RE: [Asterisk-Users] Nortel Phones.
Title: RE: [Asterisk-Users] Nortel Phones. Say... no chance that you could post your server code somewhere? We've got some i2004's kicking around doing nothing -Original Message- From: Carl Sempla [mailto:[EMAIL PROTECTED]] Sent: Monday, October 25, 2004 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nortel Phones. On Monday, 25 October, 2004 22:19 : Jim Van Meggelen [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Currently the Nortel IP phones only support Nortel's proprietary protocol, UNISTIM. There is currently no way for Asterisk to directly support these phones unless: a) Nortel releases a standards-complaint firmware image, or b) somebody is able to write UNISTIM compatibility into Asterisk. Actually, I've done some reverse engineering on the UNISTIM protocol. I have a fully functional server able to handle such phones. If someone want to port my standalone code into asterisk, it would be great. -- Carl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vegastream 50 FXO with Asterisk
Title: RE: [Asterisk-Users] Vegastream 50 FXO with Asterisk Well, I've got my Vega 50 Analog able to pick any available analog port for an outgoing line. Let's assume we've started with the QuickStart instructions that came with the Vega: - First, add a new Planner Group. Give it a name (say, POTS), set the Lan to Active. Set the Cause to 34 - Modify the Inbound_From_LAN Profile - Remove the Plan that the Quickstart told you to add - Add a plan for every analog port you want to use - Set the Src to IF:99,TEL:8.* - Set the Dest to IF:xx,TEL:1, replace the 'xx' with the port number of the analog port - Set the Group to the POTS group This will set up the Vega so that if it receives a call of the form 8xxx, it will pick the first available analog port, and dial xxx. -Original Message- From: David Liu [mailto:[EMAIL PROTECTED]] Sent: Friday, February 06, 2004 1:07 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Vegastream 50 FXO with Asterisk Hi Adam, Could you show us your configs on Asterisk and on Vega so everyone on the list can have a guide to get Vega working with Asterisk? Thanks! David - Original Message - From: Low, Adam To: '[EMAIL PROTECTED]' Sent: Friday, February 06, 2004 12:47 AM Subject: RE: [Asterisk-Users] Vegastream 50 FXO with Asterisk I have a Vega 50 BRI working without any of the issues you mentioned, the dual SIP registrations is normal for most multi-line boxes enabled split users. Rgds, Adam -Original Message- From: Glenn Dalgliesh [mailto:[EMAIL PROTECTED]] Sent: 05 February 2004 20:11 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Vegastream 50 FXO with Asterisk Anyone have any experience configuring VegaStream's with Asterisk. I have run into a few of questions. 1. It appear that after turning on registrations I am seeing two request for registration per line sip:[EMAIL PROTECTED] sip:[EMAIL PROTECTED] What is purpose and how do I handle this? 2. DTMF btw Asterisk and the Unit I was unable to get rfc2833 to work successfully with inbound or outbound DTMF. Is this a known issue? 3. How is the best way to deal with dialout and selecting a free channel on the VegaStream Any general suggestions/experiences with regard to configuring a VegaStream with asterisk would be appricated. Thanks * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
RE: [Asterisk-Users] Still looking for small fxo sip gateway
Title: RE: [Asterisk-Users] Still looking for small fxo sip gateway You might want to take a look on the Wiki pages for VoIP, in particular: http://www.voip-info.org/wiki-VoIP+Gateways Offhand at our site we're trying to set up something similar (although a little larger, 10 FXO lines, but no requirement to pick which line the call goes out... our 10 lines are all overlines). Our Vegastream 50 FXO shipped yesterday (or perhaps this morning), so we should be getting it in a day or two. (BTW: I'm in Canada) There's been rumours posted to this list that Digium is coming out with a higher-density FXO card, and Woody mentioned a Voicetronix Openline12, which appears to be a 12-port FXO card. And I believe that Intel/Dialogic puts out some multiport FXS/FXO cards... -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED]] Sent: Tuesday, February 03, 2004 6:15 AM To: Asterisk-a-users-list Subject: [Asterisk-Users] Still looking for small fxo sip gateway I've been looking around for a small external sip fxo gateway, sending emails to possible vendors, etc, and can not seem to come up with anything that fits. Suggestions anyone? (No channel bank T1 card suggestions, please. I've also just completed an eval of the Mediatrix 1204 which does not support the requirements.) The market between two fxo pstn lines (pair of x100p's) and something around four to six lines seems to be lacking, or I'm looking in the wrong search engine (or something). I fully understand the economics of when a channel bank and T1 card becomes cost effective, including the eBay costs (and risks), etc. I've also heard the comments for months now that Digium is/will be selling something real-soon-now. Specifically, I'd like to use a 4-port fxo sip gateway capable of supporting four US pstn analog lines, CallerID, Touchtone, loop style supervision, and have the capability for asterisk to direct an outbound call to a specific port on that gateway. I think that implies each port must execute a sip register command successfully. It's also expected to accept incoming pstn calls directing those to a single asterisk. (I don't care about an IP dialtone, nat, etc, just a plain-jane two-way sip gateway.) If anyone is designing such a box and need professional eval, we can certainly work with you privately (off list to radamson @ routers dot com) to accomidate those needs. Anyone seen such a beast at a reasonable price?
