similar to: PSTN to SIP line ratio

Displaying 20 results from an estimated 4000 matches similar to: "PSTN to SIP line ratio"

2010 Mar 30
2
Priority based softhangup
Hi, Is it possible to softhangup a channel based on priority. I mean I want to put some calls in higher priority lets say 100. If all channels are busy and somebody wants to dial an extension with priority higher than 100. How can softhangup drop a line which has priority less than 100? I will appreciate your valuable help. Thanks Smir
2011 Feb 04
1
keep.source when semicolons separate statements on the one line
The following is 'semicolon.Rnw' > \SweaveOpts{engine=R, keep.source=TRUE} > > <<xycig-A, eval=f, echo=f>>= > library(SMIR); data(bronchit); library(KernSmooth) > @ % > > Code for panel A is > <<code-xycig-A, eval=f, echo=t>>= > <<xycig-A>> > @ % Sweave("semicolon") yields the following 'semicolon.tex'
2010 Jan 28
2
911, location
Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I
2010 Mar 16
1
softhangup
Hi all, I am trying to drop a random channel in asterisk 1.6. The following line in extensions.conf works fine for the first channel exten => 911,4,SoftHangup(DAHDI/1-1) But I need to drop random channel for emergency not any specific one. Can someone show correct syntax for this Thanks smir
2010 Mar 03
1
911, channel full
Hi, I am trying to implement 911 funtionality in my PBX. A call should drop if all lines are busy. Here is my context nineoneone from extensions.conf [nineoneone] exten => s,1,Set(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,Set(EMERGENCY=1,g) exten => s,n,Set(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
2010 Feb 26
3
: PSTN calls
Hi All, I have installed astriesk 6 and am able to make calls using sip x-lite. Its working as I expected. Now I want to make call from sipx-lite to PSTN using asterisk. can any please suggest me which Hardware card that I can buy? and use ( pl give me all ths list of cards.... which are good.). 2) what is that I need to do after buying the card to make it talk to the real world PSTN network?
2010 Oct 14
5
How to connect asterisk PBX to PSTN
Hello community, I have successfully set up asterisk free PBX server and I am also able to connect to it by softphone. Now as next step I want to extend this to PSTN , My Required scenario: I need a number which will connect outside PSTN world to my PBX and by applying extension particular softphone or connected normal phone should get connected. Which hardware I need for it. Also please
2010 Apr 07
3
PSTN issues
Hope some can help me. I have a PSTN coming into TDM400 into Asterisk. We also have direct telephones connected to the PSTN bypassing the Asterisk. When a call comes in on the PSTN the direct connected phones ring first and if no one picks up , Asterisk picks and get routed to internal sip phones. I am not able to find what I should tune to make the calls always go through asterisk without the
2013 Jul 01
2
[LLVMdev] [LNT] Question about results reliability in LNT infrustructure
On Jun 30, 2013 8:12 PM, "Anton Korobeynikov" <anton at korobeynikov.info> wrote: > > > Getting 10 samples at different commits will give you similar accuracy if > > behaviour doesn't change, and you can rely on 10-point blocks before and > after each change to have the same result. > Right. But this way you will have 10-commits delay. So, you will need
2004 Jul 14
4
aspect ratio ?
Can someone enlighten me on what the status is of aspect ratio in theora is ? The ti structure has aspect_num and _den values, which I assume give the intended display aspect ratio (e.g. 4/3). The sample files on the bittorrent seem to say both values are 0 for all files. I'd think it should at least be made impossible to have a 0 as the denominator. The library doesn't check the
2015 Jun 03
3
sedwards@sedwards.com causes me to be knocked off the list
Someone on this list uses the address @sedwards.com I doubt this is their actual email address as there is no MX record for sedwards.com and I can't find registration for their domain either. Part of my mail servers reject these emails because they cannot be replied to, or are likely to be spam. Every so often I get a mail from the list management to say that I've been unsubscribed
2005 Aug 23
2
SIP powercycle not hanging up
I have Sipura 841's talking to a CVS-HEAD of August 4. If I disconnect the power to the Sipura, Asterisk does not hang up the channel. My sip.conf for this phone looks like: ; [super1] ; Sipura 841 disallow = all allow = ulaw callerid = "super1"
2017 May 31
8
OT: Want to capture all SIP messages
I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
2011 Jul 13
2
TDM400p susceptible to EMI?
I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5" hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? -- Thanks in advance,
2016 Mar 18
3
Incoming INVITE with Portability Info and LRN
On Fri, 18 Mar 2016, Trey Hilyard wrote: > I thought this would be as easy as > exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
2019 Jun 07
4
Find out which key ended recording?
Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.
2012 Jan 06
1
Why write your dialplan using Lua?
Hello, Reading through the Wiki: "Asterisk supports the ability to write dialplan instructions in the Lua programming language. This method can be used as an alternative to or in combination with extensions.conf and/or AEL. PBX lua allows users to use the full power of lua to develop telephony applications using Asterisk" My question is, what is the benefit of using Lua? I recently
2010 Mar 24
6
Restarting Asterisk using a script - Thanks to all -
Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service
2017 Feb 07
2
Using g729 now that patents have expired
> On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards <asterisk.org at sedwards.com> wrote: > Now that the g729 patents have expired, how do we use g729 in > Asterisk? > > Will Digium be releasing a g729 codec for 'free' use or do we > download the 'free' codec off the Internet now that we can use it > without moral or legal
2010 Aug 28
1
Play a number of files to a caller
I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the