similar to: Timeout between digits for fxs station

Displaying 20 results from an estimated 4000 matches similar to: "Timeout between digits for fxs station"

2008 Sep 23
5
Extension registration
Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part
2008 Jul 16
4
asterisk + web services
List, We're working on an upcoming job that may require us to access a web service (WS). I'm curious to hear peoples thoughts on the best way to do this with asterisk. We'll be submitting a single number to the WS and it will return a success or error. One solution would be to write a simple perl script to interface into to the WS, and use SYSTEM() from asterisk to call it.
2008 Dec 22
1
Voicepulse down
Starting around 10:00 AM EST. All services from them whether I connect by IP or DNS (both east coast and west). Anyone else? Fred Posner fred at teamforrest.com Main: +1 (212) 937-7844 Direct: +1 (503) 914-0999 www.teamforrest.com -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Aug 03
5
Difference between WaitExten and TIMEOUT (response)
Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? Regards Bilal ____________________________________________________________________________________ Shape Yahoo! in your own image. Join our Network Research Panel today!
2009 Oct 19
2
Astricon talk on wideband codecs
I missed the talk that was given on wideband codecs @ astricon last week. I tried to lookup the speaker on astricon.net, but that website seems horribly broken at the moment, showing only a tmcnet video, whatever page i click on. Would somebody have the contact details for that speaker ? Greetings, Zoa
2005 Feb 09
1
Wait for Digits
Hi all I'm being really stupid today. i simply want asterisk to answer a incomming call, then wait for digits dialed. and then dial that extenstions but i keep on getting: WARNING[3314]: pbx.c:2017 ast_pbx_run: Invalid extension '5', but no rule 'i' in context 'zap-in' my config: exten => 0,1,answer() exten => 0,2,digittimeout,5 exten =>
2010 Apr 10
10
Being attacked by an Amazon EC2 ...
Just a "heads-up" ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've
2005 Mar 25
2
WaitExten question
I'm a bit confused about how WaitExten works. I assumed that when it returns 0, the next priority in the extension would be executed, but that doesn't seem to be the case. When I get to WaitExten and enter extension 8, it plays the message, then Waits another 10 seconds and times out. [local] exten => s,1,Wait,1 ; Wait a second, just for fun exten =>
2007 May 01
10
Digital Phones
Hi List; Asterisk does not have any kind of cards that can work with it to be used with Digital Phones (digital phones differ than analoge phone and differ than IP Phones). Anyone can advise about this as I did not find this on Diguim Regards Bilal Ghayad __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around
2008 Mar 12
3
DTMF problems while greeting is playing (Background())
Hi, I have a Digium TE410p T1 card and I've noticed that under asterisk 1.4.17/18 I have problems detecting DTMF in IVRs. I think I've narrowed the problem down to some sort of interference between the greeting that is playing and the DTMF tones. DTMF detection seems to work very reliably when I am in Read() or WaitExten(), but is absolutely unusable while in Background(). I hope someone
2007 Sep 25
5
Do I need to run #modprobe zaptel for Digium
Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? What #modprobe zaptel does a things that #modprobe wctdm does not do? Any help? Regards Bilal ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels
2009 Jan 25
2
Zaptel transfer using any button or code, but not flash hook
Hi List; I need to do a call transfer using analoge phone connected to fxs, but I do not need this to be done using flash hook, let it to be using the # or * or any code, but how I can configure that this code is for transfer? Also, I do not need the flast hook to be used for trasfer as it cause usually a confusion to distinguish between the hangup and the call transfer. Any advise? Regards
2013 Feb 03
2
RTP timeout if the asterisk box behind NAT
Dears; I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides). My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial the
2005 Mar 24
2
Fun with CAPI
Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension
2004 Jun 18
3
WaitExten substitute
i am using the freebsd port, which seems to not yet have WaitExten(), which i kinda want to use thusly [ext-666] exten => _.,1,SetVar(areacode=666) exten => _.,2,Background(zz-in-who) ; give them list of extns exten => _.,3,WaitExten(10) ; let them enter extn to call include => extensions include => applications include => speeddials
2013 Jan 02
3
DAHDI: How to know since when it is used? How to shutdown after max time?
Hi; How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how? Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it). What is happening with me that from time to time, I find some DAHDI channels are stayed connected
2007 Oct 19
3
ResponseTimeOut()
Hi List; My Asterisk version is 1.4 and I am trying to use the ResponseTimeOut() application to control the response time of the Background function, but when the running arrive for the ResponseTimeOut() then the call drop and my debuging says: No Application 'ResponseTimeout' for extension (Test_Bilal,s,3) Spawn extension (Test_Bilal,s,3) exited non-zero on 'Zap/1-1' Hangup
2004 Dec 21
1
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",
2006 Nov 03
4
some simple newbie help with dialplan needed...
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
2008 May 21
1
speex, ilbc and g729 codecs, GSM with IAX
Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no