similar to: Odd problem with asterisk

Hi..... I am new to this software, and I want to implement a client (SIP or IAX) with PHP or at least to pass the main functions (connection,call, transfer, hangup, call id etc) to a CRM. Does anyone know if I could achive a project like that with AGI ? Any example using AGI with PHP ? Do I have all the functionality with AGI ? What about call id ? What is depend on ? (As I know * does not

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

Hello, 1) On OpenIndiana 151, I configured nut 2.6.1 as configure --prefix=/opt/nut/2.6.1 --with-cgi --with-hal --with-user=ups --with-group=nut CC=cc The result is -e Configuration summary: ====================== build serial drivers: yes build USB drivers: yes build SNMP drivers: yes build neon based XML driver: yes build Powerman PDU client driver: no enable SSL development code: yes

Does anyone know if SIP will/is support on handheld PCs such as the iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems like a solution to provide a wireless based sip phone for any user would be possible. Handoff between access points might be problematic but most users I know would be using their PDA phone in an airport with free wireless or at the local cafe, etc, etc... Can

Type 'mpg123' at the Linux CL. (no quotes) If the version is anything other than 59r, you problem is solved. Go to the Wiki and search for Music On Hold. Do the install of version 59r ONLY as described in the docs. Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Neil A. Hillard

When using a PRI, after the remote party hangs up, asterisk tries to spawn a call to the "h" extension. Is this normal behavior for a pri to try to call the "h" extension to try to clean things up? Call Comes In: -- Executing Dial("Zap/1-1", "SIP/16464436000@AST-237.65") in new stack -- Called 16464436000@AST-237.65 -- Accepting call from

Hi, I am investigating possibility of using asterisk as an call center controller, i.e. Clients phone in, interact with IVR, if IVR is not enough get redirected to human consultant. There should be possibility for supervisors to connect to ongoing conversation. Expected traffic will not exceed 30 concurrent calls. Asterisk box should be connected to Siemens "communication platform"

---------- Forwarded message ---------- From: Akhil dua <akhil.dua.12@gmail.com> Date: Wed, Jul 4, 2012 at 10:33 AM Subject: To: r-help@r-project.org Hi everyone I have data on stock prices and market indices and I need to run a seperate regression of every stock on market so I want to write a "for loop" so that I wont have to write codes again and again to run the

Hi, I've had NUT 2.6.1 on OpenIndiana working with a Tripp-Lite ECO550. I've been playing around with the config files a little bit, and I must have done something that messed up the demon. I now get wiley$ pfexec /opt/nut/sbin/upsd -DD Network UPS Tools upsd 2.6.1 0.000000 listening on 127.0.0.1 port 3493 0.000486 listening on ::1 port 3493 0.001279 Can't connect to

I was looking thru the archives and did not see anything, might not have looked far enough back. I installed Asterisk@Home 0.7 today, and connected it to an analog port of our phone system. Using extension 200 and SIPphone lite I can get incoming calls, problem is reaching the other extensions on the phone system that I am connected to and also dialing outgoing calls. The phone system

Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI "confbridge show profile user <profilename>". It's an all-SIP scenario with RFC2833 as the DTMF protocol.

I am trying to install: perl-Compress-Zlib perl-Digest-Perl-MD5 perl-Net-DNS perl-Time-HiRes perl-Email-Valid perl-File-ReadBackwards perl-File-Scan-ClamAV perl-Mail-SPF-Query perl-libwww-perl perl-LDAP perl-Unix-Syslog perl-Mail-SRS perl-Net-CIDR-Lite perl-Mail-SPF which are all either noarch or x86_64 rpms yet it wants to install perl i386 version? I could ask on rf list but it wont let me

ok this is a wired problem. when i use X-Lite - after i register with asterisk X-lite sends a subscribe/notify request to asterisk to determine if the account has any messages waiting. if i create a sip.conf account using: user 12345 with a voicemail box 12345 - MWI works user jwolosuk with a voicemail box 12345 MWI fails (gets a 404 not found upon a subscribe) does anyone have a clue why

I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm working on an implementation of VoIP en Linux. I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a Red Hat 9.0 (*.*.*.172) with another softphone X-lite. Both of the softphones are registering and appear in the peers (sip show peers) with the good parameters of address and port. If I try to make a call, * receive the INVITE request and send a 404 NOT FOUND answer.

I see the link here <https://alioth-lists.debian.net/cgi-bin/mailman/listinfo/nut-upsuser>, but clicking it results in a 404. Before I waste everyone's time posting my question, I really wanted to search to see if it's been solved already. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm trying to place a call to asterisk using X-Lite. Asterisk is setup with some Grandstream phones. I can call from one grandstream extension to another. When I try to an extension with X-Lite, it comes back with Status of SIP/2.0 404 Not Found. X-Lite is not registered as asterisk extension. It is just sending a sip invite to extension@IP. Does the X-Lite need to connect to

I am trying to setup a simple pstn gateway using Asterisk and a X100p card. I have got everything installed using Redhat 9 and am able to load Asterisk. I also configured sip and I am able to connect to the asterisk gateway with Xlite on the windows side. I am able to dial 1000 and get the welcome message. What I am NOT able to do is dial a seven digit local or 10 digit long distance number and

Has anyone had any luck with the DEV LITE KIT? I'm getting very erradic behavior. I'll start it up and it will answer maybe one or two calls then hang leaving my phone in an off hook state or giving a loud shrill tone. It keeps saying "Red Alarm". Once it hangs it will not answer the phone again until I reboot. Starting and stopping Asterisk or reloading the drivers does not

I have just installed * from the latest CVS and I can make calls via X-Lite to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I will get the following error message: Spawn extension (default, Number-I-Dialed, 1) exited non-zero on 'SIP/200-6a22' -- Got SIP response 404 "Not Found"