similar to: dial out and message playback

Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the "SAY NUMBER" and "GET DATA" agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and

Hi, I am facing this issue, where I get a delay of aroud five seconds when switching between contexts (through extension.conf) . This is how my extensions looks like. [salesivr] exten => _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id ${CALLERID}) exten => _X.,2,Playback(emptyy) exten => _X.,3,Background(Main_Sales) exten => _X.,4,WaitExten(2) exten => _X.,5,Goto(_X.,3)

Hi, I am a beginner to Asterisk. Can anybody clear my following doubts regarding the configuration needed? 1) What is the ideal system configuratin required?(like processer, RAM, h/d space etc) 2) How many connections it can handle at a time? 3) How many Virtual PBXs it can handle? 4) Whether Postgres or Mysql is best suited? 5) How many IVR's it can handle simultaneously? 6) How many

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 faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.

Sorry for re-posting this message - I am trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor. Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency. i.e a sentence which should finish in 4 secs finishes in much lesser time. Where can be the problem? any configuration issue? Thanks in advance. -------------- next

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

Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with

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

Some of you have been following our progress on http://farfon.convergence.com.pk as we blundered our way through the development of a low-cost ethernet IP phone that does IAX and augments the client options currently available for the kick-assterisk server. With help from the denizens of #asterisk and kind words of advice from Mr. Spencer and the rest of the gang ... we're proud to have

I'm looking to build an application which requires a phone to autodial when the handset is lifted off of the hook. Since the system will be available to the general public, I want a solid phone that will withstand abuse, but not be too expensive. I'm leaning towards using a Sipura adapter with an analog outdoor handset. However, I don't nor ever have had a sipura adapter, so I'm

I'd like to setup an autodial that will do the following: 1. First try call my multiple SIP phones at my house - if no answer within n seconds, try my mobile phone. Alternatively call both SIP phones at home + mobile at the same time. 2. Once someone answer in step 1 (on any phone) initiate calling to a specific phone number/extension Currently I have it working except that in step 1 I can

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

I'm trying to do realize the following: I have 4 condtions. If all conditions are satisfied I will paste("PASS") If any of these is not satisfied I will paste("FAIL"). But I have to paste the corresponding failure. ifelse is a good solution but for a 2 conditions. Maybe switch or something like this. Does anyone have an idea how to do? Thanks in advance. Adel -- View

Hi, I am having a weird issue with samba where once a week approximately at the same time users will lose connectivity, if i run wbinfo -u all users are displayed wbinfo -g all groups are displayed However running getent passwd only shows local-users, no remote users are shown.. To fix the issue I have to change the name of my idmap config and restart samba and winbind and everything works

Hi All, I have a doubt regarding join tables I''m having 2 models 1)Fac 2)Cont and both models have " has and belong to many" relationships so there are 3 tables 1)facs 2)conts 3)conts_facs then i''m fetching the data in controller as @conts=Cont.find(:all]) @cfacs=Fac.all(:joins=>:conts, :select=>"facs.name") but i dont know how to get the

Dear Listers: I have a couple of data frames to report and each corresponds to different condtions, e.g. conditions=c(10, 15, 20, 25). In this examples, four data frames need to be exported in a "pretty" report. I knew Perl has some module for exporting data to Excel and after googling, I found R does not. So I am wondering if there is a package in R for generating good reports. I

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

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone