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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

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

----- 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

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 -

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: 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

; ; 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!

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

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, 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

Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably with an rj-21 - but okay if not.. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110614/42cbd90d/attachment.htm>

Hi folks. Please don't flame me but I've been googling around for days, read a tremendous amount, tried everything, and still no go. This is most definitely a typical newbie question. - I sure hope there's somebody(s) out there who'll humble themselves to help me out. I've set up an 'out of the box' basic Asterisk server running on Slackware Linux. - It basically

Hi List! any body use www.simpletelecom.com? I subscribe to www.simpletelecom.com for A-Z termination and paid US$15.00 and US$70.00 via credit card in two days, but my account has US$15.00 only. I checked my credit card from the bank and they said me the payment already paid to merchant. I've lost US$70.00 :( so anyone here has experience with them? are they a SCAM? Thanks! </Madhawa>

Sorry if this not the right place to post this.... BUT... Since May 31st, ALL of these user list messages have been filtered by "spamassassin" running on my Linux box. - Claim to be listed in "Bayes" as spam. - Have no clue why this is happening. Luckily, "spamassassin" sent the messages to the "probably-spam" folder on the Linux box & I was able to

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

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.