search for: mimmus

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

Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus

Hi all, Did anyone had success with Q.Sig on * 1.2, especially with Alcatel 4400 (which seems to only support Q.Sig) ? I am thinking about interconnecting 15 sites together with asterisk (probably using IAX or SIP). I have a very heterogeneous environement using both PRI and BRI, but my pilot will start with the Alcatel at the central site. Any help/example is most welcome. BR, - Patrick -

Hi, I'm seeing a lot of 'h' as destination numbers in my CDR logs. Some time ago I solved this problem but now I'm not able to remember anymore. Something related to match-all extension? Any help? Thanks Mimmus

...-- Call accepted by 10.97.1.7 (format ulaw) -- Format for call is ulaw -- IAX2/215-33 is ringing -- IAX2/215-33 answered Zap/2-1 Why I have 'Format for call is ulaw'? I'd like to have alaw but keep ulaw to accomodate errors in various configurations (if any, not here!). -- Mimmus

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

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

Hi, I flashed my ATCom AT320 phone (PA1888S based) with IAX firmware instead of SIP but now call transfer doesn't work neither using phone buttons nor using Asterisk features. I heard that it can be a real problem. Any help? Mimmus

Hi, I know it is a FAQ but I'm interested in latest news (if any...) about SIP over TCP support in Asterisk. I found this: https://savannah.nongnu.org/projects/asterisk-tcp/ but I'm not able to understand if project is active and what is its level of development. Thanks Mimmus

...nd? In other words, do all ports on all switches from phones to server need to be configured as 'tagged'? - how can I configure ethernet card on the Red Hat server (Broadcom, tg3 driver) to support tagged traffic and to mark outgoing packets with priority 6? Thanks in advance for any help Mimmus

Can someone explain me attended transfer with Grandstream GXP-2000? Hitting TRNF button, I get: Dial number (BLIND) or Select line (ATTENDED) What's the exact meaning of 'Select line'? Thanks Mimmus

Hi, I have a lot of GXP-2000 phones not registering with Asterisk server. After two days of attempts, it seems that problem is due to the fact that phones and server are not on the sme network. Do you know if this is known issue? -- Domenico Viggiani

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

...tic 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 some time ago). I can post pri debug output in both cases, if needed. Thanks in advance for any help -- Mimmus

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

Hi, is there a way to 'manage' result of a call file (NOANSWER, BUSY, max attempts, etc) put under /var/spool/asterisk/outgoing? Thanks Mimmus

...users). We are encountering some difficulties in finding a 'good' softphone (SIP/IAX). Suggestion/experience? Is there some product available for Windows with modifiable code? Is there some freelance developer potentially interested in creating custom version for us? Thanks in advance -- Mimmus

Hi, I'm asking to myself what's the main problem in using cheap BRI cards (30-60Euro, as these HFC-based) vs. great active cards as Eicon DIVA. Any help? -- Mimmus

Hi, I have a PRI line with DID (from 100 to 499) in Italy. Now I'm seeing all calls from same DID 'main' number. Can I set outgoing CallerIdNum to the right extension? Thanks Mimmus