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...l. 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 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>

...e originating party - but the SIP softphone is not able 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

...2004-12-23 14:38:30 Resource Line:1-1401 3022241610 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

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)

...de some direction? --> --> --> --> 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>

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

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.