[Asterisk-Users] 64bit libs in /usr/lib
What is the proper way to use the Makefile for 1.2.0 so that my 64bit libs get installed into the proper place such as /usr/lib64 ? Right now they are being installed in /usr/lib and it is making packaging this software a pain. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] CentOS vs. Vanilla Kernel
On Mon, 2005-11-07 at 18:17 +, Julian Lyndon-Smith wrote: What is the opinion of this fine list - should I use the default CentOS kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable (2.6.14) Anyone got any clues / hints / tips on what should go into the kernel ? All views and comments appreciated :) Depends. Do you want to spend your time using the system and working on Asterisk, or do you want to spend your time tracking kernel changes, patching security fixes, tracking down kernel bugs, breaking rpm deps and working around that, etc, etc, etc... Red Hat puts a lot of work into making sure their kernel is solid and secure. They backport security fixes and bug fixes into their stable tree, 2.6.9. In my opinion, I'd rather let the folks that know the kernel work on it rather than spend my limited time on it. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] How to configure the communication between two Asterisk servers
On Tue, 2005-10-25 at 11:15 -0700, Tielin Xu wrote: How does server B receive the message from server A? Many thanks for your help. Nintendo eh? The Redmond office? Thats near where I live. So let me make sure I understand the problem. Server A needs to get information from Server B about where to send the call to, which will most likely be somewhere from Server B, since all SIP phones go to server B? Why not use switch? We do something like that. We have 'Pandora' which is at a remote location connected to PSTN. We have 'Asterisk' which is local and all sip phones are connected to. 'Asterisk' has a context in dialplan that lists all the sip extensions and how to dial them and whatnot. 'Pandora' has a line within the context of the incomign PSTN calls that says: switch = IAX2/Asterisk/sipphones thats it! Basically it 'includes' the sipphones context on Asterisk into the call plan for Pandora. Works great. Does this help you? -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Call Forwarding
On Thu, 2005-10-20 at 14:54 -0400, Dave Morrow wrote: Hi all. I am attempting to setup a dial plan which will allow me to forward an extension. I have followed the instructions in http://www.voip-info.org/wiki/index.php?page=Asterisk%20call% 20forwarding however it does not work correctly. Does anyone have some expertise they could lend. Not sure if it matters, but when I setup as in these instructions, and attempt to call forward my phone, asterisk logs when in fact I am attempting to forward to extension 8001 ; Post your extensions.conf excerpt where you're trying to do the forwarding. I do something as silly-easy as: exten = 5799,1,Goto(sipphones,5713,1) Which takes calls coming into 5799 and instead directs them to 5713 within the sipphones context. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Free DID's
On Wed, 2005-10-19 at 16:41 -0400, Sergey Okhapkin wrote: Callpacket.com has a free plan (up to 100 mins/month outbound, unlimited inbound, free DID). Do you have hints on using callpacket w/ Asterisk? -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Please recommend a phone
On Wed, 2005-10-19 at 16:39 -0400, Jesus Mogollon wrote: I'm in need of a phone that would blink a led to let the callee know that there is an incoming call. The GXP-2000 does this but I want an alternative to Grandstream. Any help is appreciated. Polycom IP301s and 501s have a red LED that blinks when calls are coming in. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem w/ Asterisk hanging when caller hangs up in voicemail
When I hang up in voicemail, Asterisk seems to stop responding. (hangup vs pressing # to disconnect). After that, no calls can be made until I restart Asterisk. In IRC, a developer seemed to think it had to do with me using switch = in my dial plan. Basically I never see the calling extension get the -1 signal. Can somebody help me figure out why this is happening and how I can fix it while still using switch = ? -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Problem w/ Asterisk hanging when caller hangs up in voicemail
On Tue, 2005-10-11 at 19:14 -0700, Jesse Keating wrote: When I hang up in voicemail, Asterisk seems to stop responding. (hangup vs pressing # to disconnect). After that, no calls can be made until I restart Asterisk. In IRC, a developer seemed to think it had to do with me using switch = in my dial plan. Basically I never see the calling extension get the -1 signal. Can somebody help me figure out why this is happening and how I can fix it while still using switch = ? My server setup is thus: sip - Asterisk - Pandora - Fujitsu PBX Asterisk and Pandora are server names. This pastebin has the extension.conf contents of both: http://pastebin.com/390790 I'm still trying to capture some debugging info from when this happens, problem is it doesn't always reproduce. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Where to get the latest SIP Firmware for Polycom Phones?
