Re: [Asterisk-Users] I love you!
Nah.. The only way I ever notice that someone sent me a virus is when my mailserver says beep.. And that is currently so frequent that I'm considering turning that off.. :-) Makes me think of this (The Network Auralizer): http://peep.sourceforge.net/intro.html Check out the Low load average demo mp3, you'll know what I mean.. -- Soren - Original Message - From: Thomas Gallaway [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 10, 2004 9:00 PM Subject: Re: [Asterisk-Users] I love you! [EMAIL PROTECTED] wrote: lovely, :-) Is it just me or where there allready 3 virus sent to this list today? Maybe time for denim to disallow attachments? :-) -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller-ID for alphanumeric SIP uris
My first post here, so a brief intro: I'm somewhat new to Asterisk, but have been working with SIP in depth for about 3 years. I studied Asterisk for about a year and have been experimenting with it hands-on for the past month or so. I've done 6 Asterisk installs in wildly different roles/applications, some of them test systems, others in semi-production, so I know a little bit about it. I've setup voicemail, meetme, ENUM, and other Asterisk features, and I've written some AGI scripts and done some other semi-interesting tweaks. That said, I'm curious about how others might solve the following problem. In a pure-SIP environment, if a user has an alphanumeric SIP uri, say sip:[EMAIL PROTECTED], when that user calls another SIP phone, (a real IP phone, as opposed to an ATA), via a SIP proxy, that phone can log the full URI, and 'call return' works because the SIP phone calls that URI. With Asterisk, such a call would come in with the SIP From: header (thus Caller-ID in Asterisk parlance) as something like: From: joe sip:[EMAIL PROTECTED];tag=as54f3792a In this case, Asterisk doesn't know how to return the call, nor does the SIP phone, because even if the SIP phone can dial full alphanumeric URI's with some kind of a 'call return' feature, the sip:[EMAIL PROTECTED] (where 204.16.112.70 would be the IP address of the Asterisk server), isn't a valid URI and doesn't route a call to the original SIP URI: sip:[EMAIL PROTECTED] I've thought of some tricks for handling this, and I've looked around the archives and Google searches, but haven't seen much discussion of this issue. TIA, David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SDP messages relating to rtpmap Question
SDP question if * recieves a=rtpmap:103 telephone-event/8000 it shouldn't it send out the same a=rtpmap:103 telephone-event/8000 to the other side of the connection? and not something like a=rtpmap:101 telephone-event/8000? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible TICKING sound
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ryan Courtnage wrote: | | On 11-May-04, at 8:45 AM, Ryan Courtnage wrote: | | I've fought with this problem on and off. | | Number 1 thing to check is /proc/interrupts to ensure that your card | isn't sharing an interrupt with something else. | | Number 2 is a bit of an unknown variable - my guess is either | electrical noise, or perhaps vibrations affecting your card inside | your box. I find that carefully remounting my tdm400p/x100p so that | nothing at all is touching it (no wires, no plastic, nothing - except | at the mount point) will make the problem go away the majority of the | time. If it doesn't go away, try re-mounting again. | | | Currently, I'm working with a tdm400p with 2 FXO, 2 FXS. the persistent | tick tick tick tick has come back (after 3 days tick-free operation). | | Has anyone else experienced a similar problem with ZAP channels? What | steps did you take to resolve it? Our problem ended up not being with Asterisk or Digium hardware. It was the analog cordless phone. We simply have to live with it. What happens is whenever a connection is established and the phone is off-hook, an LED on the base lights up in a blink blink . blink blink . etc. pattern. Everytime the LED lights, a pulse is sent to the phone. It's especially bad when both lines are in use, as the phone is a two-line capable device. Then you've got double the pulsing. This may have nothing to do with your problem. Just wanted to get it out there in case anyone else runs into it, too. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAoVSxuYsUrHkpYtARAuNoAJ9niwQT6VI+G+e/lHqFdH6WYIEu/wCfb0YB X42z4g4hilKtSajzZg9bFpY= =8SLd -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SDP messages relating to rtpmap Question
Actually this has come up before we can receive on one and send on another. The other end should use what we say. One leg could be using 103 and the other leg of the call could use 101. Or that's how I understand it. Bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Glenn Dalgliesh Sent: Tuesday, May 11, 2004 5:17 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SDP messages relating to rtpmap Question SDP question if * recieves a=rtpmap:103 telephone-event/8000 it shouldn't it send out the same a=rtpmap:103 telephone-event/8000 to the other side of the connection? and not something like a=rtpmap:101 telephone-event/8000? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to record all agent calls
I want to record incoming calls that are queued when the call is connected to an agent. I added the following lines to agents.conf before the list of agents: ; Enable recording calls addressed to agents. It's turned off by default. recordagentcalls=yes ; ;The format to be used to record the calls ;wav, gsm, wav49. ; By default its wav. recordformat=gsm ; ; Insert into CDR userfield a name of the the created recording ; By default it's turned off. createlink=no ; ; The text to be added to the name of the recording. Allows forming a url link. ;urlprefix=http://host.domain/calls/ ; ; The optional directory to save the conversations in. The default is ; /var/spool/asterisk/monitor ;savecallsin=/var/calls and added to the queues.conf file: ; monitor-format = gsm|wav|wav49 monitor-format = gsm ...and then issued the reload command in the Asterisk CLI console. I even created the /var/log/asterisk/monitor directory because it did not exist. Is there something else that needs to happen to record calls between agents and callers so you can hear both sides of the conversation? Thanks in advance. --- Jeff Crews Eastern Oregon Net, Inc. La Grande Oregon Email [EMAIL PROTECTED] Voice 541-963-2625 or 800-785-7873, extension 11 personal efax 503-907-6704, standard company fax 541-962-7818 web http://home.eoni.com
RE: [Asterisk-Users] Terrible TICKING sound
The Adit channel bank we are using, and XO communications who provisioned the T1 are both showing a LOT of framing errors on our system. Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tmpm Sent: Tuesday, May 11, 2004 12:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Terrible TICKING sound Ive found this in audio apps on other boxes when the power supply is really loaded down hard. Just one more maybe for you to check. Have you blown the dust out of the P/S lately? Dirt and temp variations seem to affect it as well...found this with audio equipment at a broadcast station. (Streaming server on a Linux box) Not saying its your situation, but wont hurt to check. If it does help, a beefier power supply might help here. It cured my case. Marc At 10:45 5/11/2004, you wrote: I've fought with this problem on and off. Number 1 thing to check is /proc/interrupts to ensure that your card isn't sharing an interrupt with something else. Number 2 is a bit of an unknown variable - my guess is either electrical noise, or perhaps vibrations affecting your card inside your box. I find that carefully remounting my tdm400p/x100p so that nothing at all is touching it (no wires, no plastic, nothing - except at the mount point) will make the problem go away the majority of the time. If it doesn't go away, try re-mounting again. It's a little scary, especially when you're working with a small-form-factor machine. Ryan On 10-May-04, at 8:43 PM, Paul Mahler wrote: i'm getting a tick every second or so on all my calls. All channels are zap channels. Does anyone know how to fix this? Thanks! Paul Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.signate.com/ Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SDP messages relating to rtpmap Question
http://lists.cs.columbia.edu/pipermail/sip-implementors/2002-May/003059.html Thats this same exact issue explained a bit more. bkw -- Forwarded message -- Date: Tue, 11 May 2004 17:35:33 -0500 From: brian [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SDP messages relating to rtpmap Question Actually this has come up before we can receive on one and send on another. The other end should use what we say. One leg could be using 103 and the other leg of the call could use 101. Or that's how I understand it. Bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Glenn Dalgliesh Sent: Tuesday, May 11, 2004 5:17 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SDP messages relating to rtpmap Question SDP question if * recieves a=rtpmap:103 telephone-event/8000 it shouldn't it send out the same a=rtpmap:103 telephone-event/8000 to the other side of the connection? and not something like a=rtpmap:101 telephone-event/8000? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible TICKING sound
The Adit channel bank we are using, and XO communications who provisioned the T1 are both showing a LOT of framing errors on our system. Tell Asterisk to clock from XO's T1. How is this related to your TDM400P though? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to record all agent calls
Only works in CVS-HEAD bkw - Original Message - From: Jeff Crews To: [EMAIL PROTECTED] Sent: Tuesday, May 11, 2004 4:57 PM Subject: [Asterisk-Users] how to record all agent calls I want to record incoming calls that are queued when the call is connected to an agent.I added the following lines to agents.conf before the list of agents:; Enable recording calls addressed to agents. It's turned off by default.recordagentcalls=yes;;The format to be used to record the calls;wav, gsm, wav49.; By default its "wav".recordformat=gsm;; Insert into CDR userfield a name of the the created recording; By default it's turned off.createlink=no;; The text to be added to the name of the recording. Allows forming a url link.;urlprefix=http://host.domain/calls/;; The optional directory to save the conversations in. The default is; /var/spool/asterisk/monitor;savecallsin=/var/callsand added to the queues.conf file:; monitor-format = gsm|wav|wav49monitor-format = gsm...and then issued the reload command in the Asterisk CLI console.I even created the /var/log/asterisk/monitor directory because it did not exist.Is there something else that needs to happen to record calls between agents and callers so you can hear both sides of the conversation?Thanks in advance. ---Jeff CrewsEastern Oregon Net, Inc.La Grande OregonEmail [EMAIL PROTECTED]Voice 541-963-2625 or 800-785-7873, extension 11 personal efax 503-907-6704, standard company fax 541-962-7818 web http://home.eoni.com
RE: [Asterisk-Users] how to record all agent calls
you need to combine both sides of the conversation. This should be covered in the archives. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLCPO Box 60430Palo Alto, CA94306VoIP Systems, Training Consulting From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff CrewsSent: Tuesday, May 11, 2004 2:57 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] how to record all agent calls I want to record incoming calls that are queued when the call is connected to an agent.I added the following lines to agents.conf before the list of agents:; Enable recording calls addressed to agents. It's turned off by default.recordagentcalls=yes;;The format to be used to record the calls;wav, gsm, wav49.; By default its "wav".recordformat=gsm;; Insert into CDR userfield a name of the the created recording; By default it's turned off.createlink=no;; The text to be added to the name of the recording. Allows forming a url link.;urlprefix=http://host.domain/calls/;; The optional directory to save the conversations in. The default is; /var/spool/asterisk/monitor;savecallsin=/var/callsand added to the queues.conf file:; monitor-format = gsm|wav|wav49monitor-format = gsm...and then issued the reload command in the Asterisk CLI console.I even created the /var/log/asterisk/monitor directory because it did not exist.Is there something else that needs to happen to record calls between agents and callers so you can hear both sides of the conversation?Thanks in advance. ---Jeff CrewsEastern Oregon Net, Inc.La Grande OregonEmail [EMAIL PROTECTED]Voice 541-963-2625 or 800-785-7873, extension 11 personal efax 503-907-6704, standard company fax 541-962-7818 web http://home.eoni.com signate small logo.gif
RE: [Asterisk-Users] Terrible TICKING sound - Fixed
Well, for me it was a problem with the T1 line. XO fixed the line and the ticking sound is gone! Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, May 11, 2004 3:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Terrible TICKING sound The Adit channel bank we are using, and XO communications who provisioned the T1 are both showing a LOT of framing errors on our system. Tell Asterisk to clock from XO's T1. How is this related to your TDM400P though? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible TICKING sound
On Tue, 2004-05-11 at 17:33, Jason A. Pattie wrote: Our problem ended up not being with Asterisk or Digium hardware. It was the analog cordless phone. We simply have to live with it. What happens is whenever a connection is established and the phone is off-hook, an LED on the base lights up in a blink blink . blink blink . etc. pattern. Everytime the LED lights, a pulse is sent to the phone. It's especially bad when both lines are in use, as the phone is a two-line capable device. Then you've got double the pulsing. This may have nothing to do with your problem. Just wanted to get it out there in case anyone else runs into it, too. Sounds like your phone needs either a aux power source to power that led, or possible a little modification to clip that LED. I would make sure your cordless phone's power supply is within spec. If it is, Maybe you might want to look into one of the other comments a while back on the list about upping the power on the SLIC(?). You might be able to provide enough power to the phone to not cause trouble when it blinks the LED. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk + VoiceWorks
I have a customer who has a Comdial Phone System and uses VoiceWorks for it's voicemail. I am installing an IVR / Time Clock system utilizing asterisk. But they want to have an option to hop over to the VoiceWorks system to check voicemail... -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Posted At: Tuesday, May 11, 2004 2:52 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Asterisk + VoiceWorks Subject: RE: [Asterisk-Users] Asterisk + VoiceWorks Why on earth would you wanna do something like that? Asterisk has voicemail and you even have the src so you can add those nifty features the PHB's like to have but never use! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of AstGrp Sent: Tuesday, May 11, 2004 11:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk + VoiceWorks I have a need to interface Asterisk with a VoiceWorks voicemail system. I was wondering what kind of card would be needed either a FXO or FXS interface? Any help would be appreciated. Thanks, -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + VoiceWorks
What sort of Time Clock? For employees to punch in and out? That's a very attractive idea. Is this a custom solution or is there a plugin out there for this? atr - Original Message - From: AstGrp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 11, 2004 10:07 PM Subject: RE: [Asterisk-Users] Asterisk + VoiceWorks I have a customer who has a Comdial Phone System and uses VoiceWorks for it's voicemail. I am installing an IVR / Time Clock system utilizing asterisk. But they want to have an option to hop over to the VoiceWorks system to check voicemail... -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Posted At: Tuesday, May 11, 2004 2:52 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Asterisk + VoiceWorks Subject: RE: [Asterisk-Users] Asterisk + VoiceWorks Why on earth would you wanna do something like that? Asterisk has voicemail and you even have the src so you can add those nifty features the PHB's like to have but never use! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of AstGrp Sent: Tuesday, May 11, 2004 11:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk + VoiceWorks I have a need to interface Asterisk with a VoiceWorks voicemail system. I was wondering what kind of card would be needed either a FXO or FXS interface? Any help would be appreciated. Thanks, -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID for alphanumeric SIP uris
At 3:02 PM -0700 on 5/11/04, David Beckemeyer wrote: My first post here, so a brief intro: I'm somewhat new to Asterisk, but have been working with SIP in depth for about 3 years. I studied Asterisk for about a year and have been experimenting with it hands-on for the past month or so. I've done 6 Asterisk installs in wildly different roles/applications, some of them test systems, others in semi-production, so I know a little bit about it. I've setup voicemail, meetme, ENUM, and other Asterisk features, and I've written some AGI scripts and done some other semi-interesting tweaks. That said, I'm curious about how others might solve the following problem. In a pure-SIP environment, if a user has an alphanumeric SIP uri, say sip:[EMAIL PROTECTED], when that user calls another SIP phone, (a real IP phone, as opposed to an ATA), via a SIP proxy, that phone can log the full URI, and 'call return' works because the SIP phone calls that URI. With Asterisk, such a call would come in with the SIP From: header (thus Caller-ID in Asterisk parlance) as something like: From: joe sip:[EMAIL PROTECTED];tag=as54f3792a In this case, Asterisk doesn't know how to return the call, nor does the SIP phone, because even if the SIP phone can dial full alphanumeric URI's with some kind of a 'call return' feature, the sip:[EMAIL PROTECTED] (where 204.16.112.70 would be the IP address of the Asterisk server), isn't a valid URI and doesn't route a call to the original SIP URI: sip:[EMAIL PROTECTED] I've thought of some tricks for handling this, and I've looked around the archives and Google searches, but haven't seen much discussion of this issue. TIA, David David - You're correct. This is an unfortunate side effect of Asterisk not really being a SIP proxy. It's a PBX replacement. Now, I understand that Olle's chan_sip2 has some of this type of feature functionality built into it, and you may want to take a look at that. In the interim, there are some really awful, terrible, horrendous tricks you can do that might work around this problem. It involves snagging the SIP URI on the inbound call, pushing it into a database, assigning a pseudo-random number to that entry, and then keeping that mapping... forever. If a user hit the redial button, then the inverse would happen: your Asterisk server would dig through the database looking for the key, find the 'real' SIP URI, and re-route the call to the appropriate correct endpoint. Ugly. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * INSTRUCTIONS FOR MEMBERS OF THE COMMUNITY * Please read
Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. ** The mailing list is growing The lead programmer of Asterisk, Mark Spencer at Digium, inc, writes: The Asterisk community is growing at a remarkable pace. I know there are thousands of you out there -- in fact there are over eight *thousand* subscribers to asterisk-users alone, and almost one *thousand* registered users on the bug tracker. This means that everything anyone write to this mailing list, is sent to over 8.000 mailboxes that is already flowing over with messages. I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org project is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org Their handbook The hitchhiker's guide to Asterisk is already well worth reading. Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list. You'll find it on http://lists.digium.com, which is the address where you manage your subscription to this list as well. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. Please check the bugtracker thoroughly before posting a new bug; often, your bug or feature already exists but is simply slowly making it's way through the system. Duplicate reports slow things down for everyone, so please spend a few minutes searching first. The bug tracker is also a place where you add your contribution to Asterisk. If you have coded extra functionality, make sure you give it back to the project so it can be added to the code base. This is how Asterisk grows, free contributions and consultants that are paid to add functionality on a case by case basis. ** Remember: It's Open Source, it's voluntary Asterisk.org is a Open Source project. This means you can't request help from people, demand new functions or support. However, there are many individuals and companies out there that are offering services based on Asterisk, from VoIP service providers to consultants all over the world. Of course, this is also part of Digium's business, so you have plenty of help if your willing to pay. Digium is to be found at http://www.digium.com. Service providers and consultants are listed on the wiki, where you'll find companies all over the globe that are willing to set up your PBX and get you connected to either the PSTN or the growing telephony network on the Internet. * See http://www.voip-info.org/wiki-Asterisk%20consultants Again, welcome to the Asterisk.org Open Source PBX Project! Meet you on the IRC channel :-) /oej - PS. This message will be sent regularly. If you have any corrections or additional information that needs to be included, mail me * off list *. Thank you! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I love you!
1 solution 4 yout workstation: Linux Linux Linux Linux Linux Linux Linux Linux !!! :-) On Tue, 2004-05-11 at 05:53, John Fraizer wrote: tmpm wrote: Of course, and I suggest a firewall as well, but its NOT going to do anything for a purloined email some infected machine in Bumsquatialand sending it as you. Theres only so much you can do. If you run your own mailserver, I suggest the following to keep your braindead users from being infected by the click here or open random attachment vector: http://www.pc-tools.net/unix/renattach/ It removes/renames attachments in emails, at the server. It is very configurable and is free. Next: If you opened that attachment on your own, you're an *idiot* and got what you had coming. If you *didn't* open it on your own but, your mailreader program did, you're an *idiot* for running that mailreader and got what you had coming. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Use buttons (other than #) after call is bridged?
Hi, can i somehow use the other buttons to execute some apps, *without* hanging up the call? Something like: exten = s,1,Dial/SIP(1234)|4,5,7,9 exten = 4,1,Monitor(wav) exten = 5,1,SIPDtmfMode(inband) exten = 7,1,AGI(turnoncoffeemachine.agi) exten = 9,1,System(smbnuke boss) Regards, AA _ Watch movie trailers online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk resource consumption..
No this is not another of the What hardware do I need? posts.. :) Just wondering if anyone has calculated the memory consumprion for running asterisk.. For example, when its idle it uses U MB or RAM, uses V MB for each active Zap channel, W MB for each active SIP channel, X MB for each active IAX Channel and Y MB for each VM channel.. I often read posts where guys are throwing 1-2GB or RAM at a server and this seems extreme to me but there don't seem to be any numbers for people to use as a guide (could be placed on voip-info.org).. Unfortunately my system is to small to do any real testing of this kind but I would be happy to help somone who has a bigger more active system to get the numbers.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Use buttons (other than #) after call is bridged?
Is there some way of driving external contacts with Asterisk? I've seen something running on windows that allowed a Dpin I/O port to drive up to 15 contacts, is there someway to get asterisk to do the same? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Anderson Sent: Tuesday, 11 May 2004 7:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Use buttons (other than #) after call is bridged? Hi, can i somehow use the other buttons to execute some apps, *without* hanging up the call? Something like: exten = s,1,Dial/SIP(1234)|4,5,7,9 exten = 4,1,Monitor(wav) exten = 5,1,SIPDtmfMode(inband) exten = 7,1,AGI(turnoncoffeemachine.agi) exten = 9,1,System(smbnuke boss) Regards, AA _ Watch movie trailers online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Use buttons (other than #) after call is bridged?
That should be pretty easy if you already have the interface to the DPIN I/O port. Then it would be just a matter of some system() calls on different extensions which communicate with your DPIN I/O port program. e.x.: exten = 99910,1,system(io_prog 1 0) ;turn port 1 off exten = 99911,1,system(io_prog 1 1) ;turn port 1 on exten = 99920,1,system(io_prog 2 0) ;turn port 2 off exten = 99921,1,system(io_prog 2 1) ;turn port 2 on exten = 99930,1,system(io_prog 3 0) ;turn port 3 off exten = 99931,1,system(io_prog 3 1) ;turn port 3 on I don't know if this is the optimal solution, but I would implement it that way if is was my project. Regards, Mickey -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: 11. maj 2004 12:09 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Use buttons (other than #) after call is bridged? Is there some way of driving external contacts with Asterisk? I've seen something running on windows that allowed a Dpin I/O port to drive up to 15 contacts, is there someway to get asterisk to do the same? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Anderson Sent: Tuesday, 11 May 2004 7:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Use buttons (other than #) after call is bridged? Hi, can i somehow use the other buttons to execute some apps, *without* hanging up the call? Something like: exten = s,1,Dial/SIP(1234)|4,5,7,9 exten = 4,1,Monitor(wav) exten = 5,1,SIPDtmfMode(inband) exten = 7,1,AGI(turnoncoffeemachine.agi) exten = 9,1,System(smbnuke boss) Regards, AA _ Watch movie trailers online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line appearances
I am trying to get an understanding of how line appearances work like on the cisco 7960 phones. Is there a wiki somewhere about how this works? Also, the 7960 phones let you register more than one ext. Why would you want more than one or is this connected to line appearances? Is there a way to have phones use more than one codec, say use g.711 to talk with * and g.729 to talk with another phone? -- respectfully, Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipgate to regular phones
I could call a regular phone through sipgate. Now i can not: Failed to authenticate on INVITE to 'xyz sip:[EMAIL PROTECTED];tag=as4ddd4a6f' A call from outside to my sip-phone through sipgate is OK. Can anyone verify ? Is it a sipgate problem ? greetings nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipgate to regular phones
Failed to authenticate on INVITE to 'xyz sip:[EMAIL PROTECTED];tag=as4ddd4a6f' what says sip debug? greets Markus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk resource consumption..
Unfortunately, it is not as simple as just counting channel types. You need to factor in which compression codecs are used, as each will have a different impact on the system. Are multiple codecs used requiring transcoding, or just one? And what is Asterisk being used for? Is Asterisk monitoring the channels or is native bridging permitted? Is muisc on hold used, and if so, how many different music on hold contexts? Is Asterisk recording the channels? What about meetme conferences? Regards. WipeOut wrote: No this is not another of the What hardware do I need? posts.. :) Just wondering if anyone has calculated the memory consumprion for running asterisk.. For example, when its idle it uses U MB or RAM, uses V MB for each active Zap channel, W MB for each active SIP channel, X MB for each active IAX Channel and Y MB for each VM channel.. I often read posts where guys are throwing 1-2GB or RAM at a server and this seems extreme to me but there don't seem to be any numbers for people to use as a guide (could be placed on voip-info.org).. Unfortunately my system is to small to do any real testing of this kind but I would be happy to help somone who has a bigger more active system to get the numbers.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *** Asterisk sunday news: Read the sample configs, Luke!
On Sun, 2004-05-09 at 07:41, Mark Elkins wrote: On Sun, 2004-05-09 at 14:33, Mark Elkins wrote: On Sun, 2004-05-09 at 09:59, Olle E. Johansson wrote: * Read the config sample files! (even if you're an Asterisk guru) - For those of you that have a working installation that you keep using, this is a reminder to check into the configs/ directory of the Asterisk source tree, regardless if you downloaded a tar ball or from CVS. Good advice - so I do a CVS UPDATE... and 'say.c' is broken ... The lines that begin with say.c Sorry folks... seems like a CVS Update did break - removed the file and re-updated. fine now. However - this could bit other people too.. in which case - delete the offending file - and update again (or always use 'cvs checkout' - less efficient - but..) There should have been a warning from CVS about not being able to merge the changes to that files. Then you would have been warned, and you might have learned then that the sections that had been marked by the '' would have shown you where you needed to reconsile the changes to say.c. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipgate to regular phones
I just came across the same problem. I fixed it by changing from sipgate.net to sipgate.de in the sip.conf file. Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nicolas Sent: Tuesday, 11 May 2004 21:10 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sipgate to regular phones I could call a regular phone through sipgate. Now i can not: Failed to authenticate on INVITE to 'xyz sip:[EMAIL PROTECTED];tag=as4ddd4a6f' A call from outside to my sip-phone through sipgate is OK. Can anyone verify ? Is it a sipgate problem ? greetings nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk resource consumption..
Robert Lawrence wrote: Unfortunately, it is not as simple as just counting channel types. You need to factor in which compression codecs are used, as each will have a different impact on the system. Are multiple codecs used requiring transcoding, or just one? And what is Asterisk being used for? Is Asterisk monitoring the channels or is native bridging permitted? Is muisc on hold used, and if so, how many different music on hold contexts? Is Asterisk recording the channels? What about meetme conferences? Regards. All valid points.. I realise that it would not be a perfect calculation but if it was possible to say that in an average you need X MB for running Asterisk and then Y MB per active channel it would at least give people some method to calculate the memory requirements, this is similar to most software where they will say you need a certain amout of memory to run the app and then in server systems they tell you to add an amount per user.. For example, my system uses 83 MB of RAM (excluding buffers and cache) when idle and when I make a call to the echo test it uses about 20k.. This means that by these numbers a standard system for 100 concurrent users would need no more than 128MB of RAM minimum and 256MB would be plenty.. Obviously this is very simplistic and only based on a echo test but it does mean that the people who are throwing 2GB of memory at their asterisk servers are wasting a lot of money that could have been put into faster processors to handle more calls and services.. Thats really all I was talking about so that it helps people size their systems and would probably mean fewer What system do I need? questions.. Later.. WipeOut wrote: No this is not another of the What hardware do I need? posts.. :) Just wondering if anyone has calculated the memory consumprion for running asterisk.. For example, when its idle it uses U MB or RAM, uses V MB for each active Zap channel, W MB for each active SIP channel, X MB for each active IAX Channel and Y MB for each VM channel.. I often read posts where guys are throwing 1-2GB or RAM at a server and this seems extreme to me but there don't seem to be any numbers for people to use as a guide (could be placed on voip-info.org).. Unfortunately my system is to small to do any real testing of this kind but I would be happy to help somone who has a bigger more active system to get the numbers.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Sipgate to regular phones
Thanks Now it works. nicolas Simon Brown wrote: I just came across the same problem. I fixed it by changing from sipgate.net to sipgate.de in the sip.conf file. Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nicolas Sent: Tuesday, 11 May 2004 21:10 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sipgate to regular phones I could call a regular phone through sipgate. Now i can not: Failed to authenticate on INVITE to 'xyz sip:[EMAIL PROTECTED];tag=as4ddd4a6f' A call from outside to my sip-phone through sipgate is OK. Can anyone verify ? Is it a sipgate problem ? greetings nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI.pm wait_for_digit() not working for me!!!
Ok,
the first think to do is check the permissions on the conf-background.agi ..asterisk
needs to be able to run it ...
The code I've listed below works fine for me:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
%input = $AGI-ReadParse();
$soundpath = /var/lib/asterisk/sounds/;
$timeout = 10;
while(1)
{
$input = chr($AGI-wait_for_digit($timeout));
if ($input eq *)
{
$AGI-stream_file($soundpath/banana-phone-song);
}
if ($input eq 1)
{
exit 0;
}
}
*** REPLY SEPARATOR ***
On 11/05/2004 at 10:52 Atif wrote:
Hello everybody!!!
I really need your help guys, I am using the AGI mode in meetme
application,
and I want that AGI should wait for an input from the client/user i.e. a
digit and then proceed, but I have used that AGI function
agi-wait_for_digit(), but no usemy agi just passes, or ignores this
function,
where AGI should stop here and wait for the input
.my extension in my dialplan.
exten = 21,1,answer
exten = 21,2,meetme(21|pb)
..and here is my AGI...
