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I have an analog phone connected to a Sipura 2100 which in turn connecteds to * over a 100mbps LAN. When I do ChanSpy on a bridged call, it causes massive jitter. When I attempt ChanSpy with a Grandstream GXP-2000 the monitored call is clear. Has anyone had this happen? Any suggestions? ScriptHead

I am seeking to implement the following behavor: When a headset on phone1 is picked up, phone2 rings right away, without any need to dial numbers on phone1. Is this possible to implement? ScriptHead -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060120/0892441d/attachment.htm

...alizer and modify it enough where it could be distinguished from other voices in a conference call. This could make conference calls much less confusing. Perhaps the easiest way would be to use sox as the equalizer but I am not familiar enough with * to know how to put a channel thru sox. Anyone? Scripthead

...doesn't allow for such thing to happen. So far the only way I see to do this is dumping the sip_friends table setup in favor of Asterisk RealTime Static ( http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static) which seems to be quite an ugly solution. Am I missing anything? ScriptHead -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051002/a5f33d17/attachment.htm

...Discussion Subject: [Asterisk-Users] How to have a phone ring another extension as soonas off-hook? I am seeking to implement the following behavor: When a headset on phone1 is picked up, phone2 rings right away, without any need to dial numbers on phone1. Is this possible to implement? ScriptHead -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060120/ab68ceae/attachment.htm

I have asterisk running user a user/group asterisk/asterisk like so su - asterisk safe_asterisk and one the processes utilizes way more CPU than any other. According to htop, it used 7:59:XX of CPU time. Once I kill asterisk and restart, another process does the same thing while others are running smoothly. This doesn't look like standard behavior to me. Is this some sort of a master

I am curious if anyone has tested Asterisk running transcoding G.729 -> G.711 (ULAW) on Sun T1000 or T2000. I'd like to hear about your experience. ScriptHead -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060511/a6492f6c/attachment.htm

I am setting up * to accept incoming calls and route them to our reps. What I'd like to do route the call to the rep who has been idle the most, thus distributing the load among the reps. I can't seem to find this functionality. Can someone point me in the right direction? Script Head

I have minor echo on an IAX2 channel when using Firefly and a head set. I have tried various headsets and settings but still a little bit of the echo remains and I'd love to get rid of it. After some research I stubled on zaptel/mec2.h but it seem that it works only on the ZAP channel. Is there something I can do on the IAX2 channel?

I have two T1s and I'd like to split my SIP traffic over the two. I am looking at this: http://lartc.org/howto/lartc.rpdb.multiple-links.html what bothers me about it is the note "Note that balancing will not be perfect, as it is route based, and routes are cached. This means that routes to often-used sites will always be over the same provider.". If all my traffic goes to the same