Re: [asterisk-users] Show Callee name on Display
SInce no one else has brought this up, just thought I'd let you know that it is being worked on... http://bugs.digium.com/view.php?id=8824 And it works - I am using it for months already. Note that not all phones support it. Cisco and Policom supports it, while Snom does not. __Yehavi: ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
Hi, I appreciate the help. I called the vendor of the card and they recommended removing all of the PCI cards on the system (including the video card), and moving the card to a new PCI slot. I did all of them together, ran the system headless, and ssh'ed in remotely. It worked! haha... This must be proof that I have purchased a real piece of @#$. Thanks for all of your help. Date: Sat, 8 Sep 2007 02:41:50 +0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] New Installed X100p On Fri, Sep 07, 2007 at 08:45:11AM -0700, G B wrote: Hi Tzafrir, I am not sure what to look for, so I haveattached both the contents of /var/log/kern.log as well as the outputof dmesg. If you are looking for something specific, I simply asked for a few lines around that message. Anyway, the relevant lines are: Relevant lines: [ 39.337207] Failed to initailize DAA, giving up... [ 39.337283] wcfxo: probe of :00:0c.0 failed with error -5 No more details. This may be a defective card. I have also seen some cases where some voodoo at the PCI layer was required (e.g: passing the boot option pci=noacpi). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Can you find the hidden words? Take a break and play Seekadoo! http://club.live.com/seekadoo.aspx?icid=seek_wlmailtextlink___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configure extension by software
Before an IP Phones can be registered to an Asterisk server, the extension for it must be configured in Asterisk. Usually, Asterisk adminintor must add the extension by hand. Is there any library, API to do this by software??? For example, i want to develope a software that add new extensions to Asterisk system, sothat, any IP Phones can use that extensions to establish a call. I'm digging on Asterisk-Java but this library seems not support this. Anybody has dealed with this before ?? Phan Anh Vu DT12.K49.HUT RDLab ( C9.410 ) HUT - Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] redendent asterisk server for backup
Dear all I have asterisk server with 2 E1 port now i want to redendecy for my server means one of server goes down automatically second goes in active mode is it possible and how to switch E1 to second server ?? - Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool.___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Users Conference Friday @ 12:30PM EDT
ENUM and ISN You may be interested to know that John Todd was kind enough to come by at the last minute and give us a thorough grounding in ENUM and expand our knowledge about http://Freenum.org where you should run, not walk, to get yourself an ISN (ITAD Subscriber Number). You can listen to or download an mp3 of John Todd's talk or any other conference recording on one of these pages: http://www.voipusersconference.org/topics.php - topic agenda, download links and player or http://www.voipusersconference.org/astusers.htm - Flash player for recordings or http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 If any of you have guest suggestions or if you have something you would like to come and tell us about, please contact me. On 9/6/07, randulo [EMAIL PROTECTED] wrote: FRIDAY September 7th at 12:30 PM EDT http://www.asteriskusersconference.org for more information on how to listen, talk, or both :) This week, ENUM is the main subject, although our friends at e164.org haven't been able to talk to us as planned. Come on by and share what you know about ENUM or ask questions. Also, during Astricon, we are hoping people will call us with reports, either live or recorded and maybe someone will have some video? The IRC channel on Freenode.net is #asterisk_users_conference Past conference recordings: http://www.asteriskusersconference.org/topics.php rr ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
