similar to: re: slightly OT

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 all, i have a strange problem....i am running SER in front of asterisk, and am testing softphones. x-lite works fine...i can dial, hang up, DTMF, all good. Firefly looks really cool and i'm very impressed with the IM-like interface and the skinning ability, but something strange is happening...when i call from the firefly and run something on the server and press hangup on the client,

hello all, i need some help and after trying the wiki i'm even more confused than i was. i'm trying to set up call forwarding and running into problems... i want the most basic call forwarding imaginable. 1. caller dials extension (say, 154) 2. dialplan is updated to forward caller's extension (based on CALLERIDNUM) to voicemail, instead of ringing his endpoint. 3. caller is

We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at sales@amarfone.com. Ehsanul Karim

Hi, This issue might be a slightly offtopic, but someone might have experience with it. Thanks for reading this post anyway. I have the following setup: Network 10.227.7.X is connected over a wlan (172.1.1.1 <-> 172.1.1.2) to network 128.1.1.X. This setup works, I have cross-subnet browsing going and I am able to login. When I enable IPSEC (raccoon (linux <-> freebsd)) I am still

Ronald, Grandstream products have a one year warrantee. If you don't have any luck with Pulver, contact us and we can probably get your phones exchanged. Please don't assume that your experience with Grandstream is typical. We sell a lot of these phones and the overwhelming majority of the purchasers are very happy with their units. The quality has improved tremendously over the last

Hi all, i'm trying to integrate cepstral and asterisk, and i have a problem i'd appreciate any help with (i know it's a bit tangential, but i figure this is the place with the most knowledge of app_swift and asterisk). I've installed swift from cepstral.com with alison's voice, and it works fine, from the command line i can do swift "hello there" -o test.wav and then

hi all, you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered

Hi all I have a couple of questions maybe you guys can help me with them I have sip phones , SER server , Asterisk. what is the best way to do that (also with accounting and authentication). which one of those options 1) sipphone -> SER -> ASTERISK -> SER -> PSTN 2) sipphone -> SER ->ASTERISK ->PSTN on the first option i am trying to return the call to the ser

hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it

Not too sure what to think about this, considering M$'s track record with OSS and other competitors. Kind Regards, Keith Roberts ---------- Forwarded message ---------- To: opensuse-offtopic <opensuse-offtopic at opensuse.org> From: Basil Chupin <blchupin at iinet.net.au> Subject: [opensuse-offtopic] Microsoft struggles to get Hyper-V drivers in Linux kernel The

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 It seems like Asterisk are having problems detecting DTMF digits when using an Adit 600 channel bank via MGCP. I've tried to turn on RFC 2833 on both Adit and Asterisk, but no digits at all are working then. Anyone experienced simular with Adit or other channel banks? I'm also unable to use V.90 modem through my setup (Adit600 via MGCP -

Hi guys, I have a weird thing happening sometimes with users calling from a GrandStream phone through Asterisk onto a PSTN. Sometimes after a user hangs up a call on a GrandStream phone the phone starts ringing after a couple seconds. When the call is answered there is no one there. Anyone had this before ? Kindest regards David Wilson _______________________________ D c D a t a Tel +27 33 342

Kernel news :) ---------- Forwarded message ---------- To: opensuse-offtopic <opensuse-offtopic at opensuse.org> From: Basil Chupin <blchupin at iinet.net.au> Subject: [opensuse-offtopic] Linus Torvalds delays Linux 3.0 launch due to a subtle bug *LINUX FOUNDER* Linus Torvalds has said that he's delaying the release of Linux 3.0 due to the discovery of a bug. Torvalds

Hi all. I fear I''m slightly offtopic with this question, but I''ll keep it short: who is (or feels) responsible for the #lartc irc-channel in oftc.net? I''d appreciate if that person would get in contact with me (off-list) in order to discuss some ideas I have. Thanks in advance. Bye, Mike _______________________________________________ LARTC mailing list /

Hi, First off, let me state that _YES, I am fully aware that what I am doing is insane, prone to major havoc and bad for general health_ :-)) Scenario: My GF needs an analog modem to use with her banking software (sodding backs don't supply a decent web-application for company use). I am experimenting to see if we can get it to work (albeit slow) trough our ATA186 talking g711 to

Hi, I have Asterisk 1.2 + Adit600 Channel bank(which gives analog output and also takes PSTN lines into the Asterisk system). Conversations recorded by the ASTERISK comes in two separate Files: xxxxxx.0-in (GSM Audio) for the Asterisk Extension Side of the conversation; xxxxxxx.0-out (GSM Audio) for the Caller's side of the conversation. I have Quick Time Player to playback the

Hi, Anybody out there running Adit600s? I have in my care an Adit600 channel bank connected to an old (version 1.0.6) Asterisk instance with MGCP. When trying a more recent Asterisk (1.4.21.2~dfsg-3+lenny1, Stock current Debian) calls fail. I have attempted to add the "slowsequence = yes" line to mgcp.conf. (It seemed to be the only likely candidate in the example files I found

Hi Yair, Please let me if you managed to fix the DTMF tone issue, which you were experiencing couple of months ago. If not can you share any advancement. I'm currently experiencing the same issue, I can make outbound calls but DTMF will not work when dialing IVRs. My configuration is asterisk@home 1.5, registering to Voip provider (Symbio), codec is g.729 and dtmf mode is set to rfc2833.

Hello, very sorry for posting something offtopic here, but I have been looking for a solution for quite some time now and all manual reading and google searching didn't help much. We have a MGE Comet EXtreme being connected serially to a Windows 2000 machine. We would like to achieve the functionality of a Network Management Card providing SNMP-agent capabilities so we can query the UPS