similar to: How to use 1 channel from TE110P for data transmission

Dear all, I've a live system that needs to be upgraded but, before I proceed to the upgrade I want to assure the rollback process. That's why I'm requesting your feedback, in fact this asterisk in live system isn't going so bad but.... the upgrade is essential NOTICE that the upgrade will keep the same version 1.2 not from 1.2 to 1.4 Requirements: -backup /usr/sbin/asterisk

Hi guys, I know that this could be considered a bit off the topic, I've just posted this topic at VoIPSEC mailing list but I just thought this could be very interesting for Asterisk community members so I'm posting it here too. So the point is for traditional telephony we expect service availability of 99,999% and what about VoIP providers around the world what is currently available? I

Dear all, I've created a digium certified asterisk professional - dCAP linkedin group for anyone, dCAP, interested: http://www.linkedin.com/e/gis/60298/39AE1350DBF3 Best regards, Marco Mouta dCAP November 2006 -- Esta mensagem (incluindo quaisquer anexos) pode conter informa??o confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio pretendido, n?o dever? usar,

Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

Hi guys, First of all, I know that this server must be upgraded asap, I'm just wondering if anyone of you has already faced this problem and , if so, would the upgrade solve my problems... CAPI version 0.6 Asterisk 1.2.5 AGI scripts are being used Main problems: -Dropped Calls - ps aux | grep asterisk shows that asterisk (that is started with safe_asterisk) is generating multiple

Hello I want to use Asterisk to implement a SIP Domain allowing my clients to do URI dialing and receive calls from the Internet through URIs and ENUM. My question is, should I put my Asterisk outside the firewall (in the DMZ) to allow connections to the Internet? Or should I have it inside my local network and put a SIP Proxy (like Openser) in the DMZ to implement the SIP domain? Thanks

Hi, I am trying to implement call forwarding on the event that my ATA was not reachable to Asterisk, whether due to registration timeout, network interruptions between the ATA and Asterisk, or simply because the network on which the ATA resides in unreachable for any reason. I there a way of implementing such a feature in Asterisk? I have implemented CF unconditional, and CF on busy, CF on

We're going to be setting up Asterisk at our data center, as well as our call center locations via an optical fiber point to point connection. Is it best to have the servers communicate to eachother via SIP using SER, or just use the Asterisk functions? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL:

I just read on Slashdot (at http://yro.slashdot.org/article.pl?sid=07/12/16/222243 ) that Trixbox "has been phoning home with statistics about their installations", as a Trixbox user exposed in "Trixbox Phones Home" at http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home . -- (C) Matthew Rubenstein

Hi @ all, i set a server to a costumer of mine with a TE207P for use with 2 E1 Lines. I set them together into one group in zaptel/zapata.conf The point is now, the customer has a free-volumina of 60k minutes per month, per line. How can i make a kind of load balancing, that both lines will be trafficed the same way ? I read something about DIAL(Zap/r1/.) for using round robin, and

Hello. I am going through the documentation and trying to find if asterisk can help me in my case. It is quite difficult to find answer because I do not know the exact question. I have two location. Each in different country. Both locations have Siemens HiPath - different type and software. I can not use card that would allow me to connect those PBXs using SIP. But I have some free ISDN and

Dear all , Is there any subset regression (subset selection regression) package in R other than "leaps"? Thanks and regards Alex [[alternative HTML version deleted]]

Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response "all trunk calls are busy please try your call again later" Please how can i resolve this problem . I

Hi, I have our software with SIP running on it.I configured asterisk server as proxy. How do I implement the call screening features(incoming and outgoing) using asterisk server.Please suggest me how to proceed on this. Thanks & Regards, Jahnavi. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps

We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the "only telco's get documentation" crap) Does anyone have a suggestion? Thanks, MD -------------- next

I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works

Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:01xxxxxx at myip.com I an set an inbound route! It matches a DID number. How can I route an INVITE sip:s at myip.com? The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

Hello all, I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to send the voicemails as attachment to e-mails and delete the voicemails from my PBX once it has been sent. But, I don't have a running MTA here even on the PBX itself. I just want to send the e-mails to my GMail account from my PBX. Can I just use the mail or mailx command to send the e-mail and attach the