similar to: 2 x100p + Static + echo

Relatively new to Asterisk and SIP and am trying to run a proof of concept using Asterisk to make an outbound call through an Audiocodes gateway via SIP using Asterisk version 1.6.1.12. The specific requirements of the gateway in the configuration I am trying to use specify that the Name part of the From header be blank with the outbound number that needs to be dialed in the number field of

Hi, I've been noticing a lot of these messages lately: "NOTICE[10235]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms?" Is something broken? I'm running asterisk-1.4.22.1. They seem to happen in a number of different places where a beep or recording is played, such as when someone leaves voicemail or when an AGI script I have plays a time announcement -- lots

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

On Tue, Feb 28, 2023 at 07:04:33PM +0000, Bobby Eshleman wrote: > Add support for sockmap to vsock. > > We're testing usage of vsock as a way to redirect guest-local UDS > requests to the host and this patch series greatly improves the > performance of such a setup. > > Compared to copying packets via userspace, this improves throughput by > 121% in basic testing.

On Tue, Aug 01, 2023 at 04:30:22AM +0000, Bobby Eshleman wrote: >On Thu, Jul 27, 2023 at 09:48:21AM +0200, Stefano Garzarella wrote: >> On Wed, Jul 26, 2023 at 02:38:08PM -0400, Michael S. Tsirkin wrote: >> > On Wed, Jul 19, 2023 at 12:50:14AM +0000, Bobby Eshleman wrote: >> > > This commit adds a feature bit for virtio vsock to support datagrams. >> > >

Has anyone done much work with server-side push and AJAX/J? (XML or JSON) As best I can tell, the ideal solution to this would be the multipart/x-mixed-replace MIME type. Mozilla''s XMLHttpRequest now has support for this, so this is on people''s minds, but how to make it work on the server end? You have to keep a connection open.....and a process on the server to go with it.

Hi folks! I'm using an X100P (connected to my phone line) and an S100U, and when I calls out from the phone connected to the S100U it is a very bad sound quality, it "pops" and "jitters" a lot. But internal calls from for example a SIP client to the phone on the S100U sounds good. Calls from an SIP client to the outside world using the X100P also works good! My second

From: "hafeez bana" <hafeez_bana+dusers@fastmail.fm> To: asterisk-users@lists.digium.com Date: Wed, 16 Jul 2003 05:17:59 -0800 Subject: Re: [Asterisk-Users] X100P in Australia Reply-To: asterisk-users@lists.digium.com > Thanks for your reply. I tried changing the constant DEFAULT_CIDRINGS to > 2. However I still don't get callerid information. Is this what you were >

I have a simple little asterisk setup: FXS is a PhoneJack PCI, FXO is an X100P. I have a regular old cordless phone plugged into the FXS port. I can dial anything and it's picked up properly by *. I can call FWD and it picks up the DTMF properly. I call out the FXO but nothing I call through there can hear the DTMF clearly. Bell Canada's Call Answer service, for instance, can't

Howdy y'all, I am using an ATA-186, through Vonage, connected to my X100P. It works well - except, hanging up... The hangup doesn't register, most of the time, with Asterisk. Are there any known tweaks out there that I should look into? I've noted some have altered dsp.c, and rebuilt. Just curious to see what y'all think. Below is my zapata.conf file:: [channels]

I'm having a problem getting 3 way calling to work correctly using two outside lines and one extension. The two outside lines are connected to the X100P's and a standard model 2500 phone is connected to the TDM10. When I dial the first outside destination 9xxxxxxx, the call completes correctly. When I flash the hook switch and dial the second location 9yyyyyyy. The call doesn't

Hi, I have two X100P cards in the same system. I can use both of them to initiate and/or receive PSTN calls. I want now to define separate context for each of them, in oder to route inbound calls to different extensions. This is what I have now in zapata.conf file: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes callwaiting=yes echocancel=yes

Hi everybody I am having problem detecting fax with my X100P. I have RedHat 8 as OS and an X100P and a TDM400P. The X100P being plugged into PSTN. I have successfully installed tiff-v3.5.7 and spandsp-0.0.1 and also patched Asterisk wthout problem. Here is my zapata.conf file context=cda signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hello all, I have just installed a Wildcard X100P into an Asterisk box. I connected the line socket to the internal telephone system where I work. The card is identified to asterisk etc, however I am unable to recieve or make calls. When attempting to dial I get: Executing Dial("SIP/1106-ec8b", "Zap/1/644746") in new stack Called 1/644746 Zap/1-1 answered SIP/1106-ec8b Hungup

Hi, I'm currently running asterisk 1.0.9-r1 and zaptel 1.0.9_p1-r1 (the current gentoo ~x86 versions), with the UK CallerID patches from http://www.lusyn.com/asterisk/patches.html applied. The Zap interface itself seems to work fairly well - although it's a little quiet, there is no echo. Unfortunately, there's also no CallerID. My zapata.conf is as follows: [channels]

Dear All, I am a newbie about asterisk. I have 1x X100P card 3x Sip phone I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I got no caller id, even my direct PSTN service operator. So at that moment I was using *1.0.9. than I changed to asterisk@home 1.3(1.0.9). I got same

Hi, Just wondering if anyone else in Australia is using the X100P to connect to the PSTN, and what configs they have for it? I'm finding at present when I make a call I get a fair bit of echo of myself speaking, and also the person on the other end can't hear me very well (perhaps need to up the digial Tx Gain? I don't have it configured at present) Asterisk is running on