similar to: Set CallerIDNum on a PRI

Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus

Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

Is what I am trying to do in this context possible. That is changing the incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not preceeded by a "1" I want to add a "1". Often calls come in without the preceeding "1" and this plays havoc with my redial if the 3 digit area code matches a local 3 digit extension. All my outside calls are 10 digits or 1+10

Hi all -- I'm having awesome fun with Asterisk & voicepulse connect together. So cool. I'm trying to have the caller id read back to me. Do I need to do something to have this sent across in the sip.conf? Or is there something I need to do somewhere to enable the reading of this data? Thank you! Matt Here is my extensions.conf exten => _XX.,1,Answer() exten

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani

Can someone tell me how to do this...Given the following line: exten => *97,3,VoicemailMain(${CALLERIDNUM}@default) Is it possible to add some logic to manipulate the CALLERIDNUM to send back 801 even if the extension is 601 and 901 even if the extension is 701? I have 2 branch offices where users have both Office and Home SIP phones. I want them to share a VM box. Branch1 = 8XX , Home =

When a user dials 999 he is always asked for the mailbox and has to enter his mailbox number and password. As I understand this shouldn't happen because the CALLERIDNUM is passed over to VoicemailMain. It's annoying to have to enter the number everytime ... The voice mail configuration is read from MySQL. We are using the CVS version from a few days ago. Extract from extensions.conf:

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I

I upgraded from 1.0.9, to 1.2.1. I was using this line exten => s,1,gotoif($[${CALLERIDNUM::3} = 066]?mycity,1:other,1) it selecting calls if callerid begins with some number pattern (from some city) But, it's not working anymore in Asterisk 1.2.1 when I test this with noop(${CALLERIDNUM::3}) I get full callerid, not just first 3 numbers like it use to be on 1.0.9 Why?

Hi Does anyone know how I can check if a callerID is more than 2 digits. I am setting up my phones so that if the callerID is 3 digits the phones ring one way if it is more than 3 digits it rings another i.e. internal calls and external calls. exten => 2222,1,GotoIf($["${CALLERIDNUM}" = "1111"]?5) This will tell it to jump to 5 if callerID if 1111 but how do i

Hi, I'd like to know if and how can I set CallerIDNum for outgoing calls on a PRI line with DDI. Does anyone know if italian Telecom permit this? Thanks -- Domenico Viggiani

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

Greetings all.. We have an inbound/outbound PRI installed and terminated on a T400P ? Digium Quad T1 card. We?re seeing an odd problem when sending $CALLERIDNUM when calls from the PRI are forwarded back out to the PSTN over the PRI. The $CALLERIDNUM is not being sent out along with the call. It?s sending the phone number of the PRI itself, rather than the $CALLERIDNUM information. Yes, we can

Hi, I'm actually using a slightly old version of AAH with Asterisk 1.2.1, because at first install it was perfect for my moderate knowledge of Asterisk. It is working well but I gradually introduced many changes to dialplan during normal use and now I'm feeling like in a straitjacket! Moreover I'd like to have the chance to upgrade Asterisk regularly. I have not the experience to

Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN <--> Asterisk <--> E1 cable <--> Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I

Hi, I'm planning to install a new Asterisk server with a Digium TE410P card. Can I use Red Hat Advanced Server 4 (latest update)? Is this a good choice? Is recompiling Asterisk simple with kernel 2.6? Thanks -- Domenico Viggiani

Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click "Re-register" in the web interface. I set: - Support broken Registrar: On - RTP Encryption: Off Any help? -- Domenico Viggiani

Hi, after a few of upgrades, I noticed these messages in full debug log: Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring