similar to: asteisk, sip & NAT

hi ---------modem.conf :---------- msn=240862922 incomingmsn=240866365,6365 device => /dev/ttyI2 group=1 device => /dev/ttyI1 ; ttyI3, ttyI4 ---------extensions.conf ;------- [sip] exten => _XXXXXXXXXX,1,Dial,Modem/g1:BYEXTENSION (Sjphpone) Call to : 024076xxxx result : --Executing Dial(Sip/roseau-6163","Modem/g1:BYEXTENION") in new stack -- Called g1:024076xxxx --

Does anyone Know if tha avaya 4621 IP phone work wiht asteisk? if it work it has featuras working Thanks Ignacio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051014/d3a65784/attachment.htm

Hello, I have a friend that is asking if he can use his Ericsson 3413 H.323 IP phone with Asterisk. I can't seem to find any reference to this phone on the Wiki... -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST

Got my latest Linux magazine (www.linux-magazine.com) and fetured is asterisk in home network. I've also been in contact with Novel/SUSE about their asterisk pakages. *Reinhard Max the maintainer. He has hinted at new packages for SUSE 10. The current ones work well (in production) however he is unsure about the new zaptel intergration.... but I'm keeping my fingers crossed! * -- A.G.

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

Hello, Since our server migration from OES1 (rsync-3.0.2-2.1) to OES2 (rsync-3.0.6-3.1), rsync backup do not work anymore. Here is some explanation : With a full verbose mode on the source server : rsync: writefd_unbuffered failed to write 631 bytes to socket [sender]: Broken pipe (32) rsync: connection unexpectedly closed (786 bytes received so far) [sender] _exit_cleanup(code=12,

as i am trying to use asterisk and install my newly purchased ( got it yesterday) digium cards. i am following the very detail steps of http://www.automated.it/guidetoasterisk.htm. but one thing did not seems right so i wanted to let enveyone know the page says: Once compiled make sure there is a copy in /usr/bin/mpg123 i think the location is /usr/local/bin/mpg123

I played around tonight for a while trying to place a call to the answering machine at FWD. It didn't work. I sniffed the connection, and it looks like asterisk sends out an invite. The gateway at FWD then sends an Invite back, and then asterisk responds with a "482 Loop Detected" error message. I have attached the output of a sip debug for this session. Contents of the

Hello, I just compiled and started Asterisk 1.0.2 following "Getting Started With Asterisk Version 0.1a" from http://www.automated.it/guidetoasterisk.htm I made only one change to default config files - I changed from using oss to alsa. I don't have any devices so far. I started asterisk from the command line: # asterisk -vc and I got this warning (this was also before I

hello according to this user guide found at http://www.automated.it/guidetoasterisk.htm#_Toc49248768 it says the following Voicemail - Please leave a message after the tone... Ok, so you've got the basics going, and it's great - if you happen to sit by you phone all the time. What happens if you are out/away from your desk/sleeping you'll miss those vital calls. We need to set up

Hi all, Is it possible and easy to make a CTI server with Asterisk? Florent,

I need to find some basic configuration files. Is there a place I can check out how to set up an office using sip telephone and Digium FXO and FXS ports? Don Moskaluk don@moskaluk.com www.moskaluk.com 416 737-8230 Cell 416 614-8230 Home --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date:

Hi, I'm trying to setup Asterisk on a Linux (redhat 8) machine. There are no analogue phones to be used & use is purely for internet traffic (SIPs). I've followed the setup guide from : http://www.automated.it/guidetoasterisk.htm But cannot get asterisk to run. If I type asterisk at the command prompt I get invalid instruction If I type ./asterisk start I get [FAILED] Can

hello is there a gui that would allow me to configure everything from phones, to extentions, to voice mail to basicly everything that asterisk can do? I did go to www.voip-info.org and none of the guis I saw there do the trick and the ones that come close aren't downloadable just wanted to see status on this thanks hank ---------------------------------------- My Inbox is protected by

Where can I get some asterisk books.. or tutorials..? I?ve been searching in google.. but I find just some tutorials explaining how to fast set up an asterisk server. I want to learn how to use it and how to make my own configurations. So, the thing is that I want to know what is the best book or tutorial that you know? recomendations? Thanks to everyone... Danko Miocevic

You got to be kidding about 53 calls being recorded at sametime is an issue. I have done at least twice as many on my dual xeon 3.4Ghz system and had no problem as clients like to record every call that goes through the system. Then again, in my system, the in and out channels are mixed first before they are written to the disk. ________________________________ From:

hi, i have connected a SNOM 200 to the asterisk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198

Hello veryone, I have a strange problem. I have an asterisk (latest from CVS) with latest oh323 channel driver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 -> asterisk -> DIAX :: sound is ok. When I perform an outbound call: DIAX -> Asterisk -> h323 :: sound is terrible and CPU load is 80% When I

i have problem to setup application meetme. i'm using asteisk-1.0.3 and sjphone as client. ++++++++++++++++++++++++++++++++++++ monchemin --------------------------------- Do you Yahoo!? Dress up your holiday email, Hollywood style. Learn more. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'.