On Sat, Sep 08, 2007 at 12:42:34AM -0700, G B wrote: Hi, I appreciate the help. I called the vendor of the card and they recommended removing all of the PCI cards on the system (including the video card), and moving the card to a new PCI slot. I did all of them together, ran the system headless, and ssh'ed in remotely. It worked! haha... This must be proof that I have purchased a real piece of @#$. One note: The above description still does not completely exclude the possibility that the card wasn't originally placed well. (But then again, what do I really know about PCI cards) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configure extension by software
You can use asterisk realtime which can read sip config from database ( mysql/pgsql) . Your application can just write info to database and asterisk will read it and make peers . You can also include a custom config file within sip.conf and make your application write peer settings to that file and reload asterisk by using asterisk management interface . On 08/09/2007, phananhvu [EMAIL PROTECTED] wrote: Before an IP Phones can be registered to an Asterisk server, the extension for it must be configured in Asterisk. Usually, Asterisk adminintor must add the extension by hand. Is there any library, API to do this by software??? For example, i want to develope a software that add new extensions to Asterisk system, sothat, any IP Phones can use that extensions to establish a call. I'm digging on Asterisk-Java but this library seems not support this. Anybody has dealed with this before ?? Phan Anh Vu DT12.K49.HUT RDLab ( C9.410 ) HUT -- Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge http://us.rd.yahoo.com/evt=47093/*http://tv.yahoo.com/collections/222to see what's on, when. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configure extension by software
On Sat, Sep 08, 2007 at 12:58:44AM -0700, phananhvu wrote: Before an IP Phones can be registered to an Asterisk server, the extension for it must be configured in Asterisk. Usually, Asterisk adminintor must add the extension by hand. Is there any library, API to do this by software??? For example, i want to develope a software that add new extensions to Asterisk system, sothat, any IP Phones can use that extensions to establish a call. Any IP phone? What do you mean? If you just want to allow the IP phone to call out, it is very simple - place them all under the same user (or even as guests). If you also need to be able to call *to* them then you really have to decide how you identify them. Do you want to establish some sign-in process? First-off, Asterisk does not really have a concept of extension internally. Unless you use the users.conf hack and make sure it doesn't break. The manager interface (as of 1.4) allows you to change configuration files on the Asterisk server. Alternatively your interface can rewite files itself, or take some or all information from an external storage (real-time / static real-time). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Musiconhold instead ringing
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 7 Sep 2007 19:10:04 -0500 Subject: Re: [asterisk-users] Musiconhold instead ringing On Friday 07 September 2007 07:02:01 pm wassim darwish wrote: Hi: When i get an incoming call, i want asterisk to make the caller hear musicmusiconhold instead of ringing,Can any body help me with this? my guess is you'd have to Answer() the call first, then play moh while Dial()'ing the exten. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E Hi: Can you represent it in extensions Dialplan ? Regards; Wassim _ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Musiconhold instead ringing
Hi,
Am Samstag, den 08.09.2007, 09:44 + schrieb wassim darwish:
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com Date: Fri, 7 Sep 2007 19:10:04
-0500 Subject: Re: [asterisk-users] Musiconhold instead ringing On
Friday 07 September 2007 07:02:01 pm wassim darwish wrote: Hi:
When i get an incoming call, i want asterisk to make the caller hear
musicmusiconhold instead of ringing,Can any body help me with
this? my guess is you'd have to Answer() the call first, then play
moh while Dial()'ing the exten. -- Anthony - http://messinet.com -
http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967
92DC 35DC B001 4A4E
Hi:
Can you represent it in extensions Dialplan ?
Untested, but something like this should work
exten = _X.,1,Answer
extem = _X.,2,Dial(SIP/${EXTEN}|50|m(musiconhold-class)
You have to fix this to match Your existing dialplan (Extensions,
SIP-Accounts...).
It won't work without the answer-statement.
