similar to: delay in switching between contexts

Hi, I am facing some issues while using MixMonitor and StopMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b) exten => s,2,Dial(SIP/101,13) exten => s,3,StopMonitor() exten => s,4,NoOp(Dial Status: ${DIALSTATUS}) exten => s,5,Goto(sss-${DIALSTATUS},1) exten => sss-NOANSWER,1,VoiceMail(777 at salesvoice) exten =>

Hello, I''ve got DB like this: (roughly) create table projects ( id int auto blabla, name varchar(200), user_id int, sales_id int); create table users ( id int auto.., name varchar(50)); So project.user_id has a foreign key to user.id - all is good. Everything works fine. Now my question.. How can set a relationship from two different fields to the users table ?? Fx.

Hello, I am trying to do something and I''ve been reading my Agile Web Development with Rails book and doing internet searches for several hours now and I have made no progress. Here''s the issue: I have a table of sales reps that is linked to a table of territories. There are around a thousand territories that are numbered and each sales reps has many territories. I will

I am building a click to dial and CRM type web page and I'm having trouble with something. I can make everything in the manager api work as documented, but I can't seem to get a grip on how to tell what the callerid is of an active call. Example: I know that on phone SIP/101 that there is an active call that originated from the outside. What's the best way to get the callerid

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

I''ve been struggling a bit trying to figure out the best way to design/implement a system with authentication/authorization, and was hoping some of you may be able to offer some advice.. At the moment, I have a system with 4 different types of users - clients, administrators, sales_reps, and public_users. I''m using "Authorizing Users with Roles" from the Rails Recipes

Hi Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone Alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091213/1f81f008/attachment.pgp

Hi I am having some problems installing a paravirt RHEL guest on a deb amd64 xen machine uname -a = Linux debxen 2.6.20-1-xen-amd64 #1 SMP Tue Apr 24 22:23:22 UTC 2007 x86_64 GNU/Linux ============== xm info ============== host : debxen release : 2.6.20-1-xen-amd64 version : #1 SMP Tue Apr 24 22:23:22 UTC 2007 machine : x86_64

Building with --enable-llvm failed latter on when compiling libojc: configure:5099: checking for exception model to use configure:5131: /home/rafael/dev/gcc/build/gcc/xgcc -B/home/rafael/dev/gcc/build/gcc/ -B/i686-pc-linux-gnu/bin/ -B/i686-pc-linux-gnu/lib/ -isystem /i686-pc-linux-gnu/include -isystem /i686-pc-linux-gnu/sys-include -c -xobjecti ve-c -fgnu-runtime -fobjc-exceptions -S conftest.c

Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again ..... Alex

Hi I am in the process of installing a new box and using dahdi. I have a tdm410 + hardware echo canceller. I have just read in the read me for dadhi that VPMADT032 support has been removed and unlike with the zaptel stuff i could just download and install the firmware I can't with dahdi what is the best way forward to recompile with hardware echo canceller support. Thanks Alex

Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. I have read up about a PWR2400b and it seems to use 2wire pin, I am guessing to connect to P8 just behind the molex connector on the tdm410. can any one here confirm this, or have any info on the

Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of

Hi I was reading this article on installing asterisk 1.6 + debian http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr-and-realtime-config-on-debian and I noticed they suggested to recompile to 1000Hz enable kernel, I currently have a 250Hz stock standard kernel. I am running on a soekris board - amd geode cpu. Is recompiling the kernel to the

I''ve applied julian''s paches to a 2.6.14 gentoo kernel with the appropiate options enabled, and i''m using a modified version of the mpath2.sh script also available on julian''s site http://www.ssi.bg/~ja/ Overall everything works nearly perfectly. Incomming connections to either the fios (PPPoE) connection, or cable modem get routed back out correctly. The

Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc:

On Fri, 21 Apr 2006, [UTF-8] Rafael Esp?ndola wrote: > To work aroud the problem I edited gcc/objc/config-lang.in and removed > the comment from the line > #target_libs=target-libobjc > > With this change it is again possible to build without objc. The > complete configure line was > > ../llvm-gcc4-1.7.source/configure --prefix=$CFEINSTALL > --enable-languages=c,c++

Hi, I recently bootstrapped gcc4 frontend on a i686-linux-gnu too. As I notice, if the llvm "tools-only " was build with srcdir != objdir , the building of llvm-gcc4-1.7.source encountered mistakes of unable to find some header files. So I just edited llvm-gcc4-1.7.source/gcc/Makefile.in and a line like this: INCLUDES += -I$(LLVMOBJDIR)/include With this change, the building errors

Hi I'm currently running asterisk with an fxo X100P and aTDM one port card in a small not world connected subnet, I've sucessfully setup two sip phone and one analog extension everything works fine with the analog phone but when you talk to someone on the sip phone the person at the sip phone can be heard with inteference. I've looked aroud the archives and found nothing specific.