#!/usr/bin/perl -w
#use strict;
$aginame=conf-background.agi;
use File::Copy cp;
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$char=0;
#while(1)
{
#$AGI-exec('WaitExten','25000');
#$char = $AGI-receive_char('600');
$char=chr($AGI-wait_for_digit('600'));
print STDERR input form rec char : $char\n;
if($char eq *)
{
print STDERR Dialing your number\n;
$srcfile=/tmp/mycall;
$dstfile=/var/spool/asterisk/outgoing/mycall;
open(MYCALL,$srcfile) || die Cant't open file :$srcfile
$!\n;
print MYCALL Channel:IAX2/bali:[EMAIL PROTECTED]/[EMAIL PROTECTED];
print MYCALL MaxRetries:2\n;
print MYCALL RetryTime:60\n;
print MYCALL WaitTime:30\n;
print MYCALL Context:atif\n;
print MYCALL Extension:22\n;
print MYCALL Priority:1\n;
close MYCALL;
# cp($srcfile,$dstfile);
print STDERR dialing complete...\n;
}
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Re: [Asterisk-Users] Line appearances
Hi Joseph, I'll assume you mean a 7960 with the SIP image... Yes, you can register the same SIP client multiple times. Each line appearance on a 7960 configured for SIP is a separate SIP client. Each can register with completely different SIP proxies (providers) or you can have several registrations for the same directory number (DN) so that instead of call waiting, additional calls appear at the next available line appearance. I can't answer your coded question since I always use g.711ulaw. -brian Joseph wrote: I am trying to get an understanding of how line appearances work like on the cisco 7960 phones. Is there a wiki somewhere about how this works? Also, the 7960 phones let you register more than one ext. Why would you want more than one or is this connected to line appearances? Is there a way to have phones use more than one codec, say use g.711 to talk with * and g.729 to talk with another phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line appearances
Thanks for the help. Does that mean that say I had an extension 240 on line appearance one, I could make button 2,3,4 also register the same ext number? Would * care that there were multiple entries from the same ip for the same ext? Thanks again for the tips. On Tue, 2004-05-11 at 09:28, Brian Cuthie wrote: Hi Joseph, I'll assume you mean a 7960 with the SIP image... Yes, you can register the same SIP client multiple times. Each line appearance on a 7960 configured for SIP is a separate SIP client. Each can register with completely different SIP proxies (providers) or you can have several registrations for the same directory number (DN) so that instead of call waiting, additional calls appear at the next available line appearance. I can't answer your coded question since I always use g.711ulaw. -brian Joseph wrote: I am trying to get an understanding of how line appearances work like on the cisco 7960 phones. Is there a wiki somewhere about how this works? Also, the 7960 phones let you register more than one ext. Why would you want more than one or is this connected to line appearances? Is there a way to have phones use more than one codec, say use g.711 to talk with * and g.729 to talk with another phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- respectfully, Joseph - (606) 477-2355 x140 --= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help-listening to my mailbox
How can I listen to the masseges that I have in my mailbox? regards leo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line appearances
Joseph wrote: I am trying to get an understanding of how line appearances work like on the cisco 7960 phones. Is there a wiki somewhere about how this works? General info on 79XX: http://voip-info.org/wiki-Asterisk+phone+cisco+79xx Also, the 7960 phones let you register more than one ext. Why would you want more than one or is this connected to line appearances? Yes Is there a way to have phones use more than one codec, say use g.711 to talk with * and g.729 to talk with another phone? No F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help-listening to my mailbox
add this to extensions.conf exten = 55,1,voicemailMain (replace 55 with whatever you want) exten = 55,2,Hangup reload and dial 55 - Original Message - From: leonardo [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 10, 2004 7:08 PM Subject: [Asterisk-Users] help-listening to my mailbox How can I listen to the masseges that I have in my mailbox? regards leo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible TICKING sound
I've fought with this problem on and off. Number 1 thing to check is /proc/interrupts to ensure that your card isn't sharing an interrupt with something else. Number 2 is a bit of an unknown variable - my guess is either electrical noise, or perhaps vibrations affecting your card inside your box. I find that carefully remounting my tdm400p/x100p so that nothing at all is touching it (no wires, no plastic, nothing - except at the mount point) will make the problem go away the majority of the time. If it doesn't go away, try re-mounting again. It's a little scary, especially when you're working with a small-form-factor machine. Ryan On 10-May-04, at 8:43 PM, Paul Mahler wrote: i'm getting a tick every second or so on all my calls. All channels are zap channels. Does anyone know how to fix this? Thanks! Paul Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.signate.com/ Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Long delays when talking betwwen SIP phones
I am experiencing some long delays when talking SIP to SIP (I will talk, it it takes about 2 seconds for that conversation to reach the other SIP phone, and when they talk back, same thing) on our local LAN. Traffic is relatively light on our LAN. But talking SIP to PSTN works fine, no delays. If anyone has experienced similar delays, what have you done to correct it? I have tried different codecs (GSM to GSM, ulaw to ulaw) and still the same result. I truly think this that the problem lies within the LAN, but wanted to make sure that others have not had the same problem and it turned out to be a config issue. Any assistance is greatly appreciated. Thanks in advance.Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200
Re: [Asterisk-Users] Line appearances
That's exactly what it means. And, Asterisk will do the right thing. -brian Joseph wrote: Thanks for the help. Does that mean that say I had an extension 240 on line appearance one, I could make button 2,3,4 also register the same ext number? Would * care that there were multiple entries from the same ip for the same ext? Thanks again for the tips. On Tue, 2004-05-11 at 09:28, Brian Cuthie wrote: Hi Joseph, I'll assume you mean a 7960 with the SIP image... Yes, you can register the same SIP client multiple times. Each line appearance on a 7960 configured for SIP is a separate SIP client. Each can register with completely different SIP proxies (providers) or you can have several registrations for the same directory number (DN) so that instead of call waiting, additional calls appear at the next available line appearance. I can't answer your coded question since I always use g.711ulaw. -brian Joseph wrote: I am trying to get an understanding of how line appearances work like on the cisco 7960 phones. Is there a wiki somewhere about how this works? Also, the 7960 phones let you register more than one ext. Why would you want more than one or is this connected to line appearances? Is there a way to have phones use more than one codec, say use g.711 to talk with * and g.729 to talk with another phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Line appearances
With the same extension presented twice (e.g., 2000), is there anyway to have the phone not ring on the 2nd presentation of the line if first presentation is currently engaged? Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error compiling Zaptel