HTH,
Karsten
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Re: [asterisk-users] redendent asterisk server for backup
On Saturday 08 September 2007 04:52, satish patel wrote: I have asterisk server with 2 E1 port now i want to redendecy for my server means one of server goes down automatically second goes in active mode is it possible and how to switch E1 to second server ?? http://dataprobe.com/products/switch/k3/index.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Musiconhold instead ringing
From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Date: Sat, 8 Sep 2007 14:48:18 +0200 Subject:
Re: [asterisk-users] Musiconhold instead ringing Hi, Am Samstag, den
08.09.2007, 09:44 + schrieb wassim darwish:
From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Date: Fri, 7 Sep 2007 19:10:04 -0500
Subject: Re: [asterisk-users] Musiconhold instead ringing On Friday 07
September 2007 07:02:01 pm wassim darwish wrote: Hi: When i get an
incoming call, i want asterisk to make the caller hear musicmusiconhold
instead of ringing,Can any body help me with this? my guess is you'd have
to Answer() the call first, then play moh while Dial()'ing the exten. --
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89
5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E Hi: Can you represent it
in extensions Dialplan ? Untested, but something like this should work
exten = _X.,1,Answer extem =
_X.,2,Dial(SIP/${EXTEN}|50|m(musiconhold-class) You have to fix this to match
Your existing dialplan (Extensions, SIP-Accounts...). It won't work without
the answer-statement. HTH, Karsten
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Hi:
Thank you all for your suggestions,it worked well using Karsten suggestion.
Thanks again,
Wassim
_
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Re: [asterisk-users] Musiconhold instead ringing
I would try [your-coming-in-context] exten = s,1, Dial(type/identifier, timeout, m(your_moh_class), URL) Hoai-Anh Ngo-Vi Biedenkopfer Weg 13 60489 Frankfurt am Main Email:[EMAIL PROTECTED] Telefon: +49 (0) 69 74 22 36 63 Mobil.:+49 (0) 179 66 29 520 -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von wassim darwish Gesendet: Samstag, 8. September 2007 11:45 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Musiconhold instead ringing From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 7 Sep 2007 19:10:04 -0500 Subject: Re: [asterisk-users] Musiconhold instead ringing On Friday 07 September 2007 07:02:01 pm wassim darwish wrote: Hi: When i get an incoming call, i want asterisk to make the caller hear musicmusiconhold instead of ringing,Can any body help me with this? my guess is you'd have to Answer() the call first, then play moh while Dial()'ing the exten. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E Hi: Can you represent it in extensions Dialplan ? Regards; Wassim _ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to DUNDi branch office with area code?
hi: thanks a lot for your suggestion. i have setup up an experimental environment like yours, and dundi works great. but it's not easy for us to archive this in real world. we have other pbx(like alcatel) that need to co-work with asterisk. so area code with each branch office seems easier to maintain. hope we can run pure asterisk one day.. 2007/9/8, Bruce Reeves [EMAIL PROTECTED]: DUNDi can be used in branch office and I have a similar setup to what your are referring with 11 sites. One thing that I decided to do, but did not have to is define site extensions like you did, but I use the 4 digit extension locally and via dundi. Here is the details: In my case I check for the 4 digit extension in the current site then do a look up on dundi. Each site send the request to 2 core systems that keep up with all the peers and forward the request on.Once the extension is found in the dundi context on a server the call gets routed to the correct site.By doing it this way I don't have to remember to dial a specific number for an intra site call. The other thing is I do I have each site a set of numbers, like 3100 - 3199 is site B and 3200 - 3299 is C, but with DUNDi I do not have to do it that way, it will find the extensions since I use regexten=whatever extension in my sip.conf for each phone. I hope that makes sense, JR has done an excellent job explaining DUNDi in several white papers, and I have used something from all of them. On 9/6/07, d tbsky [EMAIL PROTECTED] wrote: hi: i am new to asterisk and dundi. we have some branch office which will use asterisk in the future. they will form a full-mesh structure so every site can contact each other directly. i want to try setup dundi, then we don't need to modify every pbx when a new site add in the cloud. thanks to the great dundi document caveman can do it and other resource in the