[Asterisk-Users] X-Lite, X100P, and Speex
Title: X-Lite, X100P, and Speex I'm having a problem with using X-Lite to initiate a call via Asterisk out an X100P analog port, using the Speex codec. I've put in the registry fix for X-Lite and Speex so that works OK, and calling the echo test extension works. However, if I call out the analog port it appears that audio being initiated by X-Lite is being dropped, but audio being initiated from the analog line is being encoded and heard OK on X-Lite. /var/log/asterisk/messages keeps repeating WARNING[]: Frame too large and WARNING[]: Out of buffer space over and over again. Any ideas on what's wrong? (and if it's simply that one cannot use the speex codec with outbound calls, how would one configure asterisk to allow speex when it's a SIP to SIP call, but G.711 if it's a SIP to Analog call?) Oh, and using ztmonitor, it shows the zap channel receiving all sorts of sound, but no transmit. Asterisk 0.7.1 (Debian/Unstable package) zaptel 0.1.6
RE: [Asterisk-Users] X-Lite Asterisk: Speex iLBC not working?
Title: RE: [Asterisk-Users] X-Lite Asterisk: Speex iLBC not working? On the flip side, I have tried the registry fix with X-Lite build 1101 against Asterisk 0.7.1 (Debian/Unstable package), and it works (at least for speex) Asterisk 0.7.1 from Debian/Unstable Win2K X-Lite build 1101 -Original Message- From: Philipp von Klitzing [mailto:[EMAIL PROTECTED]] Sent: Thursday, January 29, 2004 8:36 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X-Lite Asterisk: Speex iLBC not working? Hi! check out the latests cvs it has two reg files that will fix xlite or xpro to work Doesn't work for me: WARNING[9226]: chan_sip.c:2087 process_sdp: No compatible codecs! - CVS from yesterday - Win 98 SE - X-Lite build 1082 - took registry file from bug note, couldn't import it, so added the DWORD entry for iLBC manually Question: In sip.conf it is allow=ilbc and not allow=iLBC, right? Anyway, neither of the two worked.
