similar to: OT - Experience using Gmail for AsteriskMail ingList

Hi for those who are unhappy with whatever mail reader arrangement they have reading the mailing list, I'd like to share my experience using Gmail, which I have been using for about a week or so now. I find Gmail to be excellent for the mailing list. It doesn't feel like a web mail application at all. The threading works perfectly. Responding to the list keeps the threads intact. It

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

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)

----- 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 -

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

; ; 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 =>

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

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

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

-- Executing unixODBCput("SIP/10-cc1b", "BLAH/blah=bkw") in new stack -- unixodbcput: family=BLAH, key=blah, value=bkw -- Executing unixODBCput("SIP/10-cc1b", "BLAH/blah=bk2") in new stack -- unixodbcput: family=BLAH, key=blah, value=bk2 -- Executing unixODBCget("SIP/10-cc1b", "testingget=BLAH/blah") in new stack -- unixodbcget:

I've had a dedicated box running for ages in my LAN without any kind of problems. Ssh has been installed and useable till tomorrow when a problem pop up. KERNEL: 2.6.5 no server or client settings have been changed. I can ping and nmap the host without any kind of problems. Bellow I'll paste a verbosed ssh try. bkw at tellus ~ $ ssh -vvv neptune OpenSSH_3.8p1, SSH protocols 1.5/2.0,

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

Hi bkw Yep, which is going to be a huge problem since it's only taking a line and not doing any transmittal until after you get a line out, the line of course is being rejected before I can even get there :( Of course I can't even establish connectivity to the telco whilst having it peered to the PBX too due to the D channel issue :( Lee From: "brian" <brian@bkw.org>

Anyone wishing to help build/manage openenum.net please contact me via email brian@bkw.org ... I would like to have someone assist in building and management. Thanks, bkw