similar to: VoicePulse SIP

Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard

Hi Everybody, Has anybody tried the above (or indeed any other 4XBRI cards) successfully with Asterisk. As far as I can see the above mentioned card is an active ISDN card but supported by it's own I4L driver. This leads to interesting questions particularly regarding echo cancellations (which usually doesn't work on the cheap passive cards with one exception as far as I can see).

Hi! I have some problems with the download of Asterisk-addons. I try to follow instructions that I found in http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20mysql , but nothing to do. This is my shell: [root@obelix root]# cd /usr/src [root@obelix src]# export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot [root@obelix src]# cvs login Logging in to

Hi Everybody, Got a weird problem here I think. Got a setup with an asterisk (current from cvs as of a few hours ago) in a box with an el-cheapo ISDN BRI card connected to the PSTN network and two Snom phones internally (one Snom-100 and one Snom-200). Dialing between the snom phones or dialing out to PSTN from any of the snom phones works perfectly. But when I receive a call FROM the PSTN

Hi Everybody Any significant changes to CVS HEAD over the last couple of days. I've got two asterisk boxes - both on public IP but one is dynamic. The one on dynamic IP registers at the other one - that part is fine. Calls going from the one with dynamic to the static one goes fine. Call the other way results now in: Failed to authenticate user "1101"

Dear Sirs, I've got a weird problem with IAX2 transfers. My setup consist of 3 Asterisk servers. One is located in Europe on a public IP and a local PSTN connection through ISDN. Two are located in South-east Asia - both on public, but dynamic IP. These two each have a bunch of SIP phones attached. All 3 systems are running latest CVS (ok - might be a day or two old) on Linux 2.6.6 with

Greetings, I have been experimenting with Asterisk for a few weeks and finally decided to take the plunge and purchase a few DIDs for inbound calling. Our attempts at IAX/IAX2 connectivity with VoicePulse have been less than successful. We get "Registration Refused" errors from Asterisk whenever we launch the server. The front-line support folks at VoicePulse suggested that we are

I can ping it just fine. I am on gw5.voicepulse.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040615/054c83f4/attachment.htm

Hi Everybody, I've been asked to upgrade an old NT4 server with a new server running Debian/Samba. I've got no problem with migrating the old server, but I do have one "unknown". The company mail run on an Exchange server that is most likely part of the NT4 network. Has anybody tried this setup? I did look in documentation and google and found precious little - which to me

hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am

Hello: I am new to the list. I am trying to set up asterisk with voicepulse. I have a voicepulse username + password, and SIP DID. When I login to voicepulse, I have this under my devices tab: Devices *Login:* Sysxxxxxxx *Password:* xxxxxxxxxx *Context:* VPWS *Connects to:* gw5.voicepulse.com My question is: Do I need a 2.4.x kernel? Currently I am running Debian/stable stock 2.2.x ? Has

Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl

Hey guys, Cal someone help me. I'm register voiceplus DID i try to config fllow example but not work. When i test call to number and debug iax2 in my asterisk not found packet. My iax.conf -------- register => in-xxx:yyy@gw5.voicepulse.com [voicepulse] context = voicepulse-incoming secret=yyy auth=md5 type=friend host=gw5.voicepulse.com ------ extention.conf ---- [voicepulse-incoming]

Hi Chris, I know this is quite an old email, but I was browsing through the archive :) I am currently working on "embedding" asterisk in one of Allwell's STB's. The idea is more or less exactly like yours. The STB will be solid-state and contain OS, Asterisk, Basic configuration and voice files on a flash disk. It will boot up and get a network share via DHCP. This network

Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

I am wondering if there is a way to create a SIP/IAX group of outgoing lines like Zap groups. I am currently using the following method, but would like to use features such as ?g2? that would list all the accounts for a SIP or IAX connection. exten => _1NXXNXXXXXX,1,Dial(SIP/account_name:Password@gw5.voicepulse.com/${EXTEN }) exten =>

Hi there, Since last 2 days iconnect's incoming is not working. Is it the same with everybody? For the past 5 months I've been using this service perfectly in two boxes and suddenly it stopped functioning. I'm able to call out, the version is 0.9.1. Any help is appreciated Thanks, Raj --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB

hello I have a Cisco ATA working with asterisk, but I have east error when it is tried to validate in asterisk. Feb 2 10:04:42 NOTICE[1125342512]: File chan_sip.c, Line 5210 (handle_request): Registration from '<sip:501@216.xxx.xxx.72;user=phone>' failed for '200.xxx.xxx.24' Feb 2 10:04:42 NOTICE[1125342512]: File chan_sip.c, Line 5210 (handle_request): Registration from

I have two questions regarding number portability... 1) If I port a DID over to Voicepulse, can I then move that DID elsewhere somewhere down the road. Or does voicepulse now OWN that DID? 2) Can I take a DID assigned by Voicepulse and transfer it to someone else? If not, why? -jwb