search for: asterick

...es)--- -- Executing Festival("SIP/1000-2915", "I can say numbers like") in new stack == Parsing '/etc/asterisk/festival.conf': Found == Spawn extension (mytest, 2890, 13) exited non-zero on 'SIP/1000-2915' and of course the call exits. Here is my /etc/asterick/festival.conf [general] host=127.0.0.1 port=1314 usecache=no cachedir=/var/lib/asterisk/festivalcache/ festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n Everything is running on the same box. I have rebooted... nothing is var log messages either. The local festival_client con...

...t reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have high speed internet. Currently we dial up an AT&T conf using regular analog phones. I don't have a great grasp as to what Asterick is capable of, but my thoughts were that perhaps with VoIP telephone lines (either hooked up to the company's network or just using a 3rd party VoIP provider such as Packet8, which is whatI have for personal use) and an Asterick server, that we could setup a VoIP conference bridge. Can someo...

...t;;tag=A463C4-E2A To: <sip:2486@192.168.0.2> Date: Call-ID: 31DCFDCF-10B011D9-80C88933-DAA922EF CSeq: 101 SUBSCRIBE Timestamp: 1096394892 Contact: <sip:2486@192.168.0.1:5060> Event: message-summary Expires: 600 Accept: application/simple-message-summary Content-Length: 0 asterick*CLI> 13 headers, 0 lines asterick*CLI> Using latest SUBSCRIBE request as basis request asterick*CLI> Sending to 192.168.0.1 : 5060 (non-NAT) Found user '2486' asterick*CLI> Looking for 2486 in voice-mail Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192....

I am having trouble locating webvmail on my * server. Is this a seprate porgram or does it come with *. I am running version asterick*CLI> show version Asterisk CVS-03/26/04-17:08:20 built by root@localhost.localdomain on a i686 running Linux asterick*CLI> Thanks Kurt __________________________________ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25¢ http://photos.yahoo.com/ph/print_splash

Hi All... Can Asterick detect distinctive ringing on a POTS line and answer with different configurations? Thanks...

...alls between 5+ people. 4x a month we > hold > several hour long conference calls during non-business hours. All of the > employees have high speed internet. Currently we dial up an AT&T conf > using > regular analog phones. > > I don't have a great grasp as to what Asterick is capable of, but my > thoughts were that perhaps with VoIP telephone lines (either hooked up to > the company's network or just using a 3rd party VoIP provider such as > Packet8, which is whatI have for personal use) and an Asterick server, > that > we could setup a VoIP confer...

I've got a fresh FC2 install and I'm trying to get the symlinks right according to the /usr/src/zaptel/README.Linux26 instructions. I've created two symlinks: /usr/src/linux-2.6 -> /usr/src/linux-2.6.5-1.358 /lib/modules/linux-2.6 -> /lib/modules/2.6.7-1.494.2.2 When I do a "make linux26", I get a million warnings and errors with the result being: make[2]: ***

...o 192.168.1.2 etc. 2) VOIP to someone outside in the U.S. 3) VOIP to someone overseas e.g. U.K. 4) Get a hardware card for the incoming line 5) Some extensions, perhap's Four (4). I don't see much in relation to point 1. What software (linux) can be used to connect VOIP to the astericks server ?? Regards...Martin

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12 WARNING[-1233155152]: channel.c:1860 ast_...

...5818e2837190dc735ff52f0c418cca3e@xxx.xxx.118.2 > CSeq: 102 NOTIFY > User-Agent: Asterisk PBX > Event: message-summary > Content-Type: application/simple-message-summary > Content-Length: 36 > > Messages-Waiting: no > Voicemail: 0/0 > (no NAT) to xxx.xxx.118.1:5060 > asterick*CLI> > > Sip read: > SIP/2.0 481 Call Leg/Transaction Does Not Exist > Via: SIP/2.0/UDP xxx.xxx.118.2:5060;branch=z9hG4bK6109fe24 > From: "asterisk" <sip:asterisk@xxx.xxx.118.2>;tag=as33cf63f8 > To: <sip:2486@xxx.xxx.118.1> > Date: Wed, 30 Jun 2004 20:...

