similar to: Asterisks

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've read the wiki and other resources on how to connect Vonage / Voicepulse and all these other services to Asterisk... We are attempting a connection to a Lucent iMerge. Lucent has told us that it won't work - but we feel confident that it will. Has anyone worked with the Lucent iMerge - or would be willing to help lend a hand? It is capable of H323 / MGCP. Even if I could make the

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

; ; 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

----- 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)

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,

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 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

I'm a little confused on whether the GR303 support in * will accept calls from a Siemens central office that has GR303. Anyone know for sure?

Anyone have any experience using gr303? May have a need to interface * to a Siemens Class-5 CO for pstn trunking (inbound and outbound). Rich

iH where can i find documentation on Asterisk's support for GR303??? thanks - hcir

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

> -----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

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

I'm trying to connect an asterisk server via oh323 to a Lucent iMerge. I patched the code due so that Lucent can handle asterisk's ver4 h323 http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration I can now successfully dial in to our company over multiple lines. The issue is when I dial out The first outgoing call connects to an outside user A The second call drops the first

Is it my imagination or did I just drop off the list for several days somehow... I didn't get any posts since Friday... -------------- next part -------------- A non-text attachment was scrubbed... Name: rhuddleston.vcf Type: application/octet-stream Size: 575 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/bbb0a930/rhuddleston.obj

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.

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,