search for: pisac

Somebody must have! Please, send me a g723 and/or g729 (for Asterisk) to pisac@hotmail.com (antispam subject: codec) Thanks, thanks, thanks... :-)

...And see if that helps. You need a timing source for it to work, which is why it is disabled by default, but the logging might be a bit chatty in any case. Dan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Pisac Sent: Saturday, January 14, 2006 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 1.2.1 "Silence suppression is disabled" whatthehell? Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f ----- Original Message ----- From: "BJ Weschke" &l...

I'm using bristuffed Asterisk with ISDN/ZAPHFC I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in zapata.conf, but without echocancel I have bad (incoming) echo Through PSTN/FXO sound is ok with or without echocancel. I tried other echo cancellers (in zconfig.h) two times: ECHO_CAN_KB1 (this was default) ECHO_CAN_MARK2 ECHO_CAN_MG2 after any change I compiled (make

I upgraded from 1.0.9 to 1.2.1. In 1.0.9 everything worked perfect. Now, I call in my IVR, and after navigating in menus when I get dialtone for dialing extension, Sound is choppy and I get bunch of messagess: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence

Do bristuffed zaptel (zaphfc) supporting rxgain/txgain in zapata.conf? I'm changing rxgain in zapata.conf, and reloading zaptel, but sound level on ISDN(HFC) is always the same (loud).

...risk source tree. Unless you have previously typed 'make clean' typing 'make' should only rebuild the files that have changed. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Pisac > Sent: Tuesday, January 17, 2006 6:11 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] How to compile and install just > one module? > > make what? > > If I type make in /asterisksource/channels, then all modules > will...

I upgraded from 1.0.9 to 1.2.1 My IVR which worked perfectly on 1.0.9, now hangup with no reason (at least I could not find a cause) When this hangup happen, I can read: == Auto fallthrough, channel 'IAX/user-20' status is 'BUSY' This happening also with ZAP channels I'm really disappointed with 1.2.1, what is benefit from upgrade if I must spend couple days to get my system

Voice-mail messages aren't sent to e-mail address. I have two Asterisk servers, first one is upgraded from 1.0.RC2 to 1.0.9, and second one is from 1.0.7 to 1.0.9. Both Asterisk have EXACTLY same "voicemail.conf" configuration, but second Asterisk don't sending voice mail messages through e-mail! I'm using almost default "voicemail.conf" with just one mailbox

If my extension is 22, and voice mail access number is 909, then with exten => 909,1,voicemailmain(s22) I can access voice mail 22, without number and password prompt. But, I want that every extension can access its voice mail without number and password. So, when I put exent => 909,1,voicemailmain(${calleridnum}) voicemail want only password. I want to eliminate password too, so when I

Do somebody using CISCO AS5350 with Asterisk? Which protocol do you using: H323, MGCP, SIP? This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u) -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm using sipgate.de as my sip provider. When I'm using xlite as client on sipgate.de, everything works fine: I call number, hear ringing (real progress tone form called party, not one generated in xlite) and then talking with called person. But, when I'm using Asterisk as sip client on sipgate.de, I don't hear progress tones: I hear only one (locally generated) ring tone, and

Why Asterisk showing (on SCCP and H323 phones) different CID related to type of Incoming channel: If incoming channel is SIP, on phone is displayed CALLERIDNUM If incoming channel is ZAP, on phone is displayes CALLERIDNAME It vas very frustrating! I lost couple hours of my time to find that my dialplan is not faulty, but asterisk is!

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?

When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise that this is DATA call, but answering anyway (playing IVR messages, etc...) How to stop that? I want that only VOICE calls are answered, and DATA/FAX to be ignored. (I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f, ZapHFC) Log: -- Accepting data call from 'XXXXXXXX' to '3001' on channel 0/2, span 1

...> To UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ------------------------------ > > Message: 4 > Date: Thu, 1 Sep 2005 20:48:10 +0200 > From: "Goran Dj." <pisac@hotpop.com> > Subject: [Asterisk-Users] How to speed-up > INCOMING-RINGING-ENDED > detection on X101P/zapata? > To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <007201c5af25$b500b3a0$0300a...

Driver chan_h323.so ---- If extension is exten => 0119823,1,dial(h323/0119823@10.10.10.1) then dial is OK: Executing Dial("SCCP/goran-00000002", "h323/0119823@10.10.10.1") in new stack ---- But if extension are something like: exten => _011xxxx,1,dial(h323/10.10.10.1/${exten:3}) exten => _011xxxx,1,dial(h323/${exten:3}@10.10.10.1) exten =>

How can I see which codec is in use during conversation. I can see (for example) which codecs are negotiated before SIP connection, but I don't know which is chosen: 12 headers, 12 lines Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 2 Found RTP audio format 101 Peer audio RTP is at port 217.10.79.30:15666 Found description format GSM Found description format iLBC

In sip.conf, Allow=All stopping all sounds! When I comment out this command, everything is OK. I can Allow all codecs one by one, but Allow=All produce same consequences as Disallow=All. I have Asterisk 1.0.0. Is this a bug?

Asterisk 1.0.9: IAX2 registration timeout! --- I have 2 locations with ADSL lines, both with dynamic IP (+ dynamic DNS). On location 1 => Asterisk 1.0.RC2 / Slackware 10 On location 2 => Asterisk 1.0.9 / Slackware 10 They are on private network and connected via IAX2 through NAT(win2000server), and registering to DDNS name of each other. I know that Asterisk is not very smart on handling

I want to speed-up dialing on X101P clone (Ambient modem). I probably must change wcfxo.c, but what line to change? (On usual modems, I can type ATS11=50 to get tone dialing much faster (50ms instead of default 90ms). After that, I can write configuration to nvram (AT&W) to be permanent)