Re: [asterisk-users] AsterikNow vs Trixbox
Smartass... :) Trixbox works off FreePBX which, while not as tightly integrated into Asterisk, is currently far more mature and easy to use. Note the use of the word currently. :) I wouldn't be too surprised if FreePBX made the move to the Asteri/s/kNow framework. Removes the big ugly dependency for a web server. Peter Bowyer wrote: Trixbox is easier to spell. Apparently. On 11/02/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Comments? People's opinions ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pmp_l1_check=no with zaphfc (Bristuff)
There is currently a reimplementation of zaphfc (vzaphfc). Perhaps it is more robust. Which package include such reimplementation of zaphfc ? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound Call Transfer Problem
Hello, I see that you are using T option (allow the /calling/ user to transfer the call) when dialling to internal extensions and t (allow the /called/ user to transfer the call) when receiving calls (in home context). This it is why inbound transfer works fine and only one time. So, I suggest to add t option to the Dial lines you need the transfer feature to be active for the called party (eg. in from-sip context). Best regards, ## nini @ www.modulo.ro ## Nikhil Jogia wrote: Noah Miller wrote: I am using Asterisk 1.2 and for the life of me, I am unable to transfer outbound calls (eg calls I initiate from sip extensions). When I press #, nothing happens. Inbound calls transfer fine, but only once per call. Any suggestions? I have questions: 1) what version of 1.2? version 1.2.1 2) Anything come up in the CLI? How about the logs? Have you tried turning on verbose logging in logger.conf? (be sure to turn it off when you're done) nothing at all. 3) What SIP phones are you using (hard and soft)? sipura 2000 and sjphone. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disable root shell from CLI
Hi, I configured Asterisk to run as asterisk user, but I see that a user can anyway get a root sheet using !command in CLI. I understood that it's something related to safe_asterisk and TTY console, but modifying the script safe_asterisk I wasn't able to disable this root access. Can someone help me? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Asterisk's manager interface to recieve calls
What i need is to recieve a call in a console! I mean i can call from CLI...but can i recieve calls too? If this is possible how is the console identificated and where! Actually i need to call from one Asterisc server console to another(i know what is asterisc server for, but this is a specific task)! Thanks! - Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extensions in macro
I had the same issue when the us enterd a response. I used the read cmd, set a variable and then used some gotoif's - Original Message - From: Yuan LIU [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, February 12, 2007 6:49 AM Subject: [asterisk-users] Extensions in macro Home someone can explain this: a Goto() command can walk within a macro, but if a digit is dialed from within a macro, the call flows back to the context that called the macro. Is there some way to contain the flow? Thanks. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing callerid to ring groups callerid
This is real simple. in sip.conf do: callerid=Bjørn, Marius 966 do this setting for all the sip accounts that belong to 966 - Original Message - From: Bjørn Marius [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 12, 2007 8:38 AM Subject: Re: [asterisk-users] changing callerid to ring groups callerid thanks for your reply! :) Lets look at an example: I have these extensions: 566 - Primary SIP account 466 - SIP account for my Nokia E60 366 - SIP account used from home All these extensions are members of my ring group 966. When i make a call to any other ring group or extension i want my caller ID to be 966. This way the other employees dont need to worry about remembering the numbers for all my extensions. The trick is to make several extensions look like one ;) Dovid B wrote: So you want the caller ID for the user that is recieving the call to be the caller ID for the extension that was dialed for them. Is this correct ? If so then you can just change the caller ID in the dial plan with Set CallerID. Have a look here: http://www.voip-info.org/wiki/view/Setting+Callerid - Original Message - From: Bjørn Marius [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, February 10, 2007 4:18 AM Subject: [asterisk-users] changing callerid to ring groups callerid Hi all! First off all, sorry for my bad english. I have a setup where some of the users have several extensions(work, home, mobile etc). Therefore i have made a ring group for each of the users with more than one extension. The ring group is set up to use ring all. What i want is that no mather what extension a user calls from, I want the Ring-Group number to be the callerid. That way the other user only have to remember one number for each of the other users, even though they might have several extensions. Anyone? -- Best regards Bjorn Marius ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Med vennlig hilsen / Best regards Bjørn Marius L. Skulstad WebDeal AS Phone: +47 97 17 14 96 Mail: [EMAIL PROTECTED] Web: http://www.webdeal.no ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-Java 0.3 Milestone 2
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, we've just released Asterisk-Java 0.3-m2 at http://asterisk-java.org. The Asterisk-Java package consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. Asterisk-Java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and the Manager API. Asterisk-Java is free software distributed under the terms of the Apache License 2.0. Here is the Changelog: Bug * [AJ-47] - AGI does not support multi line data * [AJ-51] - Problems with non-english locales * [AJ-52] - Fix shutdown when using the live api Improvement * [AJ-41] - Add ability to get ManagerConnection from AsteriskServer * [AJ-49] - Support socket read timeout New Feature * [AJ-35] - Support timestamp property on manager events * [AJ-42] - Add support QueueSummary action to Queue manager interface * [AJ-44] - Support PauseMonitor and UnpauseMonitor actions * [AJ-45] - Support ZapRestart action Task * [AJ-53] - Refactor BaseAgiScript to extend AgiOperations and implement AgiScript so users can extend AgiOperation to provide their own add-on features Have fun, Stefan -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF0D1pcVCZDfrn+pMRAv9GAKCB1Ozij3l5eLGOu1pOx7kq+PHG4wCePsjR Jm+tu6reyyrDtVDGgATnTwI= =My0x -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pmp_l1_check=no with zaphfc (Bristuff)
Le 11 févr. 07 à 21:25, Tzafrir Cohen a écrit : On Sun, Feb 11, 2007 at 08:57:30PM +0100, Olivier MONNET wrote: Hi, I’m using Bristuff for more than a year and half now, and I am stuck with the same problem since asterisk 1.2. When using a card from Junghans, QuadBri or OctoBri everything is ok, (driver QOZAP) but with a one port generic card (driver ZAPHFC) when the channel go to sleep, it cannot dialout any more. The only way to get it back is an incoming call, and it work until the channel go to sleep . Here in France, I can ask the ISDN operator (France Telecom) to keep the channels up, but it does not work everywehere and generally the technician on the phone don’t known what I’m talking about. With asterisk 1.0, the driver is keeping the channel up, so it works fine. For my new installations, I have switched to mISDN, With mISDN, there is the option: pmp_l1_check=no which start the call even if the channel is down. Dialout work fine but there is some echo on many calls. I have never experienced echo problem with Bristuff. It is not possible to use a 4 ports card for small PXBs (1 bri) for cost reasons. I’ve tried to contact people at junghanns.net with no answers, I also tried to post on this list last year without finding any solutions. Does anyone ever experienced this and found a solution? We have a BRI driver of our own that uses the bristuffed Asterisk. It is currently in a testing phase, but we have been able to dial in and out to France Telecom. Anyway, we started the bristuff mailing list mainly due to the non-responsiveness of Junghanns. You can also find there my own fix of bristuff to build with the latest zaptel/asterisk 1.2 . We already use bristuffed zaptel 1.4 internally. The bristuff mailling list is a great idea. I have tried to look at the zaphfc driver to see what have changed since bristuff 0.2 but the changes seems to be on libpri. Is your driver based on zaphfc or is this a new driver? Zaphfc with the patch from http://zaphfc.florz.dyndns.org/ work great but when the channel go to sleep, it is unable to wake it up. Qozap can do that, but not zaphfc. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Olivier MONNET Altiva Solutions +33 476525611 Fax: +33 476525612 http://www.altiva.fr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable root shell from CLI
On Mon, Feb 12, 2007 at 10:36:51AM +0100, jeremij jerome wrote: Hi, I configured Asterisk to run as asterisk user, but I see that a user can anyway get a root sheet using !command in CLI. I understood that it's something related to safe_asterisk and TTY console, but modifying the script safe_asterisk I wasn't able to disable this root access. Can someone help me? Why exactly do you need it? That '!' is only an escape to the shell of the user who runs asterisk -r. It is not an actual command sent over the socket. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pmp_l1_check=no with zaphfc (Bristuff)
On Mon, Feb 12, 2007 at 09:15:01AM +0100, Olivier wrote: There is currently a reimplementation of zaphfc (vzaphfc). Perhaps it is more robust. Which package include such reimplementation of zaphfc ? Thanks Currently the only public repository for it is the Debian package :-( svn co svn://svn.debian.org/svn/pkg-voip/zaptel/trunk/vzaphfc -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable root shell from CLI
Try changing the shell for the asterisk user to /bin/false. This should disallow anything passed through the ! command since it runs the command via the shell for the asterisk user. jeremij jerome wrote: Hi, I configured Asterisk to run as asterisk user, but I see that a user can anyway get a root sheet using !command in CLI. I understood that it's something related to safe_asterisk and TTY console, but modifying the script safe_asterisk I wasn't able to disable this root access. Can someone help me? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
So here's my questions then. If APIC routes the IRQs to 1-15 for real world usecan you safely have two devices on, say, 14? APIC will assign one to maybe 23 and one to 20. But are they really both on 15 with a potential for conflict? The conflict only happens if your OS is not APIC aware or buggy hardware. In fact 15, is usually used for the secondary IDE port. The reason APIC exists is to support SMP and the plethora of new devices that are present on any modern motherboard. On my nforce motherboard with IO-APIC, lscpi -vb will show lots of devices using IRQ 15. But, I've never seen IRQ misses on any one of them. The same goes for our production systems running Pentium D or Xeon 51x0. I ment are they both on 14, not 15. (Sorry not feeling good the last few days and kinda working in a cloud). Ok, so in theory, even though the BIOS is saying You guys are on IRQ 6 or You guys are on IRQ 13 as long as lscpi -v and cat /proc/interrupts shows the devices not sharing and IO-APIC I should be ok? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Resque Calls from someone who is already speaking
Imagine that situation: User 100 is speaking with user 150. I am the 105 user. In my phone (105) I press *98100 (for example) and now the user 100 is empty and user 150 is speaking with me. Someone knows what I must to do??? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 and 3.3v pci
On 2/12/07, Paradise Dove [EMAIL PROTECTED] wrote: my card has just fxo modules and is put in a 3.3v slot. when running modprobe wctdm24xxp it waits for ever and dmesg shows Resetting the modules what could be the problem? when i put this card in another system with 5v slot it works fine. I would call Digium's tech support. They open in 20 minutes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan checkup
Thanks all for your input. Based on the comments given I guess I could replace the X100p for a TDM22B and then connect the Fax machine to the TDM22B and in the dial plan hae exten = fax,1,Dial(SIP/40) (being the TDM port) and then it will ring the fax machine and pass the call in from the PSTN line through the TDM card to the fax machine? Right, this is possible also or use an ATA if not a TDM card ?/ Thanks All Barry Barry Fawthrop wrote: Hi All Curious will this work Std. PSTN line ---x-- X100p | -- Fax Machine Using a standard home phone pstn line with a splitter connecting a fax machine and X100 Asterisk Box Incoming Line: Can I have in the dial Plan [incoming] exten = s,1,Wait(1) exten = s,2,IfFax continue to ring, so that the Fax Machine gets it exten = s,3,Answer exten = s,4,Playback(Message) exten = s,5,Dial(SIP/1000SIP/2000SIP/3000) exten = s,6,Hangup() exten = fax,1,Wait(30) exten = fax,2,Wait(10) exten = fax,3,Hangup() I'm wanting the line to ring, If it is a fax coming in then Asterisk leaves the line alone and lets the fax machine handle the call. If it is a call then Asterisk answers, plays a greeting and rings the IP phones? Question is how does asterisk detect the call without answering? I'm not wanting Asterisk to handle the call if it is a fax if possible??? I look forward to your input, Thank You Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems Asterisk with Digium TDM400 card = he don't see the disconnect
Hi i have a big problems with my asterisk .. i use a Digium TDM400P for connect a analog line. And not all time (i don't know why) he don't see the end of the call and anyone can call me (occuped) For that's work, i am disconnect the phone cable and it's good anyone have a idea ? bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bad audio quality on SIP
Hi guys, I have the following configuration: 10 SIP softphones -- Asterisk -- PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually only one SIP softphone user (but each time someone else) would get complaints like that ... others seem to work okay. What could be wrong? Thanx, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Quintum gateways
I have been handed three Quintum tenor AX gateways which I am suppose to configure for use with our soon to be deployed Asterisk 1.4 system. Through some mix-up we only have hardware support even though the boxes are brand new. We are working on getting software support. I would like to begin the configuration process but I do not want to use the GUI. Does someone have a concise list of CLI commands for these boxes? Thanks,Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad audio quality on SIP
If the PSTN side is only complaining about conversations with a single phone on the SIP side, look at the SIP phone. Check the settings for that SIP phone/PC (VAD disabled, NIC settings, runaway processes). Do PSTN callers here choppiness from the SIP phone caller? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Monday, February 12, 2007 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bad audio quality on SIP Hi guys, I have the following configuration: 10 SIP softphones -- Asterisk -- PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually only one SIP softphone user (but each time someone else) would get complaints like that ... others seem to work okay. What could be wrong? Thanx, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fxotune on TDM24XXE card
Is it still needed to run fxotune on the TDM24XXE cards with hardware echo cancellation? Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad audio quality on SIP
Well, the PSTN side is complaining about a random phone on the SIP side. Yes, they do hear choppiness. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: Monday, February 12, 2007 2:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bad audio quality on SIP If the PSTN side is only complaining about conversations with a single phone on the SIP side, look at the SIP phone. Check the settings for that SIP phone/PC (VAD disabled, NIC settings, runaway processes). Do PSTN callers here choppiness from the SIP phone caller? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Monday, February 12, 2007 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bad audio quality on SIP Hi guys, I have the following configuration: 10 SIP softphones -- Asterisk -- PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually only one SIP softphone user (but each time someone else) would get complaints like that ... others seem to work okay. What could be wrong? Thanx, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad audio quality on SIP
If it's a random phone on the SIP side, we have to look further upstream. While jitterbuffers may help, in my opinion they mask a problem. What type of connection do you have to the internet? Have you done tracert's to your voip provider? What do they look like? When you say that you do QoS - how? What device and settings/app helper? MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Monday, February 12, 2007 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bad audio quality on SIP Well, the PSTN side is complaining about a random phone on the SIP side. Yes, they do hear choppiness. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: Monday, February 12, 2007 2:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bad audio quality on SIP If the PSTN side is only complaining about conversations with a single phone on the SIP side, look at the SIP phone. Check the settings for that SIP phone/PC (VAD disabled, NIC settings, runaway processes). Do PSTN callers here choppiness from the SIP phone caller? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Monday, February 12, 2007 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bad audio quality on SIP Hi guys, I have the following configuration: 10 SIP softphones -- Asterisk -- PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually only one SIP softphone user (but each time someone else) would get complaints like that ... others seem to work okay. What could be wrong? Thanx, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 and 3.3v pci
Really? It's 9:23am EST and they aren't open yet. I would call Digium's tech support. They open in 20 minutes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] phpagi - Event On Hangup
Do you know if it is possible to handle some events with phpagi? For example: On hangup (doesn't care if by caller or by asterisk) do something Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Got SIP response 482 Loop Detected
On 2/12/07, Mohamed Farid [EMAIL PROTECTED] wrote: I have a Cisco Call Manager - and need to use the IVR Feature from Asterisk. My extension is 400 and I am calling 558 on Asterisk In my extension.conf I have these lines : exten = 558,1,Answer exten = 558,2,Playback(message.wav) exten = 558,3,Dial(SIP/[EMAIL PROTECTED]) When I call 558 I heared the message then Asterisk tries to call 439 on CallManager but with this error : -- Called [EMAIL PROTECTED] -- Got SIP response 482 Loop Detected back from CallManager -- Now forwarding SIP/CallManager-097b3dc0 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/CallManager-1781) == Everyone is busy/congested at this time (1:0/0/1) How can I overcome this ... Mohamed Farid ,,, * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This e-mail (including attachments) is classified as Mediterranean Smart Cards Company confidential and proprietary information The recipient hereby is committed to hold in strict confidence the contents of this (e-mail, document, and information) and not to disclose to any third party without the prior written consent of Mediterranean Smart Cards Company. Recipient will be held liable for any unauthorized disclosure. It is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please notify the sender by return e-mail and delete the message in its entirety, including any attachments * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan checkup
On Mon, 12 Feb 2007, Barry Fawthrop wrote: Thanks all for your input. Based on the comments given I guess I could replace the X100p for a TDM22B and then connect the Fax machine to the TDM22B and in the dial plan hae exten = fax,1,Dial(SIP/40) (being the TDM port) and then it will ring the fax machine and pass the call in from the PSTN line through the TDM card to the fax machine? It might work better if you use Dial(Zap/4) where 4 is the port on the TDM card... Also make sure the fax machine answers immediately and goes into fax reception mode immediately. Right, this is possible also or use an ATA if not a TDM card ?/ Possibly. However faxing is very intolerant of packet loss, jitter, phase of the moon and so on. Remember you are carrying an encoded analogue signal in digital form, so make sure you do not use a lossy codec (ie. G711 only) and while you or I may be quite tolerant of the odd click, and packet drop, an analogue modem signal (Which is what a fax communication is) is not tolerant at all. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Auto Answer (Paging)
From what I read on the voip-info page, they did not define how to have the phone muted when a paging call would go out. I already have the paging (without mute) working using those same headers from that site. But they don't cover the issue I'm trying to solve with my Polycom Soundpoint 501s and Asterisk. I need to dial an extension, have it auto pickup the phone (which i have working), but when it picks up, it should be muted (not working). Rob Shane Spencer wrote: I hate to say this, but voip-info.org has a few different methods of handling this already defined. If you are 'intercomming' to several styles of SIP based phones, you have but to only configure the phone to accept those types of calls and add a SIP header pre Dial(). Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phpagi - Event On Hangup
Usually you should use the manager interface for that. On 2/12/07, nik600 [EMAIL PROTECTED] wrote: Do you know if it is possible to handle some events with phpagi? For example: On hangup (doesn't care if by caller or by asterisk) do something Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1 card recommendation
I'm going to need to build a few Asterisk boxes that have dual and quad T1 interfaces. I knew Digium made T1 interface cards and on this list I heard about Sangoma so I did a quick search and found the hardware page at voip-info.org which lists several manufactures I didn't know about. All that leads to this question: I'll be using T1s in the USA. What experiences have you all had with different cards and what seems to work best? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable root shell from CLI
You have people administering your asterisk server who you wouldn't trust with access to the machine? EEEK. On 2/12/07, jeremij jerome [EMAIL PROTECTED] wrote: Hi, I configured Asterisk to run as asterisk user, but I see that a user can anyway get a root sheet using !command in CLI. I understood that it's something related to safe_asterisk and TTY console, but modifying the script safe_asterisk I wasn't able to disable this root access. Can someone help me? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] T1 card recommendation
I recommend you to use Sangoma A102D or A104D. Regards, Radu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Monday, February 12, 2007 5:20 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] T1 card recommendation I'm going to need to build a few Asterisk boxes that have dual and quad T1 interfaces. I knew Digium made T1 interface cards and on this list I heard about Sangoma so I did a quick search and found the hardware page at voip-info.org which lists several manufactures I didn't know about. All that leads to this question: I'll be using T1s in the USA. What experiences have you all had with different cards and what seems to work best? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Auto Answer (Paging)
On 2/12/07, Rob Schall [EMAIL PROTECTED] wrote: From what I read on the voip-info page, they did not define how to have the phone muted when a paging call would go out. I already have the paging (without mute) working using those same headers from that site. But they don't cover the issue I'm trying to solve with my Polycom Soundpoint 501s and Asterisk. I need to dial an extension, have it auto pickup the phone (which i have working), but when it picks up, it should be muted (not working). AFAIK The phone does not need to be muted as this is handled by app_page. Even if the recipient of the call speaks, the audio will be ignored as it is in a listen-only meetme conference. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Witch kernel version may i use to run well asterisk
hello, i ve a lot of problème with zaptel 1.4 when i tried to complile it under debian kernel 2.4; so i need to compile a new kernel version 2.6.x but id don't witch kernel version is stable van you help me please? :) Younss AZ KASTERISK.COM skype: younssiga ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 card recommendation
On 2/12/07, Radu Padure [EMAIL PROTECTED] wrote: I recommend you to use Sangoma A102D or A104D. I agree, though if you are on a budget, the A101 + software echo cancellation is pretty functional these days. Cheers, Steve. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
Well an upgrade to 1.2.17 now results in blips in the audio, instead of it dropping. Guess it's time to go to SuperMicro. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI question
I'm working on writing some test IVR code in AGI. I can't get my FXO port to detect a hang-up, but I'm going to deploying this using Digital cards so I decided to just skip that problem for now. However this leaves me with a question. How does AGI detect a hang-up if everything is operating normally? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Auto Answer (Paging)
Steve, I posed a similar question to Shane, but maybe you'll know as well.. I was able to get app_page to work. So when I call... **8050, it auto answers and the callee is muted. However, what if that person wants to answer the page and pickup to talk. They are already muted. Can you unmute if you are the callee? Rob Steve Davies wrote: On 2/12/07, Rob Schall [EMAIL PROTECTED] wrote: From what I read on the voip-info page, they did not define how to have the phone muted when a paging call would go out. I already have the paging (without mute) working using those same headers from that site. But they don't cover the issue I'm trying to solve with my Polycom Soundpoint 501s and Asterisk. I need to dial an extension, have it auto pickup the phone (which i have working), but when it picks up, it should be muted (not working). AFAIK The phone does not need to be muted as this is handled by app_page. Even if the recipient of the call speaks, the audio will be ignored as it is in a listen-only meetme conference. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] AsterikNow vs Trixbox
I whole heartedly agree. Trixbox/FreePBX are much more mature and feature rich. AsteriskNOW has greater future potential because of it's tight integration and no need for MySQL/Apache. However, it's not there yet so if I was to implement something today I would go with FreePBX. Not Trixbox but that is another discussion for another thread. -Original Message- From: Rob Hillis [mailto:[EMAIL PROTECTED] Sent: Monday, February 12, 2007 12:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AsterikNow vs Trixbox Smartass... :) Trixbox works off FreePBX which, while not as tightly integrated into Asterisk, is currently far more mature and easy to use. Note the use of the word currently. :) I wouldn't be too surprised if FreePBX made the move to the AsteriskNow framework. Removes the big ugly dependency for a web server. Peter Bowyer wrote: Trixbox is easier to spell. Apparently. On 11/02/07, [EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Comments? People's opinions ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parking via ## still broken
Has anybody noted that under the latest release of Asterisk (1.2.15), using the Features.conf for parking (I have ## setup) that it works fine from SIP to SIP, but is sort of broken when trying it via IAX? When parking via IAX with ##, you hear 'transfer' Enter 700 for the parking extension Hear 90 Then immediately hear the audio file from the (i)nvalid extension. If you wait, it then jumps into my Audio Prompt recording menu. The call can retrieved and subsequent parking of the same call work fine via ##. Any ideas? Log below: -- IAX2/192.168.102.15:4569-5 is ringing -- IAX2/192.168.102.15:4569-5 stopped sounds -- IAX2/192.168.102.15:4569-5 answered SIP/4191-0826b200 -- Started music on hold, class 'video', on IAX2/192.168.102.15:4569-5 -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on IAX2/192.168.102.15:4569-5 -- Started music on hold, class 'video', on IAX2/192.168.102.15:4569-5 == Parked IAX2/192.168.102.15:4569-5 on 90. Will timeout back to extension [] s, 1 in 120 seconds -- Added extension '90' priority 1 to parkedcalls -- Playing 'digits/9' (language 'en') -- Playing 'digits/0' (language 'en') -- Executing NoOp(SIP/4191-0826b200, ANSWER) in new stack -- Executing Goto(SIP/4191-0826b200, s-ANSWER|1) in new stack -- Goto (sip,s-ANSWER,1) -- Sent into invalid extension 's-ANSWER' in context 'sip' on SIP/4191-0826b200 -- Executing Playback(SIP/4191-0826b200, local/sorry-invalid-choice) in new stack -- Playing 'local/sorry-invalid-choice' (language 'en') Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Witch kernel version may i use to run well asterisk
2007/2/12, younss azzayani [EMAIL PROTECTED]: hello, i ve a lot of problème with zaptel 1.4 when i tried to complile it under debian kernel 2.4; so i need to compile a new kernel version 2.6.x but id don't witch kernel version is stable van you help me please? :) Younss AZ KASTERISK.COM skype: younssiga ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Witch kernel version may i use to run well asterisk
On Mon, Feb 12, 2007 at 03:48:31PM +, younss azzayani wrote: hello, i ve a lot of problème with zaptel 1.4 when i tried to complile it under debian kernel 2.4; I didn't ;-) so i need to compile a new kernel version 2.6.x but id don't witch kernel version is stable van you help me please? :) For stanters: apt-get install kernel-image-2.6-686 (or -k7, or -686-smp or -k7-smp , depending on your CPU) While not the latest and greatest, it still should work better than the 2.4. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trixbox vs. Custom install
Hello, I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom step by step installation on a distribution of his/her choice? Thanks Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Auto Answer (Paging)
On 2/12/07, Rob Schall [EMAIL PROTECTED] wrote: Steve, I posed a similar question to Shane, but maybe you'll know as well.. I was able to get app_page to work. So when I call... **8050, it auto answers and the callee is muted. However, what if that person wants to answer the page and pickup to talk. They are already muted. Can you unmute if you are the callee? :) Interesting question - I believe that this would require a modification to app_meetme to allow a called-user to request to talk if they are started muted. I certainly don't see such a feature documented at the moment. You may have some luck if you create a custom feature in features.conf that executes the MeetMe UnMute command if a certain key sequence if pressed. Not sure how that would work though, just dumping random thoughts really. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe whould be a good starting place for further reading and ideas. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Witch kernel version may i use to run well asterisk
Thank you, it's very easy than what i tink :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: agi script as member in queue
i've found a solutions working like this: 1 - I set up a queue that accepts caller even if it is empty. 2 - I set up an extension that dials an Agi script 3 - Each 5 seconds i run a cron job that: - if the Agi script is busy (i parse the Action : Status) to detect that : DO NOTHING - if the Agi script isn't working: transfer the caller with the maximun waiting time to the extension What do you think about that? thanks nik On 1/22/07, nik600 [EMAIL PROTECTED] wrote: Hi i want to put an AGI script in a queue, to serve once at time the callers. Example: Queue (8 callers waiting) Agi script / IVR (serving the caller) can i do that? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues and LOCAL for members
Am Friday 02 February 2007 23:48 schrieb BJ Weschke: On 2/2/07, Thomas Winter [EMAIL PROTECTED] wrote: Hi, I have an queue stored in relatime and defined members called through LOCAL/ I found out that if I call the members through the LOCAL think the queue statistics is not updated. Any idea, or isnt possible to call members with LOCAL channel. There's been some efforts to have Local channels as viable queue members. I'm not quite sure that I understand your issue. Can you post some more details possibly in a bug on bugs.digium.com ? Thanks, I found out LOCAL is working. I have been confused because an change of queue-members and an reload has reset the queue statistic. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Trixbox vs. Custom install
Hi, I'd recommend if you need quick and easy setup - use [EMAIL PROTECTED] or Trixbox or something like this, and if you need customized setup and want to understand system in detail - use your favorite distribution and setup * from sources. I'm prefer Slackware for any * installation, but your coise on your own. -- Sincerely, Elman Efendiyev PROTECH INC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefano Corsi Sent: Monday, 12 February, 2007 18:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Trixbox vs. Custom install Hello, I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom step by step installation on a distribution of his/her choice? Thanks Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sendmail problem
Hi We have a SER + asterisk server, on the same computer. after starting sendmail service , the ser will be confused. we need sendmail to send voicemails . best Mani Never Miss an Email Stay connected with Yahoo! Mail on your mobile. Get started! http://mobile.yahoo.com/services?promote=mail ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [asterisk-users] After upgrade to 1.4 transfers don't workproperly
Sorry if this is a repeat but I didn't receive a copy of it so I'm not sure it
actually posted.
The below worked for normal transfers. Now here is another situation. When we
try to transfer a call directly to voicemail it plays the voicemail message but
we can't transfer the call. The only way I could get it to work was to do a
conference and then drop out of that conference.
My dial plan for direct dialing is:
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
When this is attempted the following message shows up on the CLI of Asterisk:
[Feb 9 11:52:03] WARNING[5054]: chan_sip.c:12310 handle_response: Notify
answer on an owned channel?
Can anyone tell me what this means and what I can do to fix this?
Thanks
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
Sent: Wednesday, February 07, 2007 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] After upgrade to 1.4 transfers don't workproperly
On Wed, 2007-02-07 at 14:12 -0600, Savoy, Kevin - Williston, ND wrote:
I have discovered an issue on my system after upgrading from 1.2.13 to
1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone.
I have confirmed this on multiple phones. When the called person
answers and tries to transfer the call to another extension, the call
successfully transfers, however the person answering the transfer
cannot hear the person that called in, the caller. My dial command
simply is
I had exactly the same problem when upgrading to 1.4 and I solved by
making sure canreinvite=no is in sip.conf for every phone.