voip-info.org. i learn the basic setup of dundi. but i want to a little advanced setup with area code. like this: site HQ: has extension 101,102,103, and site HQ has area code 99 site A: has extension 101,102,103, and site A has area code 01 site B: has extension 101,102,103 and site B has area code 02 site C: has extension 101,102,103 and site C has area code 03 we want to use 4 as prefix to call to the internal cloud. so user at site A can call 4-99-101 to contact extension 101 at HQ. site B can call 4-03-102 to contact extension 102 at site C. now i m confused about this structure with DUNDi. i don't know the best way to setup DUNDi for this structure. i think maybe i should do below when user call 4-99-101 at site A : 1. site A ask for dundi request 4-99-101 to site HQ 2. site HQ strip 4-99 and look up 101 at local context 3. site HQ return the destination to site A 4. site A use the destination to call extension 101 at site HQ i don't know if step 23 is possible in dundi.conf. the example in the internet didn't tell how to do this. or there are better/standard ways to do this? thanks a lot for any suggestion!! Regards, tbskyd ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Forgotten SIP session
Hi, I noticed today, that there was a stale SIP call on my 1.2.24 A*k server. One call (X-lite client) started yesterday into a meetme conference. For some reason the call stayed established. From network stats, I see transmit data but no receive (as obviously the client went offline). Luckily this won't of cost the company anything, as it was all soft/IP. However if it'd been PSTN based there would've been a cost which concerns me. A SIP SHOW PEER shows that the peer is offline. Is there a flag/setting somewhere to somehow control stale sessions like this? Should A*k not of closed the call down, say X minutes after the peer went offline? A. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
G B wrote: Date: Fri, 7 Sep 2007 07:13:38 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] New Installed X100p G B wrote: Hi, I just installed an X100p knockoff (OpenVox a100lp). I have seen my errors in the mailing list but without resolution. It seems that the other problems had to do with IRQ sharing. However, it doesn't seem like the X100p is sharing an IRQ channel with anything. Any ideas? snip Does OpenVox not provide support for their hardware? Thanks, Steve I purchased the card from a third party reseller. I am also contacting them for support. I'll send out an email if they have the solution. Yeah give them a try. I wouldn't even mention that you bought it from a 3rd party. Just play dumb. I am the last one to pickup a manual or call tech support but yesterday I was working on a very large industrial ShopBot (It is a robot so that is cool and it does really awesome things but why I was working on it don't ask.. http://www.shopbottools.com/applications.htm ) After trying a million things, briefly looking at the manual, I called the tech support line. The guy had me check two things, change one thing and everything was joyful. Had I done that from the start, I would have saved three or four hours (I bill by the hour so it's not so bad, but I couldn't bill the full rate since conscience told me not to) Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channels in use?
Barton Fisher wrote: I'm using version 1.2 and need a method to detect the number of channels in use from inside the dial plan. I'd like to count total channels system-wide, but even better if I can determine for a selected extension also. I've searched the wiki, and don't see such a function that does this. Any ideas? Bart Show channels or show channel. You can also break it down by technology which will even show you what codec they are using. Thanks, Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 to SIP conversion, standalone device
Alex Balashov wrote: There are lots of these. They belong to a class of appliance known as a media gateway. http://www.voipsupply.com/product_info.php?products_id=1038 If you REALLY want to pay that kind of money for something that serves this purpose for a single T1... well, we'd all love to have your budget! :) On Fri, 7 Sep 2007, Michelle Dupuis wrote: Over a year ago I saw a discussion about a standalone device which converted a T1 in/out to SIP in/out (over 10/100 LAN). Anyone recall what this device is? (I'm looking for a standalone device - not a PCI card). Thanks -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 Tenor DX (Room for expansion and more features than you will ever need) http://www.quintum.com/enterprise/en_productdetail.html?id=2 *The Tenor DX Series offers:* • Available with 2, 4, 6 or 8 T1/E1/PRI Spans • Up to 120 VoIP channels • Available in MultiPath or Gateway Configurations • Support for external Quintum Call Routing Server* • Support for external Quintum Tenor Monitor • Support for external Remote Management Session Server* • Intelligent Call Routing • VoIP and Tandem Circuit Switching • IVR/Radius AAA Compliant (Multilingual IVR) Thanks, Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
