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...ot;all" our incoming calls go to her mobile? Not just the calls to her extension. My Question: Does Call Forward on the Cisco Phones and Asterisk work? If so do I need to implement something into the dial plan. I have read on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding Is there an updated way to do this? I thought *21* was hard coded into Asterisk? If the Cisco phones wont work, i would like her to simply dial *21*<mobile number>#, any suggestions on this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... U...

Hi, Anyone knows why there's a limit of 32 callgroup/pickupgroup in * ? It is coded as unsigned int but there's an hardcoded "if( X > 31 )" like line. IMHO, 32 group is very low and I wonder what impact it would have to increase it to 2^16-1 . Anyone?

Yes, I have both Call Manager and Call Manager Express integrated with *. Prior to Call Manager 4.0 you would need to perform an H.323 integration with *. As of CM 4.0 Cisco supports SIP trunking so this would be the preferred method of integration. This config is on http://www.voip-info.org Seems like the site is having problems now otherwise I would have provided the direct link. I also

Ok, so you've read the Wiki and gotten call routing using ENUM to work (http://www.voip-info.org/tiki-index.php?page=Asterisk%20E164%20Call%20Routing) with your own ENUM-alike domain, e164.example.com. But how do you populate it with data? You can do it manually, but that gets very tedious very quickly. Or you can use the nifty DDNS updating program that comes with bind9. The first thing is to set configure your e164.example.com to a...

hello I'm trying to figure out if anyone's accomplished putting someone on hold with a hardphone that doesn't have a hold button or multiple lines. I'm thinking transferring the caller to a specific extension or something...is this possible? Has it been done? thanks hank

Hi, sorry for annoying question; i read http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup without understanding: 1. how to add an ext. to a pickup group (ie:. how to populate pickup group) 2. how 'Directed pickup' does work? "You dial the pickup number and your extension, and the call will only transfer if it is your extension" should i digit something like &...

Hi, I've got Outlook to call the number on * using the TAPI interface documented on the Wiki. Its working OK. I have downloaded the Indentapop application, and it appears to connect to * Ok using the Debug modes, but It isnt detecting incoming calls. Has anyone git identapop working? Care to share configuration details? Thanks

Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's "einen Ruf heranholen". It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to

I would like to use one of my Snom 200's 5 programmable keys to park calls. I am using image SIP 2.04g. I have tried a variety of combinations and have come to the conclusion that: 1) On the Key Mappings administration page, I must select the "Transfer" under the "Break Keys" option box to be able to successfully transfer calls using the "Transfer" button. 2)

Hi, is it possible to randomize extension which would be choosed by Dial()? I would like to forward phone calls to one of sales rep in randomized way (not to harm anyone;) ). tia mazek -- http://www.marcinmazurek.com/ ::: nic-hdl: MM3380-RIPE GnuPG 6687 E661 98B0 AEE6 DA8B 7F48 AEE4 776F 5688 DC89

It seems to me that ITSP's like to use a US dialing code eg 1-xxx Wouldn't it be cool to have an Internet dialing code?? I don't know what the structures are or how the allocations work but it would be so cool to know that 1-xxx was USA , 44-xxx was UK and yy-xxx was an internet phone.. That way the whole internet phone space could be consolidated into a single dialing structure

Group, Just want to share with the group my recent findings regarding CODECs/Vocoders and the effect it has had on voice quality and the intermittent noise and breakup problem I have which I mentioned in a previous emailing with the u-law CODEC. Calls again are placed through a SIP phone to a TDM400P to the PSTN. A good reference on the reasoning behind the selection of a CODEC was found in the

Fabio Pietrosanti (naif) wrote: > i am not criticizing the technology itself, but the fact that from the > point of view of the implementator that need to make a technological > choice while selecting a codec to be used for a mobile multimedia > application, speex does not provide affordable informations on > "supported platform".

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the