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...e willing to help lend a hand? It is capable of H323 / MGCP. Even if I could make the Asterisk register with the iMerge I would be happy enough to now that I might be able to make this system work. Please help... Thanks Robert A. Huddleston, KF4BYY Cavalier Telephone LLC. 804.422.4401 rhuddleston@cavtel.com <mailto:rhuddleston@cavtel.com>

...DTMF or audio back to the PSTN... I'm not sure if this is an issue w/ converting the signal in asterisk i.e. SIP to H323 -- or if a problem in the codec or what? The codec is G711uLaw.. Help - thanks Robert A. Huddleston, KF4BYY Cavalier Telephone LLC. 804.422.4401 <mailto:rhuddleston@cavtel.com> rhuddleston@cavtel.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040910/23475334/attachment.htm

...8.117 Released resource DSPchannel.1.8.1.32 224 2004-12-23 13:47:46 2004-12-23 14:38:30 Activity Line:1-1401 3022241610 67.62.108.117 Deleting Media Path Robert A. Huddleston, KF4BYY Cavalier Telephone LLC. (Desk) 804.422.4401 (Cell) 804.400.3686 rhuddleston@cavtel.com

...gt; --> --> PSTN Lucent Gatekeeper T1 (or broadband) Asterisk Softphone/endpoint <-- <-- <-- <-- Thanks Robert A. Huddleston, KF4BYY Cavalier Telephone LLC. 804.422.4401 rhuddleston@cavtel.com <mailto:rhuddleston@cavtel.com>

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)

...me 1.0 > my mp3 is called > mp3 > it has nothing before it > it is 0 bytes > does my mp3 of 0 bytes need to have a .mp3 or does it need to be called > anything? > thanks > hank > > ----- Original Message ----- > From: "Huddleston, Robert" <RHuddleston@cavtel.com> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Sent: Tuesday, June 28, 2005 11:52 AM > Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems? > > >> Worked for me with a differ...

..._files] ;native => /var/lib/asterisk/moh-native ;native-random => /var/lib/asterisk/moh-native,r >---------------------------------------------------------------------- > >Message: 1 >Date: Wed, 25 May 2005 08:57:28 -0400 >From: "Huddleston, Robert" <RHuddleston@cavtel.com> >Subject: RE: [Asterisk-Users] MoH: mpg123 problems >To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > <asterisk-users@lists.digium.com> >Message-ID: > <6FBCD49C09647B4584B0CD4166C0D5BF3E5202@RICHEXCHVN01.cavalier.com> >Content-Type: text/...

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!

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 Discussion' Betreff: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Anyone paying over $450 for a T1 is being ripped off... If you are in VA,MD,DC,PA,DE,NJ you can get an integra...

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

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

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

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.

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Any got experience w/ PWLIB - sorry I know it's somewhat off topic... I do not have a bison.simple file located on Fedora RC2... But when make'ing PWLIB I get ../common/getdate.y:106:1: warning: "YYPURE" redefined ../common/getdate.tab.c:43:1: warning: this is the location of the previous definition Is it safe to continue or is this a bad build?

All - NEED HELP BADLY.. Using Asterisk with H323 Channels.. Everything is installed but we cannot register w/ Lucent gatekeeper. We ran ethereal and found that it was making GRQ (Gatekeeper discovery requests).. We had provided the name of the Gatekeeper (it's IP) and cannot determine why it's trying to do a GRQ. We want it to go straight into RRQ or ARQ and skip the GRQ.. Netmeeting does

I posted previously that my Asterisk is failing to register with parent H323 gatekeeper. It appears that it is issuing a GRQ <GatekeeperDiscoveryRequest> even though I have the file <h323.conf> saying exact IP address of gatekeeper... No discovery. I'm trying to just bypass - remove the GRQ and have it Register <RRQ> Does anyone know if asterisk-oh323 will work better or