On Sat, 2005-10-08 at 14:57 +0200, [EMAIL PROTECTED] wrote: thanks for that, i knew already but it misses the actual version Oh yes, that new version. All it introduces is a digital signature on the firmeware, for use w/ the new bootrom and such that require digitally signed applications. (this is what I gathered from release notes and such, I could be wrong). I saw no need whatsoever to upgrade to that version. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Where to get the latest SIP Firmware for Polycom Phones?
On Fri, 2005-10-07 at 11:17 +0200, Kib Eki wrote: Hello, can anybody tell me where to get the latetest SIP Firmware 1.6.2 for the Polycom phones? http://www.freedomphones.net/polycom/files/ -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Voicemailmain automatic extension detection?
On Wed, 2005-10-05 at 15:46 -0400, Mason Loring Bliss wrote:
Is there a way I can have voice mail check calls coming from my internal
users automatically get to the right extension, without having the user
enter their extension?
I'm thinking that I could have the local SPA boxes translate, or have
each user live in a context where the extension in question exists
uniquely per user, but both of these seem kludgey.
Thanks in advance for clues!
I use this in extensions.conf:
exten = 999,1,Answer(); Voicemail call number
exten = 999,2,Wait(1);
exten = 999,3,VoicemailMain(${CALLERIDNUM}); This requires username of SIPs to
be their VM box #
Users are still asked for password, but an added 's' above (I forget
exactly where) will make that go away too.
--
Jesse Keating
GameHouse -- Systems Engineer
___
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[Asterisk-Users] Hanging up on VoiceMailMain w/out putting in password causes call lockup
I've got an issue w/ 1.2.0beta1, where if I call VoiceMailMain from a sip phone, and then either put in incorrect passwords or just hang up, I never get a Spawn Extension that hangs up the call, and my sip phone is not capable of making any more calls until I restart the daemon. Can anybody help me fix this? -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Hanging up on VoiceMailMain w/out putting in password causes call lockup
On Tue, 2005-10-04 at 11:12 -0700, Jesse Keating wrote: I've got an issue w/ 1.2.0beta1, where if I call VoiceMailMain from a sip phone, and then either put in incorrect passwords or just hang up, I never get a Spawn Extension that hangs up the call, and my sip phone is not capable of making any more calls until I restart the daemon. Can anybody help me fix this? Further information. This problem seems to happen if I hang up ANYWHERE inside voicemail, w/out using # to exit cleanly. For some reason Voicemail application isn't catching or passing the -1 on to * so that the call will be ended. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Recommendations for * monitoring?
On Tue, 2005-10-04 at 17:52 -0400, Charles Austin wrote: Hello, Can anyone point me in the direction of software to monitor channel usage on voice T1s? Using a TE410. The wiki documentation seems geared to SIP channel usage Thanks Charles Make sure you are CDR logging and maybe stuff it in an SQL database so that you can go back later and pull data info out and use it as necessary. We're using asterisk-stat right now for this. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Emergency calls - forcing through on channel
On Tue, 2005-10-04 at 15:37 -0700, 1 2 wrote: A native - buit in - emergency number feature for asterisk would have my vote. We're accomplishing this by having a POTS line directly into the * box just for 911 calling. A T1/PRI will service for all other incoming/outgoing calls. This ensures that the 911 line is always open, although it will only let one call through at a time -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Continue dialtone after pressing 9
On Fri, 2005-09-23 at 14:28 -0400, Brian McEntire wrote: I'd like to force a call to go out one line if we dial '9' first and then the number. Same for '8' only I will force it out a different line. There is a parameter or a method to allow the dialtone to come back after pressing the first 9... but I can't remember how to do it. ignorepat = 9 ignorepat = 8 Also, your phone digit map may need to be tweaked to allow for this as well. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Set Log Level for Messages log file
On Thu, 2005-09-22 at 22:33 +0200, Arik Funke wrote: is there a way to set the log level to the equivalent of: asterisk -vdc Or anything similar? I have I problem I cannot really trace with the standard log level. Have you looked at logger.conf and the full variable? -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Set Log Level for Messages log file
