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Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090708/cccd4587/attachment.htm

Hi there. My problem, I can't figure out how to ask this question. So, hopefully someone out here can point me to the FM on this. I would like to have either a web page or an application that I can view that whenever a call arrives on the Asterisk server the application will display the callerid information. I've found quite a few examples of the reverse of this. To where a script is

Hello all, I am using asterisk 1.4.17 together with misdn, once in a while: -when a call was put on hold -the operator tries to call a internal party for transfering the call -the internal party doesn't answer the phone -the operator wants to get the external line backup again by putting the call "off hold" And then the external line is disconnected. an exact log of events is

I have setup a few Asterisk systems for customers using Digium TDM400 cards and Aastra phones. No problems with sound quality at all except at this one site. Every time I try their system I don't hear any problems but they tell me that it is really bad. They describe it a a loud scratching sound. Are there any tests that can be done to pinpoint the problem? Has anyone seen this before? Are

Hello Is het possible to assign blf to a misdn channel? I want to watch the status of my external misdn channels on a linksys 962, e.g. green = available , red = in use and as an extra I want when I press the blf use the external line or when busy i want to barge in that call. Did somebody do this before?

When there are 3 or more incomming calls on my Linksys 962, it often happens that the calls are in the wait-state but not visible on the phone anymore This happens when we want to transfer the call but the remote party doesn't answer. When we want to take the call back it is gone, but the calling party still has music on hold after a while this times out, and the call is ended by asterisk

At the end of the day SIP calles keep stuck in asterisk, is there any way to prevent this or debug this? The sip calls which get stuck all are calles on a krik IP600v3 dect gateway, I cant tell if they originate of the ip600v3, probably this are calls TO the IP600v3 10.0.0.71 240 2c2cfcc47ca 05593/103700 0x0 (nothing) No Tx: BYE Done 10.0.0.71 238

Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok.... Can somebody point me out what i could have done wrong? my musiconhold.conf

Hello all, I know it was on the list before but i have some questions about the Kirk IP600v3, the requested configuration files were send private i guess Does anybody have the correct SIP settings for handsets connected to the Kirk. IP600v3 I am particulair intrested in settings regarding: -Voice Mailbox -Call waiting -DTMF settings for e.g. parking an extension with asterisk functionality