similar to: Credit Card machines / interop

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, I am having difficulty getting my record phone call dial-plan script working. I have tried the example record call scripts but they start recording before they are actually connected to an end point, e.g. you get ringing or announcements being recorded. It seems to me that these are bugs with the Dial() command: 1) Using M(x) in a dial command does not allow argument to be passed. Using

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 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

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.

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)

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 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 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 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 I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Rich Adamson > Sent: 08 February 2006 08:41 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update > ornot? > > However, if you expose the box to

Hi all, I have had quite a few customer complaints about attended transfer cutting off callers. The problem is when reception is busy she doesn't always wait for someone to answer the call, however hanging up a ringing transfer on attended also hangs up the caller. I have checked the scripts I don't *think* this is a dial plan error but if anyone has this working correctly on Asterisk

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

In the list I found some messages that *8 doesn't work so well. Is there any possibility to create a extention that you can call, and if you are fast enough, pick up a number? (Also if you are outside your callgroup) like pseudo code: exten => 888, 1, EnterPhoneNumber() exten => 888, 2, EnterPass() exten => 888, 3, TransferCallToThisPhone() exten => 888, 103, Invalid()

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 a micro USB cable and power supply adapter to run it but >>> after that it takes care of running

Hi everyone, I wrote a set of Mozilla Ubiquity commands that allow the user to search on rdocs, also featuring autocomplete, load of any rdoc hosted on the web and changing the default rdoc when searching. The project is hosted here: http://projects.talleye.com/ubiquity-rdoc and GitHub. Please, let me know any bug, suggestion, etc Luis Cipriani WebCo Internet brasigo.com.br

Hi sorry if this is the wrong place to post but I'm not sure where to go to and I'm a bit desperate. We just brought our Sunfire 6800 server and Storedge 9960 RAID array Solaris 8 back up after some maintenance and everything came back up 'clean' but I cannot get our SAMBA software to recognise any users or directories when logging in remotely from an apple or windows box.

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 a micro USB cable and power supply adapter to run it but > after that it takes care of running your software for you you don't have to > install anything on to your machine One would

Hey, maybe lynx package should be a requirement of the whois package...? ? $ whois ubiquity.io ? [Querying http://www.io.io/cgi-bin/whois] ? [HTTP: Unable to run web browser: /usr/bin/lynx: No such file or directory] Thx, JD