similar to: Weir long distance behaviour...

Hi all, Is it possible to use 2 incoming fxo lines (one is for my company the other for the family) with Asterisk@home? Best regards, Francois Random Thought: --------------- Errors like straws upon the surface flow: Who would search for pearls must dive below. - John Dryden, 1631 - 1700

Does any one knows of an Windows based SIP video phone???... Thanks... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, January 11, 2005 9:27 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 6, Issue 142 Send Asterisk-Users mailing

I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason. I have tried messing with the busydetect and callprogress setting them to yes and no same and still random hangups. Is there another setting I should be looking at? My zap config looks like. context = inbound-work include => extensions signalling

Hello, I have just updated my svn copy of the llvm/clang repositories after quite a long time of inactivity, and found it not compiling on Windows with Visual Studio 2008. The incriminated file is: llvm/lib/MC/MCModule.cpp Where several calls to "std::lower_bound" are made, like: atom_iterator I = std::lower_bound(atom_begin(), atom_end(),

Ive just installed a TDM4000P with 4 fxos. The zaptel config is fine, zttest comes back with configured. If i call a line when zttest it shows on the display,and then goes when the line drops. In * when a call comes in, it follows my dialplan and answers the call according to the log, but IT DOESN'T actually pick up the call, i.e. it continues ringing. I'm using KS signalling, and

Hi List, I've got * randomly hanging up on inbound or outbound calls on zap channels. I use a Digitnetworks X100P clone card. Any idea of what might be happening? Cheers, Jean-Michel.

Hi Guys, I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: [Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy, avgtone: 75, avgsilence 135 [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [h at macro-dialout-trunk:1] Macro("SIP/2111-b6a400b0", "hangupcall|") in new stack [Jul

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

All of a sudden I am getting the following warning "Ring/off-hook in strange state 6 on channel1" from chan_zap.c and I cannot answer calls. I can place calls out without a problem though. Any ideas what can be the problem. I have checked zapata.conf and zaptel.conf and they both seem fine. Thanks in advance. Dan -------------- next part -------------- An HTML attachment was

Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: [channels] language=sp signalling=fxs_ks usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ;sendcalleridafter=1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes

Hi all, I would like to connect in sip mode an Eyebeam client to a messenger via Asterisk. I want to use video. Nat is not an issue as vpn connections will be used. Is this a difficult tasks, can someone give me some pointers to get started... Have a good week-end, Francois Random Thought: --------------- Wanna buy a duck?

I'm attempting to use callprogress in my system, and I'm having trouble. Callprogress always can tell if the line is busy or ringing, but when the line is answered, the call does not get bridged. Messages showing that "line is ringing" stop in the console and if the called party hangs up, asterisk reports the line is busy. Are there any settings that I could use to help with

On Thu, Jul 4, 2013 at 12:48 AM, Benoit Perrot <benoit.noe.perrot at gmail.com>wrote: > Hello, > Hi Benoit, > I have just updated my svn copy of the llvm/clang repositories after quite > a long time of inactivity, and found it not compiling on Windows with > Visual Studio 2008. > > The incriminated file is: > > llvm/lib/MC/MCModule.cpp > > Where several

Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress

I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI->T400P->Asterisk->T400P->Adtran 750(L36 Firmware)->RAS Server. I have 4 Zap channels signalled FXO_KS to the 750

Hi List, a number of our customers are reporting dropped calls. here is the config. 1 T100P T1 Card 1 Asterisk (Mid Nov build) T1 is signalled as a PRI(National) The card will only sync up if we clock, if we line side clock the card goes into yellow alarm and won't sync up. the only errors we see are framing slips. Around 2500 slips over a 18 hour period. (this was reported from

So I think I am able to reproduce the dropped call scenario. Here is what I do to get a dropped call: 1. Call 1-800-tmobile 2. Go true their IVR and get connected to the customer service IVR 3. Enter my number and SSN 4. press 0 5. Then the audio please hold starts. After about 2-4 seconds the call gets dropped. (fast busy tone) The time on my phone will stop running (call time) and I will get

I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the

I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include