similar to: Asterisk hosted solution

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

I cannot get the transfer button to work on a Snom 190, I cannot get the # to work either. Any ideas? Regards Thorben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041207/5ac06747/attachment.htm

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, 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

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

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

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

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

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:

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone <-----> Asterisk <------> ATA (FXS) <------> (CO side) PABX <-----> Extension (eg. 1000) (2100 & 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by

I am rewriting IPSwitchBoard at the moment. I want to make IPSwitchBoard "Skinable" meaning that you can design your own skins with company logo etc. You will also be able to add graphical extension buttons, and led's that will light up ex. DND, busy/free, message waiting and much more. If you are graphically minded and would like to help with making test skins using PhotoShop or

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our

Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (>5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to

I am installing one asterisk, to establish connection with my PABX Siemens, in ISDN, link went up normally, also I obtain to internally call the branches the PABX, normally, but when I try to dial for the PSTN, through pabx with the command exten = _ 19xxxxxxxx, 1, dial(zap/g2/${EXTEN}, 30) asterisk, reports me the following error: -- Executing Dial("SIP/8110-a729",

Hi guys! I'm having one or two issues with queues hosted by an Asterisk machine where the clients are on a legacy PABX - at least for the interim. I fully expect most of these issues to be non-resolvable, but thought I'd at least ask to find out if there is some way of working around the issues. The legacy PABX is an NEC 7400 ICS connected to Asterisk via an E1 ISDN link. Calls are

I have been working on solving a major issue with IPSwitchBoard. It was reported that IPS would use all available memory and get the PC to grind to a halt. I could not understand this as I had it running on many different PC's in Denmark. I now found the bug: IPS would crash on any PC that had "." configured as decimal point (in Denmark we use ",") this meant

Hello all! I'm VERY new in using VoIP. I'm looking for any tip or trick to connect a physically PABX behind an Asterisk-System(or similar) via an SIP to Analog- or ISDN-Converter. The point is, I _need_ to deliver calls to extensions in the connected PABX directly (in ISDN-speech "DDI" (DirectDialIn)) without intervention of an operator. Is this technically possible, and if

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil