similar to: Macro's and Var Scope's

Hi all, I got a couple of Snom 190's through this week and after some initial foolishness I have them both setup no problems. But here comes the except. When there is voicemail waiting the softbutton appears but the phone always dials its own extension. No matter what I put into the "mailbox" parameter on the line settings. (Phone registers correctly with * and all standard

Hi all, One of the areas I am trying to research before I can confidently start deploying Asterisk is "Credit Card Machines". (PDQ / Streamline machines / any similar) I'm talking about the credit card swipe boxes at point of sale desks. I believe they dial out to the specific bank provider everytime a card is swiped. My question is: - Does anyone have any experience using

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.

Hello, Does anyone know how to conference a call on the SNOM 200 phone? Whenever I push the cnf/tran button it just hangs up on the active call. The manual says you have to push the cnf function key but it doesn't appear in the lcd on my phone. Thanks -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Everyone: I have a THREE questions. What is a sip proxy and what is the benefit of having one with Asterisk? I am well aware that we have a sip channel in Asterisk and that we have SIP registration. I am not sure why you would need a SIP server and OR a registration server. Second question, with Asterisk are you able to do video on VOIP video phones? Last question, does

Hi all, I have looked through the wiki guides and also Siemens user guides but they haven't proven useful. Nor has the normally trusty googling. Also have upgraded to the latest Optipoint 400 Standard SIP firmware. Having read a few previous threads on the Optipoint it seems that there isn't much take up with Asterisk. Which seems a shame as my experience with testing it has been

Hi all, I am new to this list... Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I want to implement PC-to-Phone calls in the following topology (for signalling): SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> PSTN The RTP audio packets would go direct through Softphone to gateway. Does someone have a configuration file example of

Hi, I think this is one area that needs to be developed. I am curently implementing a system for my home so cannot really justify the cost of financially supporting the development of this when all I really need to do is buy a telephone extension lead for my existing fax modem!!! I am more than willing to devote some testing/documentation time (I am not really a programmer) if that helps.

Hi all ! Let's say you have about 6 small different companies sharing the same E1 / Asterisk server, and every company needs its own fax number. Since they don't really need fax machines, what would be the most cost-effective way to handle this (keeping fax-privacy at its best) ? Is there a way to configure Hylafax or sth & one modem behind an ATA-186 to email faxes to different

Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real "English" asterisk prompts. The only one I have found is here >> http://www.g7ltt.com/VoIP/vmfiles.html <http://www.g7ltt.com/VoIP/vmfiles.html> And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc)

On this macro I keep getting exited non-zero on s,3, but s,3 is doing what it is suppose to do but the macro stops. Is there a way to make a macro ignore errors and continue to s,4? I have the lattes ver of sox 12.17.4. Also if I just run this line from the command line I don't get an error. [root@redhat monitor]# sox in.wav in-rev.wav reverse [root@redhat monitor]# [macro-record-cleanup]

This is what I'm doing it gets you both sides of the phone call...small size...and playable on windows through a share. My notes: On redhat 9 I have to run the following command for asterisk to start LD_ASSUME_KERNEL=2.4.1 asterisk -vvvvgc [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${TIMESTAMP}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) ;exten =>

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Ext No Problems Panasonic Ext -> SIP Ext No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",

[apologies for no line wrap; config lines at bottom] I have mentioned on several threads here that the Monitor application doesn't do exactly what one would expect: the originating and answering legs of a call are unsynchronized by the duration of the interval that it takes for the answering leg to pick up the phone. This can be very distracting in a final mixed version of the file. Brian

I thought I would post this in case it might be of any use to anyone. Not anything special but it does work. Keep in mind you need sox and wmix. Here is some relevant exerpts of my extensions.conf using John Todds macro. [globals] CALLFILENAME=foo FOO=foo CALLERIDNUM=foo [default] exten => 287,1,Macro(dial,SIP/agent20002|20) exten => 287,2,Voicemail(u287) exten =>

I'm trying to set on-demand call recording. Here's a snippet of the pertinent dialplan. The purpose of this is to allow one user in particular to be able to receive an email recording of the call everytime he dials *91 + number. The problem is that the email is not going out or being generated when I use the ${CALLFILENAME} variable. When I use the actual file name of the gsm recording,

I have a question on how to handle the "h" routines. I have noticed that if the call is hung up by the side that originated the call, the "h" routine is not extendable via a macro, or at least I have been unable to do it. My tests have included only SIP->SIP calls. If the originating side hangs up first: The macro is called from "exten =>

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Hi all, I'm trying to make a context that will monitor a call and when it's completed it would e-mail the wav to a specified mail adres. So I made a standard context that records a call, like this: exten => _*31*00[1-9].,1,Setvar(CALLFILENAME=CALL-${EXTEN:4}-$ {TIMESTAMP}) exten => _*31*00[1-9].,2,Monitor(wav,${CALLFILENAME},m}) exten =>

Mike Burger wrote: > On 2018-02-16 9:29 am, hw wrote: >> Mike Burger wrote: >>> On 2018-02-16 8:16 am, hw wrote: >>>> William Warren wrote: >>>>> I would just buy a cloudkey and not have to bother installing the software >>>>> onto your machine directly.? If you do not have a power over ethernet >>>>> switch you'll need