LVM2 using raid-1 array & GPT by having all my partitions (/boot,

I'm not aware of any routine that those the job, although I think that it could be relatively easily done by multiplication the manifest variable vector with the estimates for the specific effect. To make an example: v1; v2; v3; v4 are manifest variables that loads on one y latent variablein a data frame called "A" the code for the model should be like: model <-specifymodel( y

Hi All; I do not know if the SEP_Mac_address.cnf.xml of the cisco file is also has effecting on the DID and Caller ID to appear at the destination, because I found the following: <localCfwdEnable>true</localCfwdEnable> <semiAttendedTransfer>true</semiAttendedTransfer> <anonymousCallBlock>2</anonymousCallBlock>

hi, how to set caller ID for internal SIP users when dialing out on telco ISDN p2p (hfc card) line? I need to setup numbers from 0 - > 9 (10 sip users).internal caller id is working correctly .. from 0 to 9 ,but when I dial on isdn telco line -> gsm show only our prefix number ( xxxxxxY ) , only 6 (x) numbers of 7 (Y). here is extension 1 dialing on isdn line but on gsm is only 6

Hi, I got a broadvoice "business" account under the byod(bring your own device) program. I have applied the patches and created new asterisk debian packages. I have the account working on inbound and outbound. The problem area is outbound caller ID. I have 3 other accounts with IAX2 providers and have no problem setting the caller ID on outbound calls. I called them and they

Hi, is anybody out there who can set the outgoing caller id on ISDN (CAPI or misdn) channels? I've tryed everything what I found in forums, os voip-info.com but no luck. I use a fritz card with CAPI in my first installation (1 BRI), and a hfc 4 port bri card with misdn on other. The first installation have p-t-mp configuration, the second one is p-t-p. Both configuration is EuroISDN in

I'm trying to set caller ids on outgoing calls. I have a quad BRI B410P card connected to my telephony provider. I know the list of DID numbers the provider assigned to my company. If I don't set the caller id then the callee always sees the same "top-level" number. If I set the caller id to a specific DID number we own, the callee keeps seeing the "top-level" number,

Excuse me if I recently posted on this, but I cannot find it, in my, or the list archives. Is it possible, when transferring a call, to set the user ID to that of the "outgoing number" instead of the incoming number? I believe the answer is (was) yes. New twist, does it matter what the destination "media" is? Meaning, the call would be coming in on a T1, going out on a T1,

Hi, I'm using Asterisk and a TE110P E1 PRI in Chile. When I call to a disconnected number or any not operational number, the telco sends the Hangupcause disconnection code and an audio message notifying the disconnection cause to the user. Asterisk does not allow the user to hear the audio message form the telco, instead it cuts the call. Any other legacies PRI PBX I've tested allow

Hi all I have 2 snom 100's and an ix66 (sip aware firewall) set up with asterisk. I needed to register a number of lines so what I've done is make asterisk register all the lines i need (attaching them to an extention eg 1000) and then register each phone with asterisk. so for example in sip.conf: register => andy@sip.mydomain1.org/1000 register => andy@sip.mydomain2.org/1000

Hi Just moved offices in the UK and moved our Asterisk box from old one to new one. Using idefisk softphones, Junghanns quadbri card for ISDN 2e interfaces. At both offices we had one standard number and a DDI range, routed with Asterisk. We'd set up the configuration so each idefisk set its own caller ID which then got sent by the extensions.conf script. Worked fine at old place but in