similar to: Lucent iMerge

I have a problem where an Asterisk server is sending a premature DRQ... Not sure why.. Here's the setup - Asterisk using inAccess networks H323 replacement channel driver Connecting to a Lucent iMerge... The call connects fine - I get the out of the box greeting - but after exactly one Minute - the call terminates. I have had this problem on multiple different Asterisk configs... I'm

I have built latest Asterisk w/ OpenH323 channel driver. We have a SIP softphone registered to the Asterisk. We can place outbound calls from the SIP phone to the PSTN via OpenH323 connection to our gatekeeper. Everything works okay - DTMF and Audio... But in the reverse - if we call from a cellphone or landline the PSTN number we can get the SIP phone to ring - we answer and can hear the

Take mercy on me - I'm a newbie w/ Asterisks... Here's what I'm trying to do - and please someone let me know if this can be done... We have a commercial VoIP network (we are a communications carrier)... The gatekeeper (Lucent iMerge) supports MGCP/H.323 and allows for calls to be made to the PSTN cloud via GR303 links. I would like to build Asterisks with H323 (or MGCP if need be -

----- Original Message ----- From: "hank" <hanksmith4@earthlink.net> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Tuesday, June 28, 2005 10:52 PM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? >I am using asterisk@home 1.0 > my mp3 is called > mp3 > it has nothing before it

I have a Cisco 7940 when you call in from outside and dial the Cisco phone extension I get this Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[18451]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3)

Hy, I have a network with WIFI communication and VHF/ HF channels. I have integrated asterisk in the network using SIP, ZAP and IAX2 channels for WIFI communications, but I don't Know How I could integrate the VHF/ HF channels. I have heard speaking about app_rpt project, but I don't Know very much about this. Could I integrate VHF/ HF channels with this application? if the answer is

Hi, Just a quick word on this since I was fortunate enough to attend. There were about 18 people, almost all French (if you include the marseillais as French, they may have objections :) Not that I was counting, but there was one female human there. Thanks Mark for your generosity and the good choice in restaurants both this year and last June was it? The souffl? au Grand Marnier was very nice,

Hi all, I'm having trouble with H.323 outbound calls, * connects but there is no sound in both ways. I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729 licenses installed and this is onli one call. I tested my * with another ITSP over SIP and G.729 codec and there was all ok Here is my configs

On 19 Oct 2015, at 19:05, Bruce Hoult via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > I find the git imerge script extremely useful for this kind of situation. > > https://github.com/mhagger/git-imerge > > Logically, it does something similar to rebasing your local branch onto EVERY commit in the upstream branch, in turn, until it finds conflicts. There is

> -----Original Message----- > From: Renato Golin [mailto:renato.golin at linaro.org] > Sent: Monday, July 25, 2016 7:11 AM > To: Daniel Sanders > Cc: Robinson, Paul; llvm-dev at lists.llvm.org > Subject: Re: [llvm-dev] [RFC] One or many git repositories? > > On 25 July 2016 at 14:55, Daniel Sanders <Daniel.Sanders at imgtec.com> > wrote: > > I know of a way

; ; Music on hold class definitions ; [classes] default => /var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => quietmp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters (specifically embedded spaces) ;manual =>

I will second that... I have been doing dedicated IP service for my customers for $130/month in Seattle + loop. (most loops are add about $200-300/month). Anything higher is really a rip-off. John :) -----Urspr?ngliche Nachricht----- Von: Huddleston, Robert [mailto:RHuddleston@cavtel.com] Gesendet: Tuesday, June 14, 2005 12:49 PM An: 'Asterisk Users Mailing List - Non-Commercial

Hi, If it is really true that the voip-info.org website is hosted on a DSL connection without static ip, I have a server in managed.com datacenter that can host it. I still have some ip's free, so tell me if you want to use it. Bandwidth will be on my cost the first terabyte every month. Server has plenty of space left on the HD. I offer this for free, heck, I even offer mail domain with it!

Have you tried the git-imerge add-on? https://www.youtube.com/watch?v=FMZ2_-Ny_zc On Tue, Jun 7, 2016 at 5:17 AM, Robinson, Paul via llvm-dev < llvm-dev at lists.llvm.org> wrote: > > > > -----Original Message----- > > From: llvm-dev [mailto:llvm-dev-bounces at lists.llvm.org] On Behalf Of > David > > A. Greene via llvm-dev > > Sent: Thursday, June 02,

Having fought my PXE config and eventually getting it to work (had to comment out the group { } statments ? I now come across another issue I cannot resolve and would appreciate some help. in my /tftpboot/pxelinux.cfg/default file I have cause to use the APPEND statement to pass options to the kernel being used. Unfortunately this line has now become longer than 255 characters and appears to be

I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 / NCS 1.0 -- is the channel_mgcp compat with NCS -- anything I can do to make it compat?? This is what happens - below *CLI> mgcp reload Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found Use EXIT or QUIT to exit the asterisk console == MGCP Listening on 10.1.22.39:2427 == Using TOS bits 0 mgcp

Hi, all. I feel like we've strayed pretty far from the question originally posed in this thread. One of the pieces of feedback I got before I started this thread was that many people felt that, the last time the question of multiple repos vs. monorepo was discussed, it was interspersed with other topics, making it difficult for some people to weigh in appropriately (or even to be aware that

My boss is insisting we support fax, and I keep telling him that Fax over IP is very unreliable and not recommended and his immediate come-back is "Vonage does it." and it's very hard to figure out how. I don't think Vonage does T.38, the Linksys/Sipura units they're using doesn't support T.38 to my knowledge. That means they have to be using G.711Ulaw to send faxes.

hello I followed the info given and I can't seem to get this to work has any one sucessfully done this? if so can you help me out? I am trying to use a 128 kbps mp3 feed to stream to people while there on hold the info I am using is below. Shoutcast Music On Hold You can have asterisk use a streaming source for on-hold music. Make a directory and put a 0 size file ending in .mp3. I called

Hi: I want to connect a wi-fi phone to my Asterisk box through a wi-fi AP so I can make voip calls. please send me your recomendation about what wi-fi phone I should be looking for. Anybody tried the HOP1502 Wi-Fi IP phone. Its listed price $39. Regards; Chawki ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football