...hidecallerid=no callwaiting=no callwaitingcallerid=no adsi=no context=inbound-analog channel=>1-2 The output of ztcfg -vvvvv is : Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. so far so good, but when i try asterick -cvvvvvvvv : == Parsing '/etc/asterisk/zapata.conf': Found Feb 11 16:48:15 WARNING[19083]: chan_zap.c:1072 zt_open: Unable to specify channel 1: No such device Feb 11 16:48:15 ERROR[19083]: chan_zap.c:7034 mkintf: Unable to open channel 1: No such device here = 0, tmp->channel = 1,...

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show peers *CLI> sip show peers Name/username Host Mask Port Status 301...

I'm reading the Asterisk book, 2nd edition, and it describes how one can set up voicemail to be delivered by IMAP to a voicemail folder. Asterisk can monitor flags on the folder so that the "message available" light on one's phone tracks the state of the "read" flag in the folder. One can either dial in for one's voicemail or listen to it from one's favorite

...rameters as a single character string. Now I know that sys.calls()[[1]] will give me the full text of the initial call, but the problem is to detach the ... above from that as a text string. If I could do that I'd be done. Here's the incomplete code with comments -- see the gap set off by astericks. rStd=function(x,...){ if(missing(x)) # must have a specified function { cat("Error: No function specified\n"); return(invisible(NULL)); } z=as.character(substitute(x)); # must include code here to check that z is the name # of one of our alt...

...t even write protected. The only readme file was 'README.DevKitLite' and sure enough, it was an explanation how to install the dev lite kit, which I DON't have. I tried it anyway ignoring the references to S100U. There is a astinstaller, that failed complaining of no openssl-devel. Astericks has been running for the past week with no obvious problems or alerts. I am informed that mandrake has renamed it to libopenssl-blah-blah, so I look at the astinstaller entry and change it. Still fails so I remove that check. It now builds and completes. It complains [root@carol etc]# ztcfg...

First off, I have to admit that I'm uncertain if this is the appropriate forum; I'd be happy for suggestions about where else to look. I'm doing this work on a stock install of CentOS-7-x86_64-Minimal-1810.iso, with no updates. I'm trying to create an RPM database from a custom set of RPMs. One RPM ('openldap-ltb' from the LDAP Tool Box project (ltb-project.org) has a

...fc5: NOKEY 0000000 / b i n / b a s h \n / b i n / s 0000020 h \n / b i n / s h \n / b i n / s * 0000060 h \n / s b i n / l d c o n f i g 0000100 \n b e r k e l e y d b - l t b That asterick where 0000040 (and its contents) should be is worrisome to me. To my eye, something is amiss. -- Paul Heinlein heinlein at madboa.com 45?38' N, 122?6' W

I get this error no matter what I do. I've tried switching to different locations on the TDM400p to no avail. I've got the card setup with the green module in slot1 and red in slot2. My config files are exactly like the "Configure Dev-Kit PCI (TDM11B)" example. http://www.digium.com/index.php?menu=configuration#TDM11B When I run ztcfg -v, I get: 2 channels configured.

Hi? 1. When a mobile user dial an IVR short code , mobile network able to divert that call to Asterisk platform.? 2. There would be web servers which are holds Voice XML . 3. Asterisk would be able to redirect the mobile request to certain Voice XML server accordingly. Just for like this setup , how do we install asterisk with minimum of asterisk modules ? Do we need to install complete

Hi If I do not use any DAHDI - hardware , can I ignore the DAHDI linux and tools installation ? Can I use just asterisk ? is there any dependencies while executing Asterisk with DAHDI modules ? Thank you Luke -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130607/b7c8e748/attachment.htm>