On Fri, 2005-09-23 at 01:10 +0200, Arik Funke wrote: Hi Jesse, According to voip-info logger.conf is for Win32 platform only. Is this wrong? If not, I do not quite understand how it will help with a linux system. I am a bit at a loss here... Thanks for the effort! That must be outdated info. I use logger.conf to process my logging on a CentOS4 x86_64 system. Thats not windows (: pandora*CLI logger reload == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger restarted pandora*CLI logger show channels Channel Type StatusConfiguration --- --- /var/log/asterisk/full.pandora.game File Enabled- Debug Verbose Warning Notice Error /var/log/asterisk/messages.pandora. File Enabled- Warning Notice Error Console Enabled- Warning Notice Error -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Web based application for call History
On Wed, 2005-09-21 at 21:44 -0700, Pradeepa Ramamurthy wrote: I am just thinking to develop this using Java and Jsp How to implement this?...Need help for the same This is all being reported by CDR tracking. We log CDR into a pgsql database, then this database can be queried by whatever application you want and formatted how you want. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap calls dropping just after answer
I've got a problem w/ zap calls being dropped right after they are answered. I have a log file: http://pastebin.com/368526 Everything looks OK except for the DEBUG[25563] chan_zap.c: Exception on 9, channel 1 that seems to come up quite often. As soon as the other end of the Zap answers (my cell phone), and I can even hear a half second of noise, the line goes dead and gets hungup. In my case, Zap/1 is the first channel of a 24 channel T1 line to a Fujitsu PBX. All seems configured right as I can dial out and the cell phone will ring. This is very frustrating, can anybody help out w/ this? -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Zap calls dropping just after answer
On Mon, 2005-09-19 at 16:37 -0700, Jesse Keating wrote: In my case, Zap/1 is the first channel of a 24 channel T1 line to a Fujitsu PBX. All seems configured right as I can dial out and the cell phone will ring. This is very frustrating, can anybody help out w/ this? This was solved. My T1 link to the PBX needed to use em signaling instead of fxs_ks. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Polycom 501 Multiple Line Instances
On Fri, 2005-09-09 at 15:39 -0500, Matthew Boehm wrote: I tried following the Wiki page regarding the Polycom 501 and having the same extension appear on all 3 line buttons (just like my cisco) but I'm having no luck. Has anyone else had success in doing this? Perhaps someone who has been successful can update the wiki? I followed the wiki except for I have it registering at the Register point of SIP Configuration. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Speaking of Polycom phones...updated ROM: ouch!
On Wed, 2005-09-07 at 14:18 -0500, Doug wrote: I again followed instructions here: http://www.voip-info.org/tiki-index.php?page=Polycom+SoundPoint+IP+501 So yeah, the instructions are a bit misleading. I had to set register to yes prior to the line information stuff. Without that the phone wouldn't register. Now it registers, and I still get 3 buttons dedicated to a single extension. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Speaking of Polycom phones...updated ROM: ouch!
On Tue, 2005-09-06 at 17:41 -0500, Doug wrote: After I did this, it appears that the Web interface for the phone won't change the settings, nor will it actually reboot the phone now. What do I have to set on the phone itself, so I can update info in the Web interface, and then restart the phone? What you need to do is 'clear local config' before you start making changes. Menu - Settings - Advanced ( - password ) - Admin Settings - Reset to Default - Reset Local Config Once you've done that and rebooted, you should be able to make your changes through web or on the phone itself. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Contact Directory on Polycom IP-501 phones
On Thu, 2005-09-01 at 15:59 -0400, Jeremy Melanson wrote: Hi Jesse. A couple questions.. What firmware version are you using? Bootrom 2.6.2.20032 Sip 1.5.2.0054 How does your phone get it's config (FTP, TFTP, Manual config)? Initially it got the config from TFTP w/ the new boot rom. After that I did manual config on the phone. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contact Directory on Polycom IP-501 phones
I'm testing out some IP501 phones and I ran into an issue. WHen I try to add a new contact into the directory, I am not able to. A window blinks really fast but the entry isn't saved. When you exit the Contact Directory system you get a 'Busy! Please try again' window. What the heck could be going on? -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Contact Directory on Polycom IP-501 phones
On Thu, 2005-09-01 at 13:04 -0700, Jesse Keating wrote: Bootrom 2.6.2.20032 Sip 1.5.2.0054 I rolled back to Sip 1.4.1.0040 and I can save entries, but the menu system is all different and not easy to navigate. This is not so good. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
