similar to: Sending '#' with Dial

Hi All, After searching and didnt found it, im just sending my situation here, maybe someone knows where i should look. Im using Asterisk 1.6.1.10 Internally the user with a sip phone dials a number for instance 0623456789 It goes fine to the specific dial rule: which is: exten => _0[6].,2,Dial(SIP/0${EXTEN:1}#@xs4all-out,60,tTwWkK) This works fine without a charm, but the situation is that

Does anyone have some method, or AGI scripts that will automatically call a list of numbers from a database and play a pre-recorded message? For example, you have a database of FirstName LastName PhoneNumber Jon -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

Is it possible to change error messages for Validation helpers? I am writing an app against a existing database (so no control over column names), but when there is validation error (e.g. with validate_presence_of) I would like to customize the field name. For example for telephone whose field name is PhoneNumber I would like to chnage it to "Telephone Number cannot be empty" rather

Hi, I?m looking for settings to configure net2phone carrier in my asterisk. I found this configurations, but it?s not work. I don?t known if this configuration is for voice line or voice access account. Anybody can help me, with other configuration? Thanks. ---- *sip.conf* [general] useragent = X-Lite release 1103m register => PHONENUMBER:PASSWORD@sip.net2phone.com [net2phone] type = peer

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming

Hi all, I've been trying to use BroadVoice as a SIP service provider. They don't officially support * but are helpful when it comes to answering questions for setup parameters. They claim they have no firewalls or access lists that need to be set up so I can get access to their servers. However, something's still not quite right when I use the parameters. It looks like our Asterisk

I am trying to place an analog outbound call from a Sipura SPA-841 through a * server with a TDM400P and 4 FXO's. When I call in from an analog line everything works fine, I can talk over the SIP phone. When I call out, * says: == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on 'SIP/sipphone-d29d' -- Executing Dial("SIP/sipphone-9eb0",

I have an asterisk 1.8.x box that intermittently returns a 401. Calls come through the same peer all the time, from the same carrier. However intermittently the asterisk box returns a 401. Below is the output of a failed call (1st) and a successful call (2nd). I can't see any difference until we get to these lines. Bad call: --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT)

Hello! My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on Fedora Core 4. I am making automatic call-out campaign with this setup on 4 PRI. The scripts for this: ==================================================================== caller php script write this to outgoung folder: fwrite($outfile,"Channel: Zap/g1/$phonenumber\n"); fwrite($outfile,"MaxRetries:

Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=<phonenumber> authuser=<phonenumber> secret=<registration password> Dan

Hello! I have an Asterisk@home instalation with 7 users working OK, and I'ld like to implement either a -- Web dial feature, where the user would fill one form field with a phone number and a connection would be created between his extention and the entered number. OR -- Dial using an URI (callto:xxxxx link in a web page), having AstTapi installed and configured in all workstations.

I''m trying to setup some mildly complex associations for a project we''re working on and can''t seem to find much documentation on n-way has_many :through associations. I have the following models: Person, PhysicalAddress, EmailAddress, PhoneNumber. Each person can have multiple PhysicalAddresses, EmailAddresses, and PhoneNumbers, and multiple people can share the same

Hi all, I am struggling with the following and I can't get it work: In the Netherlands where I live it is possible to use special codes (aka vertical service codes) to set special 'behaviour' of phonecalls. So e.g. when I want to dial out with a normal phone and I dial *31*<phonenumber to dial> then it will turn off my numberindication (CID) at the called party. They seem to

Hi all, Still having the one way sound problem. Any suggestions how to hunt the problem down ? Regards, Thorsten --------------------------------------------------------------- Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26

Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk <208>) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco

I am trying to use the simple CID name management script on the wiki. http://www.voip-info.org/wiki-Asterisk+tips+managing+CID+names I can not see what is wrong. The values never get entered in the database. Here are the files: I have asterisk running as the user asterisk also. ---cid-store.php---- <HTML> <HEAD> <TITLE>Storing Asterisk CID data</TITLE> </HEAD>

Hi List, Additional with the latest tries from the below I get a nice random seg fault when I hangup on PSTN. (With obviously no sound on x-lite, still!) asterisk -vvvvgc results after hanging up the pstn line in: -- Executing Hangup("SIP/1087997-d79f", "") in new stack == Spawn extension (sip-phone-out, h, 2) exited non-zero on 'SIP/phonenumber-d79f' Segmentation

Hi everyone, I'd like to use Asterisk to build a phonetree (www.phonetree.com) type of application, like this: 1. Read a text-based name/phonenumber file. 2. Call every number and play a recorded message. 3. If a beep is detected, replay the message from scratch (to leave messages on an answering machine). 4. Write results to a log file. Does anything like this exist already? Can this be

On Wed, Sep 19, 2007 at 05:24:12PM +1000, Emil Mikulic wrote: > http://goanna.cs.rmit.edu.au/~emil/llvm2.1-check-debug.txt Here's an ARM test that cores: $ llvm-as < /home/emil/ll/llvm-2.1/test/CodeGen/ARM/2007-01-19-InfiniteLoop.ll | llc -march=arm -mattr=+v6,+vfp2 Segmentation fault (core dumped) $ gdb `which llc` llc.core [...] (gdb) where #0 0x0853d606 in

On 2019/05/23, Ben Skeggs wrote: > On Thu, 23 May 2019 at 01:03, Emil Velikov <emil.l.velikov at gmail.com> wrote: > > > > From: Emil Velikov <emil.velikov at collabora.com> > > > > Cc: Ben Skeggs <bskeggs at redhat.com> > > Cc: nouveau at lists.freedesktop.org > > Signed-off-by: Emil Velikov <emil.velikov at collabora.com> >