search for: webcontrolcenter

Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages

Hi, Is there a way to create trunk groups while asterisk is running. For exemple let's say that zapata.conf defines g0 as channels 1-23 I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23 Any hints appreciated. Andre Courchesne

Hi guys, I''m setting up asterisk to run with another pbx server. This pbx server support a feature that allows 2 extensions connect to the same FXS. No I put asterisk in the middle. Asterisk receives the call and dial to a SIP/peer. How the pbx installed support 2 extensions to one fxs... How can I figure out in asterisk which extension was dialed before the call came to asterisk?

I should've mentioned that before. I've tried doing that and it has no effect. I've tried both upper and lower-case 'r's. I've also tried a workaround that I thought would work, but it doesn't: Answering the call and then using the playtones(ringing) command before connecting to my cellphone. -----Original Message----- Date: Sat, 1 Apr 2006 19:59:46 +0100 From:

how to adjust voice volume for sipura 2000 and cisco ata186?

i've got a asterisk server running on slackware 10.2 sice 3 weeks, i will upgrade the kernel to 2.6.x from 2.4.x , i will upgrade to the last stable release of kernel... i've got asterisk 1.2.4, zaptel 1.2.4 and libpri 1.2.2, correctly compiled and configured, obviusly if i try to load the new kernel i can't load wtcxx modules, should i recompile zaptel 1.2.4 with make && make

...can use SIP based hard/soft phones but using iax based ones is cheaper and faster. Alyed ---------------------------------------- Return-Path: <asterisk-users-bounces@lists.digium.com> Mon Oct 30 12:30:21 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Mon, 30 Oct 2006 12:30:21 -0700 I am looking for phones that work well (or at all) when outside of the network and behind a router, such as at someone's home or in a hotel. My Polycom IP601s do not seem to be up to the task, so I am hoping that there is a good alternative for...

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: Eric Pinet.vcf Type: text/x-vcard Size: 473 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20061116/beef984a/EricPinet.vcf

Hi! I have a E1 PRI connected to my TE400P card on span 1, and two channelbanks on span 3 and 4 and * 1.2. Every few hours I get this message and asterisk dies just after that: Warning: No D-channels available! Using Primary on channel 16 anyway! When this happens restarting zaptel and asterisk services, generally puts the system back online my zaptel.con reads: span=1,1,0,ccs,hdb3

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

Does anyone have some method, or AGI scripts that will automatically call a list of numbers from a database and play a pre-recorded message? For example, you have a database of FirstName LastName PhoneNumber Jon -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

Hi i'm planning to migrate a callcenter to asterisk and VOIP, the call center can have up to 25 cuncurrents agents logged in. I'll have some simplty IVR business logic and the some queues. Can a normal server with 1 GB ram 100 GB HDD Pentium 4 3.6 Ghz CPU Ethernet 10/100/1000 Support this? Would you suggest me a particular products? The server and the agents will be in the same LAN,

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

I am having a problem trying to get a Sangoma A101 to work with Unicall. I have installed the sangoma drivers and everything seems to be well but when I try to run ztcfg I get the following error: CAS signalling on span 1 conflicts with HDLC with FCS check on channel 16. Here is my /etc/zaptel.conf # MFC/R2 normalmente no usa CRC4 span=1,0,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101

Hello everybody, I am running 1.2.10-BRIstuffed-0.3.0-PRE-1s with florz-patches on Linux 2.6.16 with 4 HFC-Cards in TE-mode connected to 4 PtP ISDN-"Anlagenanschluesse". There are about 40 SIP-clients connected (mostly Sipura/Linksys PAP2, and some SNOMs and softclients) to this server. Everything works fine, except that my CDR reports some very long _concurrent_ calls from one sip

when i compile zaptel make linux26 make install i got these errors: make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp' make -C datamods clean make[1]: Entering directory `/usr/src/zaptel-1.4/datamods' make -C /lib/modules/2.4.27-3-386/build SUBDIRS=/usr/src/zaptel-1.4/datamods clean make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386' make: *** arch/i386/boot: No

Does anyone have the polycom soundpoint ip's successfully remotely provisioning? I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information. Any help would be appreciated :) -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk

I wrote a nice detailed post before, and then my mail program lost it for me... so here I go again... I've followed the same process with three different versions of asterisk, my local source copy from about 1 week ago CVS, current CVS from about 24 hours ago, and version 1.0.1, all three versions had identical results: I compiled/installed libpri, zaptel, asterisk I copied config file from

Hello, How do you guys implement LCR in Asterisk? Thanks,