similar to: Cisco MC3810 -> Asterisk

Hello Everybody, Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2: [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729) [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729 [Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing [Dec 28 21:06:00] DEBUG[1756]

Hi Everyone, I'm new to asterisk and trying to get together the hardware to run a few POTS phone extentions and one or two POTS lines for starters. For these low port counts, I could just go with FXS and FXO cards, but... I can get a Cisco MC3810 with a mixture of FXO and FXS ports, the MC3810 comes with a built in ethernet port and I believe it does SIP too... Will this mean that I

Has anyone here successfully gotten a cisco MC3810 talking with asterisk? I am getting the dreaded - Got SIP response 400 "Bad Request - 'Malformed/Missing URL'" back from xxx.xxx.xxx.xxx If you've gotten it to work you can feel free to email me off list. If your willing to share config's that also is a definate plus. Thanks, --Tammy

OK, I'm going nuts here trying to correctly identify null values, specifically when callerID info is not available. FYI, I'm running Asterisk CVS-HEAD-08/17/04-13:08:53, and Bison 1.875a (debian Sid). A snippit of my dialplan looks like this: exten => s,1,SetCIDNum(${CALLERIDNUM}) exten => s,2,NoOp,${CALLERID} exten => s,3,DBGet(temp=idiot/${CALLERIDNUM}) ; Is the person calling

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

Hello, I'm trying to run an ATA behind a NAT device, and am confused on exactly what the qualify config option does, other than send NOTIFY packets. Outbound calls work fine, but inbound calls do not go through. With qualify=yes and nat=yes, my show sip peers looks like: 7771111001/7771111001 10.0.0.10 D N 255.255.255.255 1222 OK (36 ms) So, it has established a

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

Hello folks! I'm trying yo set up a email2fax and fax2email on my asterisk box. The rxfax works fine in my setup. The problem is with the txfax. I have tryed all snadsp version (0.0.2x and 0.0.3x) but I get this errors. Because I can't find anything on Internet I'm hoping u can give me a hand. here are my logs: -- Attempting call on SIP/sip_provider/1234 for application

The document describes search(query, options) sort: An array of SortFields describing how to sort the results. I have created index with two fields: ''file'' and ''content'' When I give SortField name as ''file'' while searching, it results into error. The exact command given by me: index.search_each("sleepless AND dreams", :num_docs

Hi all, I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in my office. the out going calls symptom like when called party pickup the phone but the calling party still hearing the ring tone from the IP phone. Please light me up. it been many sleepless night by googling around trying to get the right answers. The digium card running on Intel 915G chipset. Below are my zaptel

Can rmovie be installed in a windows environment? When I run "gem install rmovie". It gives the following error: Building native extensions. This could take a while... ERROR: While executing gem ... (Gem::Installer::ExtensionBuildError) ERROR: Failed to build gem native extension. ruby extconf.rb install rmovie ''make'' is not recognized as an internal or external

hi all, just had a terrible and sleepless weekend at the office trying to get asterisk going, its just tough love ;) have tried several asterisk versions but i currently have the following setup on debian lenny that kind of works. asterisk-1.6.2.0 dahdi-linux-complete-2.2.0.2+2.2.0 libpri-1.4.10.2 freepbx-2.6.0 setting up the sip devices is no problem at all, the difficulty i have is setting

Hi Everyone, Merry Christmas :-).... My Asterisk Box doesn't have a sound card, it is running Asterisk 1.02 Zaptel 1.02 Libpri 1.02 Mpg123 0.59r All compiled from source with kernel 2.6.9-1.6 on Fedora Core 2 Any help would be very much appreciated..... The error I am getting is -- Executing WaitMusicOnHold("SIP/snom-james-849d", "30") in new stack Dec 12 00:27:29

is there a way to do element-by-element multiplication as in Gauss and MATLAB, as shown below? Thanks. --- a 1.0000000 2.0000000 3.0000000 x 1.0000000 2.0000000 3.0000000 2.0000000 4.0000000 6.0000000 3.0000000 6.0000000 9.0000000 a.*x 1.0000000 2.0000000 3.0000000 4.0000000

in llvm-3.6.2.src 1. when I put this around one of my instruction definitions in my target "InstrInfo.td" file, let TwoOperandAliasConstraint = "$dst = $rs1" in { } I do not see any TIED_TO in the generated GenInstrInfo.inc file for the OperandInfo used by the instruction, the question is what am I doing wrong ? 2. I've noticed that TwoOperandAliasConstraint

Hi I created a custom route that looks like this: map.connect '':controller/:table/:action/:id'' Hierarchically, :table is above :action and :id. So, if we call ActionController::Base#url_for with an argument :table=>nil, we''re supposed to get a url with :action and :id set to nil as well. However, this is not the case: url_for(:table=>nil) from

Ok, i change the subject of the old thread. (was thread : replication issues solved by adding GUID name ... ) i'v analyzed this problem.. I think where this is comming from. i did a : samba-tool dbcheck --cross-ncs -v same here 0 error. and thank you Rowland for this one. ldbsearch -H /var/lib/samba/private/sam.ldb -b

Ok, i'v analyzed this problem.. I think where this is comming from. i did a : samba-tool dbcheck --cross-ncs -v same here 0 error. and thank you Rowland for this one. ldbsearch -H /var/lib/samba/private/sam.ldb -b "OU=TOPFLOOR,OU=RTD,OU=COMPANY,DC=INTERNAL,DC=COMPANY,DC=TLD" -s base replPropertyMetaData --show-binary i didnt look exactly at the outcome but i saw the

I want to start a community based voip network projcet and am thinkimg of using VOCAL and asterisk gateways..... my question is, has anyone bench marked asterisk vs VOCAL? is it a wise idea to use VOCAL + Asterisk or Asterisk all the way.........am expecting 1000 -> 5000 users.. your thoughts would be appreciated. _________________________________________________________________ Don't

Hi all, sorry for the OOT. I've been trying to setup a sms gateway for a couple of sleepless nights now. I use Itegno 3000 USB gsm modem, gnokii sms manager, and playsms a web interface for sms http://freshmeat.net/projects/playsms/ on FC4. Here's what happen: 1. Using minicom, I can send sms 2. Using xgnokii (x interface of gnokii), I can send sms 3. Using playsms, I CANNOT send sms, it