--
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
___
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Re: [asterisk-users] Re: Asterisk Faxing Support
On Feb 8, 2007, at 6:55 AM, Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Asterisk 1.2 has no support of t.38 whatsoever, the call will drop before t.38 is ever utilised, not even pass-thru. 1.4 Adds support for T.38 pass through only and no other sort of faxing, the endpoint must support T.38 and you must send your call to a T.38 gateway and you must not use NAT anywhere in your network and you must enable re-invites which could cause CDRs not to reflect the true details of the call. Asterisk/Digium also has no interest in any further interest in expanding T.38 or faxing support in Asterisk. Steve Underwood and the other fine persons that have helped to develop the software DSPs and other stuff required for FoIP support also have no interest in writing any further faxing support for Asterisk (RxFax, TxFax + the newest span_dsp wont even compile, much less work under Asterisk any more) probably because they know it will never be included into the Asterisk code. Someone please tell me this isn't truth. Of course this isn't true. We never, ever, deny good patches. What reason would we not be interested in having fax support in asterisk? We just don't maintain or own those patches, so we are limited to what we can do with them. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad audio quality on SIP
It's all in the local LAN network - client computers (with SIP softphones) are connected and registered at Asterisk SIP proxy via 100 MB connection each. The QoS is enabled under TCP/IP protocol in LAN connection in Windows (cause SIP softphones are running in Windows environment), and tos in sip.conf is set to 0x18. Unfortunately I don't have access to switch to tell you how it's set up there, but the network technicians said it is enabled. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: Monday, February 12, 2007 3:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bad audio quality on SIP If it's a random phone on the SIP side, we have to look further upstream. While jitterbuffers may help, in my opinion they mask a problem. What type of connection do you have to the internet? Have you done tracert's to your voip provider? What do they look like? When you say that you do QoS - how? What device and settings/app helper? MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Monday, February 12, 2007 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bad audio quality on SIP Well, the PSTN side is complaining about a random phone on the SIP side. Yes, they do hear choppiness. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: Monday, February 12, 2007 2:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bad audio quality on SIP If the PSTN side is only complaining about conversations with a single phone on the SIP side, look at the SIP phone. Check the settings for that SIP phone/PC (VAD disabled, NIC settings, runaway processes). Do PSTN callers here choppiness from the SIP phone caller? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Monday, February 12, 2007 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bad audio quality on SIP Hi guys, I have the following configuration: 10 SIP softphones -- Asterisk -- PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually only one SIP softphone user (but each time someone else) would get complaints like that ... others seem to work okay. What could be wrong? Thanx, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune on TDM24XXE card
It's advisable to run it on any TDM FXO interface. Even with hardware echo cancellation, fxotune makes it easier for the echo canceler to do its job. Matthew Fredrickson On Feb 12, 2007, at 7:56 AM, Jerry Geis wrote: Is it still needed to run fxotune on the TDM24XXE cards with hardware echo cancellation? Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel install...
I am having trouble getting Asterisk to compile the zaptel stuff. Here are the specifics: Linux Kernel 2.5.9-42.0.8.EL Asterisk 1.4.0 I compiled libpri, zaptel, asterisk and asterisk-addons (in that order). This is a fresh install of CentOS. Following the CentOS install, I did yum -y update until there were no updates left. Here is my src directory: drwxr-xr-x 24 root root 4096 Feb 9 23:25 asterisk drwxr-xr-x 9 root root 4096 Feb 9 23:28 asterisk-addons drwxr-xr-x 3 1000 1000 4096 Dec 6 2005 asterisk-sounds drwxr-xr-x 6 root root 4096 Feb 6 17:56 kernels drwxr-xr-x 2 root root 4096 Feb 9 23:19 libpri lrwxrwxrwx 1 root root38 Feb 9 23:22 linux-2.6 - /usr/src/kernels/2.6.9-42.0.8.EL-i686/ drwxr-xr-x 7 root root 4096 Feb 6 10:43 redhat drwxr-xr-x 10 root root 12288 Feb 9 23:25 zaptel [EMAIL PROTECTED] src]# cd /usr/lib/asterisk/modules/ [EMAIL PROTECTED] modules]# ls -l *zap* -rwxr-xr-x 1 root root 119069 Feb 9 23:26 app_zapateller.so That's the only thing there (with zap, that is). The zaptel compiled and installed ok, as I can run the zttool or ztcfg to see the cards being recognized and configured. What am I missing? -- Butch Evans Network Engineering and Security Consulting 573-276-2879 http://www.butchevans.com/ My calendar: http://tinyurl.com/y24ad6 Training Partners: http://tinyurl.com/smfkf Mikrotik Certified Consultant http://www.mikrotik.com/consultants.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox vs. Custom install
Stefano Corsi wrote: Hello, I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom step by step installation on a distribution of his/her choice? I started by trying out [EMAIL PROTECTED] I found that learning Asterisk internals was a bit more challenging trying to read and understand the [EMAIL PROTECTED] scripts. Eventually, I ended up writing a Windows GUI of my own to help learn Asterisk. The nice things about GUI's in my opinion is that routine chores such as setting up extensions, dialing extensions, hunt groups, etc. are less likely to contain scripting bugs or typos. The downside from what I gather with many GUI's is that the friendly abstraction that insulates you from the nuts and bolts of scripting and configuration also makes it difficult to customize the dialplan in some cases. Of course, you should take this with a grain of salt since I tried [EMAIL PROTECTED] (now TrixBox) for a total of 2 weeks before gutting it. Now, I just use my own GUI for everything from graphical setup to scripting. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with semaphores - SOLVED
Mitch Thompson wrote:
I'm looking for some help from any Asterisk heavy who might be doing
something similar to what I'm trying to do...
Background:
I work for a research lab, testing telephony products and tools.
Historically, we used Ameritec Crescendos and Fortissimos to act as
load generators and call sinks when testing equipment. However, the
equipment we are testing gets more and more complex, and the scripted
scenarios the Ameritecs give have become a limiting factor for
testing. Therefore, Asterisk was chosen as a possible solution (we're
a cheap lab).
I've been learning Asterisk as I go, but I've learned a lot. Here's
the basic scenario:
We are using an Asterisk (AAH 2.8, specifically) to sink calls. I do
this by taking the ${EXTEN} and breaking it down by sections until I
get to the last 4 digits (i.e., 2105551212). Once I get to the
4-digit extension, I am trying to set a flag, or semaphore, to do
Busy/Idle testing. Here is my extensions_custom.conf fragment:
[SATX_555_Extensions]
exten = 1212,1,System(cat /tmp/{orig_num}) ; ${orig_num} is set at
the beginning of [from-trunk-custom] to the full dialed digits in
${EXTEN}, before I break it down.
exten = 1212,n,Busy(); if the file exists, someone else has already
called this number, return busy
exten = 1212,102,System(echo ${UNIQUEID} /tmp/${orig_num}) ;
basically, create a file in /tmp whose name is the full number from
the beginning. In this case, the full
; filename would be /tmp/2105551212. I don't really care about the
contents, though.
exten = 1212,103, Goto(Idle,1) ; from here, we jump to a new
extension called Idle, where we do a Random to decide whether to
simulate no one home (ring no answer) or
; we send ring for
about 10 seconds, then Answer() and play some .wav files, then
hangup. The last thing we do in either case is to delete
; the
/tmp/${orig_num} file.
The above code works very well at low call volumes. However, I'm
running into race conditions at high call volumes where several calls
are getting through the test in priority 1 before the file is created
in priority 102 (n+101).
I've tried to implement semaphores by using both local and global
variables, but it doesn't seem to work.
My ultimate question: Is anyone doing something similar, and what did
you do to implement the busy/idle.
I appreciate any help anyone can offer.
Mitch Thompson
I wanted to follow-up with the solution I came up with, thanks to
excellent feedback from this group, and Trevor Peirce in particular.
Our fix action was a blending of my solution (above) and Trevor's
suggestion to stop using System() calls and use the DB directive. We
ended up with something like this:
exten = 1212,1,GotoIf(${DB_EXISTS(busy/${orig_num})}?busy:idle)
exten = 1212,n(busy),Busy()
exten = 1212,n(idle),Set(DB(busy/${orig_num})=${CALLERID(num)})
exten = 1212,n,Macro(disposition,$(orig_num}) ; Call a macro, pass
along the original CdPN, and do something.
In the disposition macro, we have this:
[macro-disposition]
exten = s,1,Set(orig_num=${arg1})
exten = s,n,Random(25:s-rna,1)
exten = s,n,Ringing()
exten = s,n,Wait(6)
exten = s,n,Answer()
exten = s,n,Wait(1)
exten = s,n,Playback(lots-o-monkeys)
exten = s,n,Hangup()
exten = s-rna,1,Ringing()
exten = s-rna,n,Wait()
exten = s-rna,n,Hangup
exten = h,1,DBDel(busy/${orig_num})
The above gives us a very simplistic coin-toss as to whether to answer
the phone or not.
With the above dialplan, we have successfully dialed one number with
eight (8) T-1 PRIs worth of calls simultaneously and only have 1 answer
(the remaining 183 calls were shown as Busy on the Fortissimo). This
was repeatable over a 2 hour period. I'm sure we will have calls slip
through, but for now we are satisfied.
Trevor, thanks again. I learned about the DB application that day.
___
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Re: [asterisk-users] pmp_l1_check=no with zaphfc (Bristuff)
I have installed the vzaphfc and this what I was looking for. I will do some more testing and I will post my results here. Thank you for your help Le 12 févr. 07 à 11:56, Tzafrir Cohen a écrit : On Mon, Feb 12, 2007 at 09:15:01AM +0100, Olivier wrote: There is currently a reimplementation of zaphfc (vzaphfc). Perhaps it is more robust. Which package include such reimplementation of zaphfc ? Thanks Currently the only public repository for it is the Debian package :-( svn co svn://svn.debian.org/svn/pkg-voip/zaptel/trunk/vzaphfc -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Olivier MONNET Altiva Solutions +33 476525611 Fax: +33 476525612 http://www.altiva.fr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNOW Migration
I currently have a customer that a previous employee setup with Gentoo\Asterisk. I'm looking to migrate to AsteriskNOW. They have a custom menu, which I would assume is easily replicable in AsteriskNOW. The only other thing I can think of is the sound bites for the menus. Does anyone have any advise or migration recommendations for this move? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad audio quality on SIP
The problem will be on the outside of your Asterisk PBX. In other words, your asterisk server's external NIC (or if just one NIC), connection to your firewall/router, to you voip provider. You need to run tracert's from your Asterisk box to your voip provider. QoS on the windows clients is useless (and doesn't matter in this case). QoS is often misunderstood. Without knowledge of the protocols you are running, your network admin could not have setup the router/firewall to shape traffic properly. Prioritizing based on QoS bits offers minimal benefits. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Monday, February 12, 2007 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bad audio quality on SIP It's all in the local LAN network - client computers (with SIP softphones) are connected and registered at Asterisk SIP proxy via 100 MB connection each. The QoS is enabled under TCP/IP protocol in LAN connection in Windows (cause SIP softphones are running in Windows environment), and tos in sip.conf is set to 0x18. Unfortunately I don't have access to switch to tell you how it's set up there, but the network technicians said it is enabled. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: Monday, February 12, 2007 3:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bad audio quality on SIP If it's a random phone on the SIP side, we have to look further upstream. While jitterbuffers may help, in my opinion they mask a problem. What type of connection do you have to the internet? Have you done tracert's to your voip provider? What do they look like? When you say that you do QoS - how? What device and settings/app helper? MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Monday, February 12, 2007 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bad audio quality on SIP Well, the PSTN side is complaining about a random phone on the SIP side. Yes, they do hear choppiness. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: Monday, February 12, 2007 2:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bad audio quality on SIP If the PSTN side is only complaining about conversations with a single phone on the SIP side, look at the SIP phone. Check the settings for that SIP phone/PC (VAD disabled, NIC settings, runaway processes). Do PSTN callers here choppiness from the SIP phone caller? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Monday, February 12, 2007 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bad audio quality on SIP Hi guys, I have the following configuration: 10 SIP softphones -- Asterisk -- PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually only one SIP softphone user (but each time someone else) would get complaints like that ... others seem to work okay. What could be wrong? Thanx, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] i m looking for a document that allow me to install well an asterisk server
re Hi, I m looking for a good document that allow me to install zaptel libpri asterisk without errors, i ve a TDM400 TE110P, so please can you help me Kind Regards Younss AZZAYANI KASTERISK.COM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: i m looking for a document that allow me to install well an asterisk server
i forgot to tell you that i m using a debian 2.6.8 kernel version ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Trixbox vs. Custom install
Of course, you should take this with a grain of salt since I tried [EMAIL PROTECTED] (now TrixBox) for a total of 2 weeks before gutting it. Now, I just use my own GUI for everything from graphical setup to scripting. There is nothing wrong with starting out with Trixbox. I still use it because I like the Linux distro (CentOS) and I like the fact that it sets up lots of stuff that I don't have to bother with. I used Trixbox to learn a lot about how to use Asterisk, then I went back and did a clean install on a separate machine to learn about setting up and installing Asterisk. For me, having a working system first, playing with it, breaking it, etc. was very useful because it gave me perspective when setting up a system from scratch. Now I actually have two systems to play with: one Trixbox and one scratch * install. (I get the best of both worlds, but I have nothing in production just yet. I'll decide later which way to go once I'm doing playing with my two 'sandboxes.') Bottom line is this: you need to start somewhere. Would you rather start by using a working system or by building from the ground up? Neither way is perfect for everyone. If you have the luxury of doing both then I can highly recommend it - each method has taught me valuable lessons that the other method didn't. HTH... -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel install...
cOMO REALIZAS LA INSTALCION, DEBES CONSEGUIR LOS TAR.GZ DE ZAPTEL y luego compilar asi: #yum install kernel kernel-devel gcc Con esto actualizas el kernel con sus fuentes y el gcc #vi /etc/grub.conf seleccionas el nuevo kernel como por default # reboot reinicias para que se ejecute el nuevo kernel luego de eso instalas lo necesario y cuando te toque el zaptel descomprimes la version de zaptel que tengas zaptel.x.y.z.tar.gz en /usr/src y luego ejecutas #make linux26 #make install y listo 2007/2/12, Butch Evans [EMAIL PROTECTED]: I am having trouble getting Asterisk to compile the zaptel stuff. Here are the specifics: Linux Kernel 2.5.9-42.0.8.EL Asterisk 1.4.0 I compiled libpri, zaptel, asterisk and asterisk-addons (in that order). This is a fresh install of CentOS. Following the CentOS install, I did yum -y update until there were no updates left. Here is my src directory: drwxr-xr-x 24 root root 4096 Feb 9 23:25 asterisk drwxr-xr-x 9 root root 4096 Feb 9 23:28 asterisk-addons drwxr-xr-x 3 1000 1000 4096 Dec 6 2005 asterisk-sounds drwxr-xr-x 6 root root 4096 Feb 6 17:56 kernels drwxr-xr-x 2 root root 4096 Feb 9 23:19 libpri lrwxrwxrwx 1 root root38 Feb 9 23:22 linux-2.6 - /usr/src/kernels/2.6.9-42.0.8.EL-i686/ drwxr-xr-x 7 root root 4096 Feb 6 10:43 redhat drwxr-xr-x 10 root root 12288 Feb 9 23:25 zaptel [EMAIL PROTECTED] src]# cd /usr/lib/asterisk/modules/ [EMAIL PROTECTED] modules]# ls -l *zap* -rwxr-xr-x 1 root root 119069 Feb 9 23:26 app_zapateller.so That's the only thing there (with zap, that is). The zaptel compiled and installed ok, as I can run the zttool or ztcfg to see the cards being recognized and configured. What am I missing? -- Butch Evans Network Engineering and Security Consulting 573-276-2879 http://www.butchevans.com/ My calendar: http://tinyurl.com/y24ad6 Training Partners: http://tinyurl.com/smfkf Mikrotik Certified Consultant http://www.mikrotik.com/consultants.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Carlos Gómez R. Quito, Ecuador Home: (593)-2-2591218 Mobil: (593)-9-2060171 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] AsteriskNOW Migration
I would suggest you grab the menu from the .conf file and paste it into the new setup. (After even a little asterisk experience, they should be able to get away from the gui). The sound files could be copied as well. I'm guessing from your question that you/your client may not having Linux experience...if you've ever worked in DOS you can figure this one out! ( Or if you're not comfortable in Linux, get a consultant to help you migrate). MD Disclaimer: Yes I work for an Asterisk consulting company - so this message can appear self serving.. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Monday, February 12, 2007 1:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AsteriskNOW Migration I currently have a customer that a previous employee setup with Gentoo\Asterisk. I'm looking to migrate to AsteriskNOW. They have a custom menu, which I would assume is easily replicable in AsteriskNOW. The only other thing I can think of is the sound bites for the menus. Does anyone have any advise or migration recommendations for this move? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Trixbox vs. Custom install
Hi Stefano, I am a proponent of the step-by-step installation on a complete linux distribution. Like someone said in another posting, the GUIs are nice, but isolate you from the .conf files to the point where customization can be a bit tricky. However, Trixbox w/ FreePBX and A2Billing works out of the box with very little patching or configuration needed. If A2Billing is all he/she anticipates needing to do, FreePBX is a mature and stable and he/she will probably be happy with it. FreePBX is a bit of a chore to install and configure without Trixbox, if you don't have a solid understanding of dependencies, linux security, apache and MySQL. Same with A2Billing. Edward Halman (718) 705-7451 [EMAIL PROTECTED] -- Message: 12 Date: Mon, 12 Feb 2007 17:42:15 +0100 From: Stefano Corsi [EMAIL PROTECTED] Subject: [asterisk-users] Trixbox vs. Custom install To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 45CB1BDE00381960@ (added by [EMAIL PROTECTED]) Content-Type: text/plain; charset=us-ascii; format=flowed Hello, I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom step by step installation on a distribution of his/her choice? Thanks Stefano -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: i m looking for a document that allow me to install well an asterisk server
Have you tried this link? http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_debi an.html Edward Halman (718) 705-7451 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, February 12, 2007 2:00 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 31, Issue 49 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. i m looking for a document that allow me to install well an asterisk server (younss azzayani) 2. Re: i m looking for a document that allow me to install well an asterisk server (younss azzayani) -- Message: 1 Date: Mon, 12 Feb 2007 18:43:49 + From: younss azzayani [EMAIL PROTECTED] Subject: [asterisk-users] i m looking for a document that allow me to install well an asterisk server To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed re Hi, I m looking for a good document that allow me to install zaptel libpri asterisk without errors, i ve a TDM400 TE110P, so please can you help me Kind Regards Younss AZZAYANI KASTERISK.COM -- Message: 2 Date: Mon, 12 Feb 2007 18:46:05 + From: younss azzayani [EMAIL PROTECTED] Subject: [asterisk-users] Re: i m looking for a document that allow me to install well an asterisk server To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed i forgot to tell you that i m using a debian 2.6.8 kernel version -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 31, Issue 49 ** ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox vs. Custom install
Michael Collins wrote: Of course, you should take this with a grain of salt since I tried [EMAIL PROTECTED] (now TrixBox) for a total of 2 weeks before gutting it. Now, I just use my own GUI for everything from graphical setup to scripting. There is nothing wrong with starting out with Trixbox. I still use it because I like the Linux distro (CentOS) and I like the fact that it sets up lots of stuff that I don't have to bother with. I used Trixbox to learn a lot about how to use Asterisk, then I went back and did a clean install on a separate machine to learn about setting up and installing Asterisk. For me, having a working system first, playing with it, breaking it, etc. was very useful because it gave me perspective when setting up a system from scratch. Now I actually have two systems to play with: one Trixbox and one scratch * install. (I get the best of both worlds, but I have nothing in production just yet. I'll decide later which way to go once I'm doing playing with my two 'sandboxes.') Bottom line is this: you need to start somewhere. Would you rather start by using a working system or by building from the ground up? Neither way is perfect for everyone. If you have the luxury of doing both then I can highly recommend it - each method has taught me valuable lessons that the other method didn't. [EMAIL PROTECTED] was very nice so I can only assume that TrixBox is great. An associate of mine (whom got me interested in Asterisk) sells TrixBox systems like they're going out of style. I was merely relaying my own experience and agree with you that no way is ever perfect and the more choices we have, the better. Personally, I tend to learn new concepts better if I build a solid foundation of the basics first so starting with a bare asterisk install ended up working better for me. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: i m looking for a document that allow me to install well an asterisk server
no, i'll try bouththe linke that you gave me the link that Cohen had given to me thank you very mutch Cohen Edward 2007/2/12, Edward Halman [EMAIL PROTECTED]: Have you tried this link? http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_debi an.html Edward Halman (718) 705-7451 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, February 12, 2007 2:00 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 31, Issue 49 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. i m looking for a document that allow me to install well an asterisk server (younss azzayani) 2. Re: i m looking for a document that allow me to install well an asterisk server (younss azzayani) -- Message: 1 Date: Mon, 12 Feb 2007 18:43:49 + From: younss azzayani [EMAIL PROTECTED] Subject: [asterisk-users] i m looking for a document that allow me to install well an asterisk server To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed re Hi, I m looking for a good document that allow me to install zaptel libpri asterisk without errors, i ve a TDM400 TE110P, so please can you help me Kind Regards Younss AZZAYANI KASTERISK.COM -- Message: 2 Date: Mon, 12 Feb 2007 18:46:05 + From: younss azzayani [EMAIL PROTECTED] Subject: [asterisk-users] Re: i m looking for a document that allow me to install well an asterisk server To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed i forgot to tell you that i m using a debian 2.6.8 kernel version -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 31, Issue 49 ** ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI question
in your dialplan: [context] ... h,1,AGI(...) David Ruggles [EMAIL PROTECTED] wrote: I'm working on writing some test IVR code in AGI. I can't get my FXO port to detect a hang-up, but I'm going to deploying this using Digital cards so I decided to just skip that problem for now. However this leaves me with a question. How does AGI detect a hang-up if everything is operating normally? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Have a burning question? Go to Yahoo! Answers and get answers from real people who know.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Asterisk/callerid with pay as you go VOIP providers
I am curious how others handle call out VOIP and callerid. I have found numerous providers that are cheap and seem to have good voice quality but offer no provisions for callerid. I find it almost useless to use call out when the receiving party gets some bogus callerid number that has no relation to my call. I understand the big thing is spoofing callerid but are there any companies that offer a means of qualifying callerid so it works right? Like it or not callerid is used heavily and without a proper return ID many callee's don't answer and if they tried to return the call they get no where. Seems like a big problem to me. Very aggrevating. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI question
That's right, but i think that you should use: exten = h,1,DEADAGI( ) because in h extension the channel is considered as 'dead channel' , Regards, -- J. Espinal Slackware-es.com chester c young wrote: in your dialplan: [context] ... h,1,AGI(...) */David Ruggles [EMAIL PROTECTED]/* wrote: I'm working on writing some test IVR code in AGI. I can't get my FXO port to detect a hang-up, but I'm going to deploying this using Digital cards so I decided to just skip that problem for now. However this leaves me with a question. How does AGI detect a hang-up if everything is operating normally? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have a burning question? Go to Yahoo! Answers http://answers.yahoo.com/;_ylc=X3oDMTFvbGNhMGE3BF9TAzM5NjU0NTEwOARfcwMzOTY1NDUxMDMEc2VjA21haWxfdGFnbGluZQRzbGsDbWFpbF90YWcx and get answers from real people who know. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] colors in the console
I'm wondering if anyone else has experienced this. Up until a few days ago, when accessing the CLI from my terminal program (Private Shell), the output was in color. I haven't upgraded, rebuilt, or to my knowledge, changed anything in Asterisk that would change this. My terminal settings were the same as well. I have two computers that I access the CLI regularly on, and neither show color anymore. When I disconnect, Private Shell shows the disconnect in red, just like before. This tells me that Private Shell is still doing color. What controls the color coding in the CLI? I found something in the source about it, but again, since it has been recompiled, this should not have changed. Is there a config file somewhere that I'm too blind to find? Thanks! -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNOW Migration
On Mon, Feb 12, 2007 at 02:06:31PM -0500, Michelle Dupuis wrote: I would suggest you grab the menu from the .conf file and paste it into the new setup. (After even a little asterisk experience, they should be able to get away from the gui). Note that confiugration of AsteriskNow rewrites extensions.conf, and thus an #include of an external file will not work as planned. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNOW Migration
That is incorrect. AsteriskNOW (actually, the AsteriskGUI) edits files in place, leaving any old information in them. This allows you to fully customize your users and dialplan without interfering with the GUI's operation. Aaron Daniel Community Relations Specialist [EMAIL PROTECTED] (256) 428-6010 - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, February 12, 2007 2:51:21 PM GMT-0600 US/Central Subject: Re: [asterisk-users] AsteriskNOW Migration On Mon, Feb 12, 2007 at 02:06:31PM -0500, Michelle Dupuis wrote: I would suggest you grab the menu from the .conf file and paste it into the new setup. (After even a little asterisk experience, they should be able to get away from the gui). Note that confiugration of AsteriskNow rewrites extensions.conf, and thus an #include of an external file will not work as planned. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colors in the console
Lacy Moore - Aspendora wrote: I'm wondering if anyone else has experienced this. Up until a few days ago, when accessing the CLI from my terminal program (Private Shell), the output was in color. I haven't upgraded, rebuilt, or to my knowledge, changed anything in Asterisk that would change this. My terminal settings were the same as well. I have two computers that I access the CLI regularly on, and neither show color anymore. When I disconnect, Private Shell shows the disconnect in red, just like before. This tells me that Private Shell is still doing color. What controls the color coding in the CLI? I found something in the source about it, but again, since it has been recompiled, this should not have changed. Is there a config file somewhere that I'm too blind to find? Thanks! -- Lacy Moore Somewhere I wish I wasn't I believe that only the CLI console provides color: e.g. asterisk -c. Connecting to an already-running asterisk process will not provide color: e.g. asterisk -r. Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colors in the console
Incorrect. I connect to asterisk -r all the time and get colour. Is it possible your terminal emulation has changed in Private Shell? Is it VT100, or ANSI? On 2/12/07, Earle Clubb [EMAIL PROTECTED] wrote: Lacy Moore - Aspendora wrote: I'm wondering if anyone else has experienced this. Up until a few days ago, when accessing the CLI from my terminal program (Private Shell), the output was in color. I haven't upgraded, rebuilt, or to my knowledge, changed anything in Asterisk that would change this. My terminal settings were the same as well. I have two computers that I access the CLI regularly on, and neither show color anymore. When I disconnect, Private Shell shows the disconnect in red, just like before. This tells me that Private Shell is still doing color. What controls the color coding in the CLI? I found something in the source about it, but again, since it has been recompiled, this should not have changed. Is there a config file somewhere that I'm too blind to find? Thanks! -- Lacy Moore Somewhere I wish I wasn't I believe that only the CLI console provides color: e.g. asterisk -c. Connecting to an already-running asterisk process will not provide color: e.g. asterisk -r. Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Small CDR Billing Program
Hi Guys I am just looking around for a small billing program but can't really find what I am looking for. It needs to bill straight off the CDR. It should grab all the CDR records from the asteriskcdrdb mysql database then have a rates table to that it calculate a bill from. Is there any open source packages or commercial packages that will account for billing say only 5 extensions? Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colors in the console
I have seen this when I have restarted the server from the asterisk CLI and not a service asterisk restart command. I'm not sure as to why, but I always assumed it had to do with the safe_asterisk file. On 2/12/07, Earle Clubb [EMAIL PROTECTED] wrote: Lacy Moore - Aspendora wrote: I'm wondering if anyone else has experienced this. Up until a few days ago, when accessing the CLI from my terminal program (Private Shell), the output was in color. I haven't upgraded, rebuilt, or to my knowledge, changed anything in Asterisk that would change this. My terminal settings were the same as well. I have two computers that I access the CLI regularly on, and neither show color anymore. When I disconnect, Private Shell shows the disconnect in red, just like before. This tells me that Private Shell is still doing color. What controls the color coding in the CLI? I found something in the source about it, but again, since it has been recompiled, this should not have changed. Is there a config file somewhere that I'm too blind to find? Thanks! -- Lacy Moore Somewhere I wish I wasn't I believe that only the CLI console provides color: e.g. asterisk -c. Connecting to an already-running asterisk process will not provide color: e.g. asterisk -r. Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colors in the console
On Mon, Feb 12, 2007 at 04:20:44PM -0500, Matt wrote: Incorrect. I connect to asterisk -r all the time and get colour. Is it possible your terminal emulation has changed in Private Shell? Is it VT100, or ANSI? Asterisk seems to disable the colors when you don't start it in a terminal. Funny. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsterikNow vs Trixbox
Apart from the feature/maturity issue there is a far more important (IMHO) difference in the architectural approach of the two GUIs. FreePBX assumes it owns the world and completely re-writes its configuration files once changes are made through the GUI. While it makes token efforts to enable one to edit custom files which are includes into the configuration files -- the ability to make significant manual changes to the dialplan while still using the GUI is limited and challenging. AsteriskNOW's GUI (which still in its early days) takes a much less dictatorial approach and will, to a much greater degree, cooperate with manual configuration changes. Because of the difficulty of manually tweaking configurations generated by FreePBX, I have had to remove FreePBX from every customer who has started down this road (since they inevitably ask for something that cannot be configured through FreePBX and that requires manual tweaks). While I have no yet deployed a 1.4 system with the Asterisk GUI at a customer site, my early experiments in the lab suggest that Digium's approach will be much more cooperative and flexible. g. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best phone for easy provisioning
Aastra are a delight -- no need for a compiler (like the Grandstream and Linksys phones) -- and extremely well documented configuration files. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best phone for easy provisioning
On Mon, 12 Feb 2007, George Pajari wrote: Aastra are a delight -- no need for a compiler (like the Grandstream and Linksys phones) -- and extremely well documented configuration files. While I agree that Grandstream phones might not be the easiest things in the world, I did find this recently: http://www.pkts.ca/gsutil.shtml and initial results look very promising... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colors in the console
On 2/12/07, Bruce Reeves [EMAIL PROTECTED] wrote: I have seen this when I have restarted the server from the asterisk CLI and not a service asterisk restart command. I'm not sure as to why, but I always assumed it had to do with the safe_asterisk file. Bruce, that may have been it. I just exited from the CLI, did a service asterisk stop and then start, and went back into asterisk -r and have color again. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colors in the console
On Mon, Feb 12, 2007 at 11:27:48PM +0200, Tzafrir Cohen wrote: On Mon, Feb 12, 2007 at 04:20:44PM -0500, Matt wrote: Incorrect. I connect to asterisk -r all the time and get colour. Is it possible your terminal emulation has changed in Private Shell? Is it VT100, or ANSI? Asterisk seems to disable the colors when you don't start it in a terminal. Funny. If it bothers anybody, try the patch in http://bugs.digium.com/view.php?id=9048 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SayUnixTime Alternate Path?
Does anyone know how I could get the SayUnixTime application to say files from a different sound directory? It looks like it uses the language as a base to determine where to play sound files from. I need to override that. Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colors in the console
On Mon, Feb 12, 2007 at 04:30:54PM -0600, Lacy Moore - Aspendora wrote: On 2/12/07, Bruce Reeves [EMAIL PROTECTED] wrote: I have seen this when I have restarted the server from the asterisk CLI and not a service asterisk restart command. I'm not sure as to why, but I always assumed it had to do with the safe_asterisk file. Bruce, that may have been it. I just exited from the CLI, did a service asterisk stop and then start, and went back into asterisk -r and have color again. You seem to start asterisk with safe_asterisk. That script starts asterisk on a console of its own. Maybe it wa done to allow the use of colors. If you want a plain 'asterisk' to run with colors, try the patch in http://bugs.digium.com/view.php?id=9048 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Asterisk Faxing Support
Matthew, ok, but is realy possible change the dsp code in the Asterisk? Guys around The OpenPBX change the dsp to Steve's spandsp and has the native T38 support now. Tomas Urbanek -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Monday, February 12, 2007 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Asterisk Faxing Support On Feb 8, 2007, at 6:55 AM, Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Asterisk 1.2 has no support of t.38 whatsoever, the call will drop before t.38 is ever utilised, not even pass-thru. 1.4 Adds support for T.38 pass through only and no other sort of faxing, the endpoint must support T.38 and you must send your call to a T.38 gateway and you must not use NAT anywhere in your network and you must enable re-invites which could cause CDRs not to reflect the true details of the call. Asterisk/Digium also has no interest in any further interest in expanding T.38 or faxing support in Asterisk. Steve Underwood and the other fine persons that have helped to develop the software DSPs and other stuff required for FoIP support also have no interest in writing any further faxing support for Asterisk (RxFax, TxFax + the newest span_dsp wont even compile, much less work under Asterisk any more) probably because they know it will never be included into the Asterisk code. Someone please tell me this isn't truth. Of course this isn't true. We never, ever, deny good patches. What reason would we not be interested in having fax support in asterisk? We just don't maintain or own those patches, so we are limited to what we can do with them. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel install...
On Mon, 2007-02-12 at 11:59 -0600, Butch Evans wrote: I am having trouble getting Asterisk to compile the zaptel stuff. Here are the specifics: Linux Kernel 2.5.9-42.0.8.EL Asterisk 1.4.0 I compiled libpri, zaptel, asterisk and asterisk-addons (in that order). This is a fresh install of CentOS. Following the CentOS install, I did yum -y update until there were no updates left. Here is my src directory: drwxr-xr-x 24 root root 4096 Feb 9 23:25 asterisk drwxr-xr-x 9 root root 4096 Feb 9 23:28 asterisk-addons drwxr-xr-x 3 1000 1000 4096 Dec 6 2005 asterisk-sounds drwxr-xr-x 6 root root 4096 Feb 6 17:56 kernels drwxr-xr-x 2 root root 4096 Feb 9 23:19 libpri lrwxrwxrwx 1 root root38 Feb 9 23:22 linux-2.6 - /usr/src/kernels/2.6.9-42.0.8.EL-i686/ drwxr-xr-x 7 root root 4096 Feb 6 10:43 redhat drwxr-xr-x 10 root root 12288 Feb 9 23:25 zaptel [EMAIL PROTECTED] src]# cd /usr/lib/asterisk/modules/ [EMAIL PROTECTED] modules]# ls -l *zap* -rwxr-xr-x 1 root root 119069 Feb 9 23:26 app_zapateller.so The zaptel modules are installed into the kernel modules directory and not where you are looking. For your kernel look into: /lib/modules/2.6.9-42.0.8.EL/misc Also remember to do a make config in the zaptel source to install the init files so zaptel will load automatically when you boot. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
Well an upgrade to 1.2.17 now results in blips in the audio, instead of it dropping. Guess it's time to go to SuperMicro. 1.2.17 ? (1.2.13 zaptel?) i have supermicro mobo(P8SCT) and have same problem with shared interrupts bash#lspci -bv | grep -i IRQ 5 --before-context=2 00:02.0 VGA compatible controller: Intel Corporation E7221 Integrated Graphics Controller (rev 05) (prog-if 00 [VGA]) Subsystem: Super Micro Computer Inc: Unknown device 7480 Flags: bus master, fast devsel, latency 0, IRQ 5 -- 00:1d.3 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #4 (rev 03) (prog-if 00 [UHCI]) Subsystem: Super Micro Computer Inc: Unknown device 7480 Flags: bus master, medium devsel, latency 0, IRQ 5 -- 02:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 795e:0001 Flags: bus master, medium devsel, latency 32, IRQ 5 -- 03:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Super Micro Computer Inc: Unknown device 02c6 Flags: bus master, fast devsel, latency 0, IRQ 5 can you someone explain what's mean by (zaptel 1.2.13 changelog) 2007-01-23 21:28 + [r1936] Matt Frederickson [EMAIL PROTECTED] * wcte11xp.c, wct1xxp.c, wctdm.c, wctdm24xxp.c: Make sure we don't clear the interrupt before we might have received it in shared interrupt line scenarios. --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [asterisk-users] Small CDR Billing Program
Hi Mark, Take a look at the YakaVOIP solution from http://www.yakasoftware.com/ http://www.yakasoftware.com. Probably suits your requirements. Greetz, Roland. _ Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von MBIT Technologies Gesendet: 12 February 2007 22:23 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Small CDR Billing Program Hi Guys I am just looking around for a small billing program but can't really find what I am looking for. It needs to bill straight off the CDR. It should grab all the CDR records from the asteriskcdrdb mysql database then have a rates table to that it calculate a bill from. Is there any open source packages or commercial packages that will account for billing say only 5 extensions? Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel install...
On Mon, Feb 12, 2007 at 11:59:55AM -0600, Butch Evans wrote: I am having trouble getting Asterisk to compile the zaptel stuff. Here are the specifics: Linux Kernel 2.5.9-42.0.8.EL Asterisk 1.4.0 I compiled libpri, zaptel, asterisk and asterisk-addons (in that order). This is a fresh install of CentOS. Following the CentOS install, I did yum -y update until there were no updates left. Here is my src directory: drwxr-xr-x 24 root root 4096 Feb 9 23:25 asterisk drwxr-xr-x 9 root root 4096 Feb 9 23:28 asterisk-addons drwxr-xr-x 3 1000 1000 4096 Dec 6 2005 asterisk-sounds drwxr-xr-x 6 root root 4096 Feb 6 17:56 kernels drwxr-xr-x 2 root root 4096 Feb 9 23:19 libpri lrwxrwxrwx 1 root root38 Feb 9 23:22 linux-2.6 - /usr/src/kernels/2.6.9-42.0.8.EL-i686/ drwxr-xr-x 7 root root 4096 Feb 6 10:43 redhat drwxr-xr-x 10 root root 12288 Feb 9 23:25 zaptel [EMAIL PROTECTED] src]# cd /usr/lib/asterisk/modules/ [EMAIL PROTECTED] modules]# ls -l *zap* -rwxr-xr-x 1 root root 119069 Feb 9 23:26 app_zapateller.so That's the only thing there (with zap, that is). The zaptel compiled and installed ok, as I can run the zttool or ztcfg to see the cards being recognized and configured. What am I missing? Zaptel is installed. The problem is with the configuration of Asterisk. What version of zaptel have you installed? To check if the configure script detected zaptel: grep ZAPTEL= build_tools/menuselect-deps To check if it is actively disables by the menuselect: grep -w chan_zap menuselect.makeopts In the latter: if you see 'chan_zap' in the line MENUSELECT_CHANNEL, it is disabled (which is a bit counter-intiutive). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Asterisk with Digium TDM400 card = he don't see the disconnect
Noc Phibee wrote: Hi i have a big problems with my asterisk .. i use a Digium TDM400P for connect a analog line. And not all time (i don't know why) he don't see the end of the call and anyone can call me (occuped) For that's work, i am disconnect the phone cable and it's good anyone have a idea ? This is a common problem. You have to answer a few questions if you want to fix it: Does your telephone company provide any kind of analog disconnect supervision? This is sometimes called Calling Party Control. Some companies will configure this if you ask them. If the answer is no, you will have to use tricks to make Asterisk consistently detect disconnection. The tricks don't work well. Where are you located? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got SIP response 482 Loop Detected
Mohamed Farid wrote: On 2/12/07, Mohamed Farid [EMAIL PROTECTED] wrote: I have a Cisco Call Manager - and need to use the IVR Feature from Asterisk. My extension is 400 and I am calling 558 on Asterisk In my extension.conf I have these lines : exten = 558,1,Answer exten = 558,2,Playback(message.wav) exten = 558,3,Dial(SIP/[EMAIL PROTECTED]) When I call 558 I heared the message then Asterisk tries to call 439 on CallManager but with this error : -- Called [EMAIL PROTECTED] -- Got SIP response 482 Loop Detected back from CallManager -- Now forwarding SIP/CallManager-097b3dc0 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/CallManager-1781) == Everyone is busy/congested at this time (1:0/0/1) How can I overcome this ... First start a fresh thread rather than replying to a different one. In other words: Don't pick a message, hit reply, and then rewrite the subject line. Instead - Click New Message, write a fresh subject line, and put the asterisk-users list address in the To: field. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad audio quality on SIP
Michelle Dupuis wrote: The problem will be on the outside of your Asterisk PBX. In other words, your asterisk server's external NIC (or if just one NIC), connection to your firewall/router, to you voip provider. You need to run tracert's from your Asterisk box to your voip provider. QoS on the windows clients is useless (and doesn't matter in this case). QoS is often misunderstood. Without knowledge of the protocols you are running, your network admin could not have setup the router/firewall to shape traffic properly. Prioritizing based on QoS bits offers minimal benefits. This has been my experience also. The only place where I have seen QoS provide any real advantage is in enterprise environments where one administration team controls all the network hardware and VOIP is only used to the network boundary. QoS is a fantastic sales tool for Cisco, though! If you are having trouble with chops, blips, and other call quality problems, you have a connectivity or configuration issue that QoS will not help. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox vs. Custom install