RE: [Asterisk-Users] 8 lines - best approach
Title: RE: [Asterisk-Users] 8 lines - best approach Hey neighbour! I'll be posting on here what sort of experience we'll have with 8 (actually 10) incoming FXO lines going to a Vegastream gateway... -Original Message- From: Darren Martz [mailto:[EMAIL PROTECTED]] Sent: Friday, January 23, 2004 6:10 PM To: Chris Albertson; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 8 lines - best approach Thanks Chris!! Running copper does not seem logical to me either. The last time I checked (Sept03), the cost of 8 lines in a T1 was almost double the cost of split lines. I have been considering NuFone, and have investigated it. We have also been looking for a decent IP based business phone, but I'll post a separate question for that :) I have three businesses all with the same problem. A decent T1 price would be the best, so I could centralize everything and only outsource the long distance side. I'm in Vancouver, BC Canada with the Telus Inc monopoly. What range of prices do most American telco's charge for a T1?? The price (in C$) I was quoted was $450/m plus $27/m per voice channel with zero features. Plus there was a $1200 setup fee. - Original Message - From: Chris Albertson [EMAIL PROTECTED] To: Darren Martz [EMAIL PROTECTED] Sent: Friday, January 23, 2004 8:49 AM Subject: Fwd: [Asterisk-Users] 8 lines - best approach First get rid of those 8 analog lines. then you'l have two options: 1) Have the local phone company provide you with a T1 line that you can plug directly into the Digium card. After all it seems silly for ther phone company to split out the lines to 8 pairs runs 16 coppr wires only to have you re-combine them. 2) Get 8 DID numbers from a VOIP provider like NuFone or Iconnect and have all your incomming calls come in over your Internet link. Now yu've got zero hardawre, except for the PC. I suppose you would want local extensions... Depending on the numbr you might want a channel bank and anlog desk phones or go with all IP Phones --- Darren Martz [EMAIL PROTECTED] wrote: From: Darren Martz [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 8 lines - best approach Date: Fri, 23 Jan 2004 07:30:42 -0800 I have 8 lines coming into an existing PBX system and am looking for a cost effective way to replace the existing system with Asterisk. We need some of the features in Asterisk, including its ability to support remote offices (long distance savings). At first glance this appears to require a T100P card and a channel bank, but that seems rather expensive. My estimated price on that would be roughly $2600 for 8 lines given that system - perhaps my estimate is way off Is there another way that is more cost effective? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 8 lines - best approach
Title: RE: [Asterisk-Users] 8 lines - best approach One solution that we're investigating is using a gateway product instead of a channel bank. There's a couple to choose from...take a look at http://www.voip-info.org/wiki-VoIP+Gateways Sounds like you want one of the FXO devices. We haven't actually purchased ours yet, but we're looking into buying one of the Vegastream 50 Analog units. 10 incoming FXO ports. Saves us the cost of a proper channel bank, and a T1 card for the * server. I'm not recommending them specifically (just an example... you may be able to get better prices from other resellers), but Atacomm (http://www.atacomm.com) lists the price of a Vega 50 FXO at $2350, a Multitech MVP810 at $2999, AudioCodes MP108 FXO at $1429. (US $) -Original Message- From: Darren Martz [mailto:[EMAIL PROTECTED]] Sent: Friday, January 23, 2004 7:31 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 8 lines - best approach I have 8 lines coming into an existing PBX system and am looking for a cost effective way to replace the existing system with Asterisk. We need some of the features in Asterisk, including its ability to support remote offices (long distance savings). At first glance this appears to require a T100P card and a channel bank, but that seems rather expensive. My estimated price on that would be roughly $2600 for 8 lines given that system - perhaps my estimate is way off Is there another way that is more cost effective? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Debian Packages and Mirrors
Title: RE: [Asterisk-Users] Debian Packages and Mirrors Note that there are also asterisk packages in the standard Debian repositories http://packages.debian.org/cgi-bin/search_packages.pl?keywords=asterisk=names=1=insensitive=all=all v0.1.11 in stable, v0.5.0 in testing, v0.7.1 in unstable (unless you're not on an i386) The source for the zaptel interface is there too (package name: zaptel). Haven't looked for libpri... we don't have a PRI service...) What do you have different in your packages? -Original Message- From: William Waites [mailto:[EMAIL PROTECTED]] Sent: Friday, January 23, 2004 8:09 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Debian Packages and Mirrors FYI and to whom it may concern, I have made Debian packages of Asterisk et. al. You still need to build a new kernel and the zaptel modules from source, but Asterisk and libpri are manageable with dpkg. The debs as well as mirrors of the source distribution are here: http://www.ntgos.com/Projects/Asterisk/Download http://parc.styx.org/asterisk I would also like to mirror the CVS repository as well as set up a cvsweb... -w -- /~\ The ASCII Ribbon Campaign \ / No HTML/RTF in email X No Word docs in email / \ Respect for open standards
[Asterisk-Users] Channel Banks
Title: Channel Banks OK, I'm having some trouble finding which equipment I need What I'd like to do is take about a dozen incoming analog lines and bring them into an * server. Of course one is going to have a hard time fitting a dozen X100P cards in a case, so an alternative would be a channel bank and a T100P in the * server. Now here's where my confusion comes in. I _think_ I need a channel bank that has a T1 interface on one side (to go to the * server), and FXO interfaces on the other (to accept the incoming analog lines from the telco)? And are there suggestions out there as to which channel banks one should select for this sort of deployment?