Hi, I just finished downloadingasterisk and when trying to compile the zaptel drivers, get the following errors. I dont have a clue whats going on... can someone help. In file included from /usr/include/linux/module.h:20,from zaptel.c:44:/usr/include/linux/modversions.h:1:2:#error Modules should never use kernel-headers system headers,/usr/include/linux/modversions.h:2:2: error but rather headers from an appropriate kernel-source package./usr/include/linux/modversions.h:3:2: #error Change -I/usr/src/linux/include (or similar) to/usr/include/linux/modversions.h:4:2: #error -I/lib/modules/$(uname -r)/build/include/usr/include/linux/modversions.h:5:2: #error to build against the currently-running kernel.make: *** [zaptel.o] Error 1 Thanks San
Re: [Asterisk-Users] Error compiling Zaptel
On Tue, 2004-05-11 at 10:16, San Singhania wrote: Hi, I just finished downloading asterisk and when trying to compile the zaptel drivers, get the following errors. I dont have a clue whats going on... can someone help. /usr/include/linux/modversions.h:1:2:#error Modules should never use kernel-headers system headers, /usr/include/linux/modversions.h:2:2: error but rather headers from an appropriate kernel-source package. /usr/include/linux/modversions.h:5:2: #error to build against the currently-running kernel. make: *** [zaptel.o] Error 1 Lets draw your attention to these excerpts of the errors you sent. Once you read those error messages, you should formulate the opinion that having the kernel source for your running kernel is IMPORTANT. Then you will also need to make sure you have the proper modversions.h built by the kernel source too. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * INSTRUCTIONS FOR MEMBERS OF THE COMMUNITY * Please read
On Tuesday 11 May 2004 01:10, Olle E. Johansson wrote: References: [EMAIL PROTECTED] In-Reply-To: [EMAIL PROTECTED] Welcome to the Asterisk users community! Might I make the request that the next time you post this, you do NOT reply to a message that's two months old? Those of us with threaded mail readers have to page back two months in our histories to read your message. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Error compiling Zaptel
Title: Message Your question would be better posed on a Linux site, as you're having problems with your linux installation, not so much with Asterisk itself. Search google for "linux recompile kernel" and add the actual linux distro as a search term (redhat, debian, freebsd, mandrake, whatever). See if you find your answer, if not, ask in the same place you find close matches. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of San SinghaniaSent: Tuesday, May 11, 2004 11:11 AMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Error compiling Zaptel Ok, how do I go about doing that? I am not exactly a linux expert, infact this is the 2nd time I am installing it. I would really appreciate some direction on this. Thanks San Lets draw your attention to these excerpts of the errors you sent. Once you read those error messages, you should formulate the opinionthat having the kernel source for your running kernel is IMPORTANT. Thenyou will also need to make sure you have the proper modversions.h builtby the kernel source too. -- Steven Critchfield [EMAIL PROTECTED]
Re: [Asterisk-Users] Sipura, Asterisk, *0, and Call Waiting
You have to fix the dialplan or it won't work. I think it hijacks em by default. bkw - Original Message - From: PBX Tech [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 11, 2004 10:22 AM Subject: [Asterisk-Users] Sipura, Asterisk, *0, and Call Waiting I have searched through all the lists and found that some people have had luck with flash hook, then *0 to answer a call waiting call. I have an Asterisk server with one FXO card, the dialtone for the fxo card is providing by another pbx called a Definity. When I am on the Sipura, and another call comes in, I hear the call-waiting indicator, when I flash hook I just hear tone, if I dial *0 I just hear dead air. Its like the Asterisk isnt flashing the fxo line. Any suggestions? _ MSN Toolbar provides one-click access to Hotmail from any Web page - FREE download! http://toolbar.msn.com/go/onm00200413ave/direct/01/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + VoiceWorks
I have a need to interface Asterisk with a VoiceWorks voicemail system. I was wondering what kind of card would be needed either a FXO or FXS interface? Any help would be appreciated. Thanks, -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the most popular pre-paid billing system?
Hi there, We are writing one right now from scratch using MySQL as backend. We will probably release it open source once the code is mature... It will work with Asterisk via AGI. David Deltapath [EMAIL PROTECTED] 04/04/04 7:47 PM We have a system with multiple Asterisk gateways and an SER proxy. I was wondering if people could share what they have been doing for prepaid billing on either multiple Asterisk gateways, or a single SER proxy? Do people use a radius or do they go right from the mysql database? I have seen people talking about both, but I was wondering if anyone has it working. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Error compiling Zaptel
Do you have a symlink in /usr/src as follows? lrwxrwxrwx 1 root src 20 May 7 11:01 linux - kernel-source-2.4.18 (note that it may differ depending on the kernel source you have?) If youve installed via an apt style package manager, and havnt recompiled your kernel, then visit www.kernel.org, download a stable kernel (I recommended 2.4.26). Extract it to /usr/src/linux-kernelnumber Then create a symlink ln s /usr/src/kernel source /usr/src/linux This should resolve the issues youre having there, else Ive missed the point and just waffled for 5 minutes ;) Hope that helps, Ben Merrills Griffin Internet From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of San Singhania Sent: 11 May 2004 16:17 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Error compiling Zaptel Hi, I just finished downloadingasterisk and when trying to compile the zaptel drivers, get the following errors. I dont have a clue whats going on... can someone help. In file included from /usr/include/linux/module.h:20, from zaptel.c:44: /usr/include/linux/modversions.h:1:2:#error Modules should never use kernel-headers system headers, /usr/include/linux/modversions.h:2:2: error but rather headers from an appropriate kernel-source package. /usr/include/linux/modversions.h:3:2: #error Change -I/usr/src/linux/include (or similar) to /usr/include/linux/modversions.h:4:2: #error -I/lib/modules/$(uname -r)/build/include /usr/include/linux/modversions.h:5:2: #error to build against the currently-running kernel. make: *** [zaptel.o] Error 1 Thanks San
[Asterisk-Users] Kernel Freezes with T100P
Hey Folks, I have an extremely reproducable problem with the T100P card. I'm running one card in an Athlon XP 1800 machine running Fedora Core 1. The kernel is a stock 2.4.26 kernel pulled from kernel.org. I'm going to be using the T1 interface for data only. If I bring the interface up, then down, then up again, the machine will freeze within 30 seconds or so. There are are no debugging messages even if debugging is turned on. Occasionally it takes another up/down cycle to cause the machine to freeze, but that is rare. I've seen other incidents of machine's freezing in the archives, but no real answers to the problem. Any idea where to start with this? Thanks, -Zach. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kernel Freezes with T100P
