similar to: Lets try this again then! Q: SIP error from dialplan I suspect!

I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial("IAX2/firefly@89280250/3",

Ok I have spent the last week working on getting my small PBX to work. I will in the end only have 4 SIP extensions being either softphones of IP phones. Currently only 1 SIP config for testing. And at the this point it should be all fairly easy with all inbound and outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via IAX. Inbound does work in it's current basic state.

Hi, I looked at the docs and probably missed it: is there a way to set a dialplan on the GXP-2000? (to avoid having to press "Send") Thanks, -- "Computers are useless. They can only give answers." - Pablo Picasso

My cdrs are missing accountcodes for incoming calls from other asterisk servers.. I've seen a few people mentioning this on the list and the solution seems to be setting up a dialplan for incoming calls from a particular sip peer.. in my opinion this does not scale well at all and I am looking for a solution to correct this problem. example sip peer: [asterisk_gw] type=friend

Does anyone have any success using AudioCodes FXO terminating calls ? Ehsan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051002/5cfef736/attachment.htm

I was doing a daily make update for asterisk. On the 19th the new version compiled fine. I installed it. All of my ata 186's can call out to pstn etc. All inbound calls, the phones ring but when you pickup, just silence both local and remote with no complaints in the cli. I backed down to the r 1.0 1.0.3 on the 20th CVS-v1-0-12/20/04-18:24:52 it worked ok. Yesterday I did a cvs update on the

Ok I have a challange that I can't seem to find a way to fix it. My Voicemail in * timesout after 30secs without fail everytime no matter what I do. I have incomming calls comming in through Freshtel IAX2, if it goes to SIP extension when it is online it can hang on for what ever time the call goes for. If however it goes to the Voicemail it will timeout at 30sec and I can't seem to

I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get "you have" and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. -------------- next

Maybe I've missed it but I'm wondering if there has been any movement towards getting t.38 support into asterisk.. has there been any news? Where is t.38 support at? will it even happen? -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 279 bytes Desc: not available Url :

Haven't seen email since the 29th.. just testing. -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 288 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/96555713/mhess.vcf

I am currently experiencing intermittent silences with my asterisk system. The symptoms are as follows: * Both for incoming and outgoing calls, I (and other users) occasionally experience a brief period of silence. * The silence lasts anywhere from 3 to 10 seconds. * It is not due to silence suppression, because the silences generally occur in the middle of sentences. * Silences occur at

As my previous try on getting an answer was hijacked I thought I would try again. Ok I have a challange that I can't seem to find a way to fix it. My Voicemail in * timesout after 30secs without fail everytime no matter what I do. I have incomming calls comming in through Freshtel IAX2, if it goes to SIP extension when it is online it can hang on for what ever time the call goes for. If

Hi all! I need a simple plan for the following: *answer call *wait for 4 digit extension *send call to 4-digit extension entered. I tried the following, but that doesn't work... exten => 998,1,Answer() exten => 998,2,Background(agent-newlocation) exten => 998,n,WaitExten(20) exten => 998,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr) WaitExten obviously does not fill EXTEN with

Hi, I created a dummy dialplan where I ask the user to enter the age. [macro-age] exten => s,1,Background(my/age) ;;Play recorded message to enter age exten => s,n,WaitExten(10) exten => _XX,1,Set(AGE=${EXTEN}) ;; this line is not executing, instead dialplan is terminating with error given below. exten => s,n,NoOp(${AGE}) exten => s,n,GotoIf($[${LEN(${AGE})} >

I've got a Mobile-to-PBX gateway installed and I want the ability to dial from my mobile phone into my PBX and next dial a land-line from the PBX so I can make cheep mobile-to-land-line calls while on the go. I've contemplated using the WaitExten application but it only seems to wait for ONE digit! Is there a way to put the calling mobile phone into a context and wait for a full-length

I have to design a dialplan for mulitple contexts (multiple companies) and I'm not sure how to go about it and I thought someone may offer help. Here is some background. There are three separate companies, let's say A, B and C. Each has their own context and each has their own set of numbers (these are just examples, not the actual config): [ContextA] exten =>

Hello, I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI as channels. A call comes in via IAX2 and should be redirected to CAPI. So I wrote the following dialplan: [fromiax] exten => _8XXX,1,Answer exten => _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r) [fromcapi] exten => 265,1,Answer exten => 265,2,Dial(IAX2/PoC/11@from-lw) exten => 265-BUSY,1,Busy exten

I am revising our DialPlan strategy for our Asterisk system (1.4.2) and looking for some info on 'best practices' for this. Here's what I'm trying to do: I have an ACD menu that gives the caller the options as follows: - Press 1 for sales - Press 2 for support - Press 3 for customer service - Press 8 for a 'Dial by Name' list or enter the extension number at anytime

hi community, I'm new to this list & asterisk in general, so let me first say thx to everybody involved in providing such great tools & ressources!! I'm currently trying to implement a simple voicebox-system. for demonstration purposes, I've successfully connected my cellphone via bluetooth using the current chan_cellphone-patch on the current SVN-version of asterisk.

Hello, I am running Asterisk 11 on CentOS 6.x. I have configured several queues as follows in extensions.conf: exten => s,1,Queue(myqueue,rtnC,18) same => n,Background(user_unavail) same => n,WaitExten(10) exten => 1,1,Voicemail(1111 at my-vm,s) This rings the phones in the queue for 18 seconds. If no queue members answer, the caller is then prompted to press 1 and leave a