Lee Jenkins wrote: Stefano Corsi wrote: Hello, I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom step by step installation on a distribution of his/her choice? I started by trying out [EMAIL PROTECTED] I found that learning Asterisk internals was a bit more challenging trying to read and understand the [EMAIL PROTECTED] scripts. Eventually, I ended up writing a Windows GUI of my own to help learn Asterisk. The nice things about GUI's in my opinion is that routine chores such as setting up extensions, dialing extensions, hunt groups, etc. are less likely to contain scripting bugs or typos. The downside from what I gather with many GUI's is that the friendly abstraction that insulates you from the nuts and bolts of scripting and configuration also makes it difficult to customize the dialplan in some cases. It also makes troubleshooting problems a handful-and-a-half. And woe is you if you need kernel customizations to make your hardware work. I would say this -- if all you're ever going to use is VOIP trunks, by all means use Trixbox. It's great for that. But if you're using any kind of PSTN hardware (TDM cards, Sangoma) just stick with straight Asterisk. I've just had my second go at Trixbox (version 2.0 now) and after wasting a bunch of time with hardware problems, I'm going to replace it with a generic install. Here's another reason to seriously consider generic: the userbase is larger, AND they're more likely to know what they're talking about when a problem does arise. Trixbox attracts a lot of amateurs who are themselves new to IP telephony; that's why they choose it. Of course, you should take this with a grain of salt since I tried [EMAIL PROTECTED] (now TrixBox) for a total of 2 weeks before gutting it. There is a good reason people don't stick with it for long. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox vs. Custom install
Michael Collins wrote: Of course, you should take this with a grain of salt since I tried [EMAIL PROTECTED] (now TrixBox) for a total of 2 weeks before gutting it. Now, I just use my own GUI for everything from graphical setup to scripting. There is nothing wrong with starting out with Trixbox. I still use it because I like the Linux distro (CentOS) and I like the fact that it sets up lots of stuff that I don't have to bother with. I used Trixbox to learn a lot about how to use Asterisk, then I went back and did a clean install on a separate machine to learn about setting up and installing Asterisk. For me, having a working system first, playing with it, breaking it, etc. was very useful because it gave me perspective when setting up a system from scratch. Now I actually have two systems to play with: one Trixbox and one scratch * install. (I get the best of both worlds, but I have nothing in production just yet. I'll decide later which way to go once I'm doing playing with my two 'sandboxes.') This is a fair statement, unless you can't get Trixbox working in the first place. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
Er... no you don't :)My problem and everyone elses with Dell is that Dell builds the Mobos to share the PCI IRQs with the NIC cards. I've got some SuperMicro MoBos running VoIP and they DO share exactly like you showed. There is nothing wrong with sharing your VGA (Video) with your PSTN card. 99.% of the time that video is just going to sit there doing nothing. And when it is in use, it isn't much.. unlike a network card that, well, with a VoIP server, kinda gets hit hard. I wouldn't worry about your IRQ sharing... that is exactly the kind of sharing that is ok. However, sharing real-time NIC with real-time PSTN interface == BAD. On 2/12/07, marek cervenka [EMAIL PROTECTED] wrote: Well an upgrade to 1.2.17 now results in blips in the audio, instead of it dropping. Guess it's time to go to SuperMicro. 1.2.17 ? (1.2.13 zaptel?) i have supermicro mobo(P8SCT) and have same problem with shared interrupts bash#lspci -bv | grep -i IRQ 5 --before-context=2 00:02.0 VGA compatible controller: Intel Corporation E7221 Integrated Graphics Controller (rev 05) (prog-if 00 [VGA]) Subsystem: Super Micro Computer Inc: Unknown device 7480 Flags: bus master, fast devsel, latency 0, IRQ 5 -- 00:1d.3 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #4 (rev 03) (prog-if 00 [UHCI]) Subsystem: Super Micro Computer Inc: Unknown device 7480 Flags: bus master, medium devsel, latency 0, IRQ 5 -- 02:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 795e:0001 Flags: bus master, medium devsel, latency 32, IRQ 5 -- 03:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Super Micro Computer Inc: Unknown device 02c6 Flags: bus master, fast devsel, latency 0, IRQ 5 can you someone explain what's mean by (zaptel 1.2.13 changelog) 2007-01-23 21:28 + [r1936] Matt Frederickson [EMAIL PROTECTED] * wcte11xp.c, wct1xxp.c, wctdm.c, wctdm24xxp.c: Make sure we don't clear the interrupt before we might have received it in shared interrupt line scenarios. --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
ARG. I need to stop posting to this list until I recover from this cold. I see now that you are sharing with your NIC card. That *IS* bad. Go into the BIOS. Turn off USB, and Parallel, Serial... basically everything you don't need. Now, save and reboot. Go back into BIOS. See if you can, now, set the Digium card to be on a different IRQ. On 2/12/07, Matt [EMAIL PROTECTED] wrote: Er... no you don't :)My problem and everyone elses with Dell is that Dell builds the Mobos to share the PCI IRQs with the NIC cards. I've got some SuperMicro MoBos running VoIP and they DO share exactly like you showed. There is nothing wrong with sharing your VGA (Video) with your PSTN card. 99.% of the time that video is just going to sit there doing nothing. And when it is in use, it isn't much.. unlike a network card that, well, with a VoIP server, kinda gets hit hard. I wouldn't worry about your IRQ sharing... that is exactly the kind of sharing that is ok. However, sharing real-time NIC with real-time PSTN interface == BAD. On 2/12/07, marek cervenka [EMAIL PROTECTED] wrote: Well an upgrade to 1.2.17 now results in blips in the audio, instead of it dropping. Guess it's time to go to SuperMicro. 1.2.17 ? (1.2.13 zaptel?) i have supermicro mobo(P8SCT) and have same problem with shared interrupts bash#lspci -bv | grep -i IRQ 5 --before-context=2 00:02.0 VGA compatible controller: Intel Corporation E7221 Integrated Graphics Controller (rev 05) (prog-if 00 [VGA]) Subsystem: Super Micro Computer Inc: Unknown device 7480 Flags: bus master, fast devsel, latency 0, IRQ 5 -- 00:1d.3 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #4 (rev 03) (prog-if 00 [UHCI]) Subsystem: Super Micro Computer Inc: Unknown device 7480 Flags: bus master, medium devsel, latency 0, IRQ 5 -- 02:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 795e:0001 Flags: bus master, medium devsel, latency 32, IRQ 5 -- 03:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Super Micro Computer Inc: Unknown device 02c6 Flags: bus master, fast devsel, latency 0, IRQ 5 can you someone explain what's mean by (zaptel 1.2.13 changelog) 2007-01-23 21:28 + [r1936] Matt Frederickson [EMAIL PROTECTED] * wcte11xp.c, wct1xxp.c, wctdm.c, wctdm24xxp.c: Make sure we don't clear the interrupt before we might have received it in shared interrupt line scenarios. --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Trixbox vs. Custom install
IMHO, If you don't know enough about Linux/Asterisk/FreePBX to be able to set it up yourself you should not be doing it for a Production install in a business environment. NOTE: Production install in a business environment does NOT include setting it up in your house with extensions for the kids and wife! That is what Trixbox is for. -Original Message- From: Edward Halman [mailto:[EMAIL PROTECTED] Sent: Monday, February 12, 2007 12:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Trixbox vs. Custom install Hi Stefano, I am a proponent of the step-by-step installation on a complete linux distribution. Like someone said in another posting, the GUIs are nice, but isolate you from the .conf files to the point where customization can be a bit tricky. However, Trixbox w/ FreePBX and A2Billing works out of the box with very little patching or configuration needed. If A2Billing is all he/she anticipates needing to do, FreePBX is a mature and stable and he/she will probably be happy with it. FreePBX is a bit of a chore to install and configure without Trixbox, if you don't have a solid understanding of dependencies, linux security, apache and MySQL. Same with A2Billing. Edward Halman (718) 705-7451 [EMAIL PROTECTED] -- Message: 12 Date: Mon, 12 Feb 2007 17:42:15 +0100 From: Stefano Corsi [EMAIL PROTECTED] Subject: [asterisk-users] Trixbox vs. Custom install To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 45CB1BDE00381960@ (added by [EMAIL PROTECTED]) Content-Type: text/plain; charset=us-ascii; format=flowed Hello, I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom step by step installation on a distribution of his/her choice? Thanks Stefano -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Trixbox vs. Custom install
If you are installing in a business environment I think being able to use both and know the benefits of both are pretty essential. FreePBX is a great tool and should be used to its potential because it has some great features. It can also lessen the time it takes to do an install. Some have said that FreePBX is a control freak and wants control of everything. This is not true. If you know how to manipulate freepbx properly then you shouldn't have any trouble adding in custom features as well as using the main features built into FreePBX. Trixbox can have some problems with hardware because it is a prebuilt RPM. If you can't install asterisk and other components from source you really shouldn't be doing production installs. Being able to patch the source with different features is also very essential. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym Sent: Tuesday, 13 February 2007 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Re: Trixbox vs. Custom install IMHO, If you don't know enough about Linux/Asterisk/FreePBX to be able to set it up yourself you should not be doing it for a Production install in a business environment. NOTE: Production install in a business environment does NOT include setting it up in your house with extensions for the kids and wife! That is what Trixbox is for. -Original Message- From: Edward Halman [mailto:[EMAIL PROTECTED] Sent: Monday, February 12, 2007 12:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Trixbox vs. Custom install Hi Stefano, I am a proponent of the step-by-step installation on a complete linux distribution. Like someone said in another posting, the GUIs are nice, but isolate you from the .conf files to the point where customization can be a bit tricky. However, Trixbox w/ FreePBX and A2Billing works out of the box with very little patching or configuration needed. If A2Billing is all he/she anticipates needing to do, FreePBX is a mature and stable and he/she will probably be happy with it. FreePBX is a bit of a chore to install and configure without Trixbox, if you don't have a solid understanding of dependencies, linux security, apache and MySQL. Same with A2Billing. Edward Halman (718) 705-7451 [EMAIL PROTECTED] -- Message: 12 Date: Mon, 12 Feb 2007 17:42:15 +0100 From: Stefano Corsi [EMAIL PROTECTED] Subject: [asterisk-users] Trixbox vs. Custom install To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 45CB1BDE00381960@ (added by [EMAIL PROTECTED]) Content-Type: text/plain; charset=us-ascii; format=flowed Hello, I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom step by step installation on a distribution of his/her choice? Thanks Stefano -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
