similar to: Transfer on Snom 190

I am having a hell of a time with transfers. First the Snom issues: The transfer button on the Snom 220 does not work. I have read about setting break key off in the advanced page of the web config but the Snom 220 has no such option. At the moment I am having to use the # transfer hack which makes this phone look really stupid to have buttons on it that cannot be used. Anyone know how to

I realize why leaving email addresses in the archives would definitely _not_ be a good thing. However, is there a way to reply off list to an archived message? I would like to contact 'ulexus' about his asterisk/hdlc setup. Would anyone have his email address? Sorry for the interruption, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575

Hi all, I have just published my last few weeks of hard work: IPSwitchBoard BETA. Please let me know what you think and post comments on the Wiki. http://www.voip-info.org/wiki-IPSwitchBoard+BETA Thank you

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with

Hi, I believe the issues raised by this message are the same as mine, more on a commercial sense than for self use, but mostly the same. I've seen posts where real-life installations are mentioned, but not a reference to how Asterisk is working on production (and productive) environments. Any experiences would be very welcome I believe, not only on pure technical, but wider, sense. Thanks

Hi all, Is it possible to download Danish, German and French audio files for Asterisk somewhere, or does everybody just record them? Thank you in advance Thorben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041202/02cec53d/attachment.htm

Is it possible to park a call through the manager interface? If yes; how? Regards Thorben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050318/ec2a5f90/attachment.htm

http://voip-info.org/wiki/view/Easy+PABX With Easy PABX you can create your own virtual PABX online in just minutes. Easy PABX is based on Asterisk and best of all - it's completely free. Regards thorben.dk

When using buildinstall and CentOS 5.2 to create my own CD minus a bunch of unneeded RPM's and plus a few of my custom RPM's I find that the install CD that gets build does not install an initrd which renders the system unbootable. I also notice that when I run buildinstall with the -debug option it says: unpacking

I have a queue with linear strategy. When I add dynamic members it does NOT ring the members in the order they are added. I use the command "AddQueueMember" to add members but it seems to be random how it rings the members. Hope somebody can help. This is the description of linear strategy: *linear: Rings interfaces in the order they are listed in the configuration file. Dynamic

Hi All, Can someone explain to me the difference between g711's ulaw and alaw codecs? Is it just different header info or is the actual payload in each encoded differently? I have thus far noe been able to find any difinative information onthe matter. All I've managed to find out that they are "similar", they sound the same and that it doesn't matter which you use. Could

Release 0.66 of IPSwitchBoard is now available for FREE download at: http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA Enhancements: Support for Call Parking and retrieve/forward them again. Last Call on the Queues Page now displays a date-time in human readable format. Added CallerID on the Queue Members listing on the Queue page. New page with Agent information. Minor bug

IPSwitchBoard Version 0.67 Release notes: CRM integration, can call a web page with callerid when there's an incoming call. You can specify the min. and max. length of the callerid. Drop any active call. Help file integrated in IPSwitchBoard. Play button for sound files. Bug fixes - thank you for all your feedback. Download IPSwitchBoard for FREE here:

Hey, I just ran into a situation where I would like to expect a method call with an argument I know and another one, which is a random number. I think mocking up the rand method is somehow ugly so I thought maybe this is the first time where I can take something from Java to Ruby ;) Java''s EasyMock mocking library knows things like "anyObject()" and "anyInteger()" in

I am so far unable to get the busy lamps on a Snom 220 to work either with current cvs or asterisk 1.0. I am using the hint extension and the Snom 220 just as described in the "mini-howto" on: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg49781.html There are also a couple of wiki pages referencing this: http://www.voip-info.org/wiki-Asterisk+standard+extensions

Good day all I'm still looking for a SIP client that will work on fedora core 1? Thanks

Hello, As I understand * don't supports T.38 in Zap channels (please correct me if I'm wrong, BTW is there plans for such support?) I believe it's should support T.38 in "pass-thru" mode. I mean setup like this: Hardware gate with T.38 ------ Asterisk ------ Hardware gate with T.38 But I had troubles with this setup (no faxing) while two gates conneted directly with same

As far as I know, btrfs comes from "btree file system", but why does btrfs pronounce "butter-eff-ess"? -- Wang Shaoyan -- To unsubscribe from this list: send the line "unsubscribe linux-btrfs" in the body of a message to majordomo@vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Could you take 7960 and use the 6th line in a similar fashion to the all setup maybe? Thoughts ideas? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Can some people post some configurations they've implemented when deploying an * system for let's say 25-50 stations and maybe a larger 200 station system? I would assume some kind of chassis with some DSP boards and some kind of system board with a hard drive for running the system and storing the voice mails - obviously I'm interested in specific chassises and boards used and