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...- or would be 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>

...to transmit 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

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

...67.62.108.117 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

...--> --> --> 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>

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

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)

...asterisk@home 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...

...no -s [moh_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...

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!

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>

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...

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

I just set up a WRT54GS and now I can't dial out or in. sip show registry shows: CODE: SELECT ALL Host dnsmgr Username Refresh State Reg.Time draytel.org:5060 N xxxxx 120 Request Sent I seemed to recall that running in cli always showed a response back, but there's nothing now. Using

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

Hey; I figured I would ask here as I seem to get better results. I am using the voip.ms <http://voip.ms/> VoIP service. I have no problem configuring my Asterisk server 1.8x to dial out with my Softphone. HOWEVER, for some reason, I cannot get inbound. All that I hear is a busy signal. I know this is not much for you folks to go on, but what would be a good place to start

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.