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When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

Ok I have tried to post to this list server but have just gotten the automated reply saying the moderator has to approve it to the list first which was my mistake for sending from the wrong email account. So if the moderator finally approves my questions and you see the same post again "Sorry". My situation is this: I havn't installed Asterisk yet but am curious the general way

Does anyone know if this unit works with Asterisk? Thx. B.

Hi, Companies that offer telephone service to the public are obliged to offer tapping to all kind of authorities. Does anyone know how to tap in Asterisk? I.e. record (or copy) a conversation based upon their telephone number? Thanks Dan _________________________________________________________________ Stay in touch with absent friends - get MSN Messenger http://www.msn.co.uk/messenger

Does anyone have any insignt on this? Any client programs that could be used? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030925/0e9e9920/attachment.htm

Does anybody know why I can NOT hear the MusicOnHold - using SJphone on another PC in our network (normal playback is not a problem) . See the * output and the line configured in extension.conf below (also mp3player does not function) Any suggestions? *Asterisk output:* *CLI> -- Executing WaitMusicOnHold("SIP/jeroen-bf54", "30") in new stack --

I used to be able to pass dial strings to IAX2 providers with # characters at the end of the string. This is how we end dial strings for international calls. So, I would like to be able to selectivity chop off any # characters at the end of string, only if they exist. Basically as follows (chopping off the leading '9' with ${EXTEN:1} syntax: EXTEN from Phone EXTEN for Dial String

Instead of using a PCI card is it possible to use an outside SIP service for "CO" lines? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030808/33b131e3/attachment.htm

Is there a way to determine what codec the remote server wants to use in a SIP session for an incoming call by looking at something, possiby sip debug? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030820/0692e436/attachment.htm

This looks rather interesting. They also have an IP phone which is probably low cost, but it seems to only support G.723. Has anyone used any of these products? http://www.nicstel.com/2001/e_3023w.html http://www.nicstel.com/2001/e_products02.htm -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am aware of a least a few people (including me) who were using the Packet8 service along with Asterisk for outgoing calls. Last night Packet8 did a software upgrade and both last night and this morning I have been unable to make any outgoing calls. Has anyone else noticed this behavior and/or been able to correct it? I get Got SIP response 403 "Forbidden" back from 4.42.235.170 when

Still I cannot get ADSIprog to work, I have tried everything. -- Executing ADSIProg("Zap/1-1", "") in new stack -- ADSI Available on CPE. Attempting Upload. == Spawn extension (co1, s, 1) exited non-zero on 'Zap/1-1' * Hungup 'Zap/1-1' It hangs up right after it says it is attempting to upload. -------------- next part

Is there any reason why this combination shouldn't work? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031105/bc043610/attachment.htm

Does MGCP transverse NAT? Seeing as the only decent yet cheap IP phone is the Swissvoice, it would be rather helpful.

The past week or so I have been experimenting with Asterisk and overall find it to be a nice software suite, although I have encountered some problems, and have found almost no documentation (For example in sip.conf I needed the commands fromuser= and fromdomain= and only figured out this was possible after spending a few hours browsing on the internet and reviewing some person's configuration

I am trying to get Asterisk to compile on FreeBSD 4.8. Per bug 389, BSD support should be in CVS. I have also tried applying the patch in bug 374, but always get these messages: click# make "Makefile", line 21: Missing dependency operator "Makefile", line 23: Need an operator "Makefile", line 72: Missing dependency operator "Makefile", line 74: Need an

What is the proper method to install/configure an X100P FXO card? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031009/ce8a7803/attachment.htm