I have had Digium tech support tell me to do the same thing Thanks, Steve G B wrote: Hi, I appreciate the help. I called the vendor of the card and they recommended removing all of the PCI cards on the system (including the video card), and moving the card to a new PCI slot. I did all of them together, ran the system headless, and ssh'ed in remotely. It worked! haha... This must be proof that I have purchased a real piece of @#$. Thanks for all of your help. Date: Sat, 8 Sep 2007 02:41:50 +0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] New Installed X100p On Fri, Sep 07, 2007 at 08:45:11AM -0700, G B wrote: Hi Tzafrir, I am not sure what to look for, so I haveattached both the contents of /var/log/kern.log as well as the outputof dmesg. If you are looking for something specific, I simply asked for a few lines around that message. Anyway, the relevant lines are: Relevant lines: [ 39.337207] Failed to initailize DAA, giving up... [ 39.337283] wcfxo: probe of :00:0c.0 failed with error -5 No more details. This may be a defective card. I have also seen some cases where some voodoo at the PCI layer was required (e.g: passing the boot option pci=noacpi). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Can you find the hidden words? Take a break and play Seekadoo! Play now! http://club.live.com/seekadoo.aspx?icid=seek_wlmailtextlink ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configure extension by software
Thank you both of you Jaswinder Singh, Tzafrir Cohen. Though your guides don't directly solve my main problem but it gave me the light from the end of the tunnel. So now i can deal with my headache problem. I have 2 choices: (1) Use Realtime (2) Inclue configure file Thanks again to you and good luck. Phan Anh Vu DT12.K49.HUT RDLab ( C9.410 ) HUT - Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Udev issue on zaptel install
since the udev not installed in by the sequence, that may not supported in your distribution, use the correct version of udev for linux kernel version. i got the same problem with another device, udev wont create the device node automatically, if yours seems to be the same, this approach may solve the problem Hariharan.v RD Engineer, NEEVEE Technologies On 9/6/07, Tzafrir Cohen [EMAIL PROTECTED], rcom.com wrote: On Thu, Sep 06, 2007 at 12:48:44PM -0700, Markham, Craig (FRTC Contractor) wrote: Debian GNU/Linux 3.1 (Sarge). This version supports udev 0.056-3 , but it is not installed as a normal part of the setup process. Which is my problem...probably. Now I have to figure how to set this up. udev is not a prerequirement for zaptel. Debian Sarge uses devfs by default, and Zaptel supports devfs as well. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Build your own appliance concept
Only had 4 licenses for the G729 codec. Never had any trouble with these. Michael --Original Message Text--- From: Olivier Date: Fri, 7 Sep 2007 08:34:45 +0200 Michael, How many simultaneous calls could you get ? Regards 2007/9/7, Michael Graves [EMAIL PROTECTED]: I run Astlinux on a T5700. Have done for over a year. Works great. Much better than the Net4801 it replaced, especially using G729 codecs. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on Ubuntu Feisty
Hi all, Have just installed v1.4.11 of Asterisk, but I am trying to have it start at boot but with no luck. I have used the make config command but it doesn't start. Any help would be apreciated, many thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Difference in show channels
Hi all what is the difference between show channels sip show channles i see the difference in both show channels show me 30 channels sip show channels shows me 221 channels any description on this ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overhead paging over IP...
http://www.bogen.com/products/voip/index.html On 9/5/07, Dave Fullerton [EMAIL PROTECTED] wrote: Carlos Chavez wrote: I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. I have a similar setup where I work. I used a Viking PI-1 unit connected to the amp and a SPA-3000 connected to the viking. This gave me overhead paging and ringing. It did require a little tweaking on the SPA side because the PI-1 only provides 12V instead of the normal 48V to CO port. It's been working fine since it was put in about 8 months ago. I think the Viking unit was about $120. More info in it here: http://www.vikingelectronics.com/products/view_product.php?pid=199# -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