Zach Chambers wrote: Hey Folks, I have an extremely reproducable problem with the T100P card. I'm running one card in an Athlon XP 1800 machine running Fedora Core 1. The kernel is a stock 2.4.26 kernel pulled from kernel.org. I'm going to be using the T1 interface for data only. If I bring the interface up, then down, then up again, the machine will freeze within 30 seconds or so. There are are no debugging messages even if debugging is turned on. Occasionally it takes another up/down cycle to cause the machine to freeze, but that is rare. I've seen other incidents of machine's freezing in the archives, but no real answers to the problem. Any idea where to start with this? Thanks, -Zach. Ever tried to use the latest fedora kernel. Also what type of chipset do you have? Might be an chipset incompatiblity. Did you ever try swapping arround the pci card into a nother PCI slot or playing arround with the IRQ's? -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible TICKING sound
Currently, I'm working with a tdm400p with 2 FXO, 2 FXS. the persistent tick tick tick tick has come back (after 3 days tick-free operation). Has anyone else experienced a similar problem with ZAP channels? What steps did you take to resolve it? Are you seeing anything in dmesg? after you've heard some ticking, run dmesg and look for anything suspicious about the TDM card; things about power module resets and stuff. I find that when I hear a good click or buzz I see that the power supply farted or otherwise confused the card. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk + VoiceWorks
Why on earth would you wanna do something like that? Asterisk has voicemail and you even have the src so you can add those nifty features the PHB's like to have but never use! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of AstGrp Sent: Tuesday, May 11, 2004 11:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk + VoiceWorks I have a need to interface Asterisk with a VoiceWorks voicemail system. I was wondering what kind of card would be needed either a FXO or FXS interface? Any help would be appreciated. Thanks, -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura, Asterisk, *0, and Call Waiting
Take a look here: http://www.voip-info.org/wiki-Asterisk+Avaya t o n y On Tue, 2004-05-11 at 13:38, brian k. west wrote: You have to fix the dialplan or it won't work. I think it hijacks em by default. bkw - Original Message - From: PBX Tech [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 11, 2004 10:22 AM Subject: [Asterisk-Users] Sipura, Asterisk, *0, and Call Waiting I have searched through all the lists and found that some people have had luck with flash hook, then *0 to answer a call waiting call. I have an Asterisk server with one FXO card, the dialtone for the fxo card is providing by another pbx called a Definity. When I am on the Sipura, and another call comes in, I hear the call-waiting indicator, when I flash hook I just hear tone, if I dial *0 I just hear dead air. Its like the Asterisk isnt flashing the fxo line. Any suggestions? _ MSN Toolbar provides one-click access to Hotmail from any Web page - FREE download! http://toolbar.msn.com/go/onm00200413ave/direct/01/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need help: X100P connection/configuration in GERMANY
hi i need help in connecting a X100P-clone in germany. Basic questions: a)what configuration do I need ? what is the difference (maybe explain in german? via email?) fxs_ls: FXS (Loop Start) fxs_gs: FXS (Ground Start) fxs_ks: FXS (Kewl Start) b) can I use one card for connecting to the telephone network (in this case my analog pbx) and the same card for connecting an analog phone (the card has two connectors with line and telephone signs). I want to connect the asterisk pbx to my analog pbx and want to make calls in booth directions. with FXS Krwl Start I can make calls from the analog-pbx to asterisk (and connected SIP-phone). but in the other direction (from asterisk to analog pbx) I cant make calls. the extensions.conf definition is exten = 111,1,Dial,Zap/1 but nothing happen if I dial this extension from my sip phone. Executing Dial(SIP/thorstengehrig-5c59, Zap/1) in new stack -- Called 1 -- Zap/1-1 answered SIP/thorstengehrig-5c59 regards Thorsten Gehrig (Thorsten at Gehrig.de)
Re: [Asterisk-Users] Kernel Freezes with T100P
Ever tried to use the latest fedora kernel. Also what type of chipset do you have? Might be an chipset incompatiblity. Did you ever try swapping arround the pci card into a nother PCI slot or playing arround with the IRQ's? -- Thomas Thomas, I could not get the zaptel drivers to compile using any of the fedora kernels which is what drove me to the stock kernel to begin with. I'll check the archives again on that issue. The chipset is a VIA chipset. I don't know much about it other than that yet. I'll try PCI slot move as well. Thanks, -Zach. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need help: X100P connection/configuration in GERMANY
You need to tell it what to dial. exten = 111,1,Dial,Zap/1/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thorsten Gehrig Sent: Tuesday, May 11, 2004 2:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Need help: X100P connection/configuration in GERMANY hi i need help in connecting a X100P-clone in germany. Basic questions: a)what configuration do I need ? what is the difference (maybe explain in german? via email?) fxs_ls: FXS (Loop Start) fxs_gs: FXS (Ground Start) fxs_ks: FXS (Kewl Start) b) can I use one card for connecting to the telephone network (in this case my analog pbx) and the same card for connecting an analog phone (the card has two connectors with line and telephone signs). I want to connect the asterisk pbx to my analog pbx and want to make calls in booth directions. with FXS Krwl Start I can make calls from the analog-pbx to asterisk (and connected SIP-phone). but in the other direction (from asterisk to analog pbx) I cant make calls. the extensions.conf definition is exten = 111,1,Dial,Zap/1 but nothing happen if I dial this extension from my sip phone. Executing Dial(SIP/thorstengehrig-5c59, Zap/1) in new stack -- Called 1 -- Zap/1-1 answered SIP/thorstengehrig-5c59 regards Thorsten Gehrig (Thorsten at Gehrig.de) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kernel Freezes with T100P
Zach Chambers wrote: Ever tried to use the latest fedora kernel. Also what type of chipset do you have? Might be an chipset incompatiblity. Did you ever try swapping arround the pci card into a nother PCI slot or playing arround with the IRQ's? -- Thomas Thomas, I could not get the zaptel drivers to compile using any of the fedora kernels which is what drove me to the stock kernel to begin with. I'll check the archives again on that issue. The chipset is a VIA chipset. I don't know much about it other than that yet. I'll try PCI slot move as well. Thanks, -Zach. I actually just installed the zaptel driver for fedora from cvs. Christian helped me getting the latest CVS version of asterisk, libpri and zaptel. Then what I did is grabbed the kernel-source for fedora. I guess in your case you want to do an kernel update. yum update yum install kernel-source It should install the 2188 kernel and 2188 kernel source. Then just go to /usr/src/zaptel make clean make make install and same with libpri and asterisk. Just make sure you removed the wcfxo kernel mod be4 you install zaptel. And then reboot the box as when I modprobe wcfxp it gave me an actuall kernel panic :-) After reboot everything worked just fine. It even seems like my dropped call issue is gone. If you need a copy of the yum.conf file let me know. -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Areski CDR graph incorrect
Is anyboby using the areski CDR reporting tool? I have installed asterisk-stats v1.2 three days ago, but I found a possible bug in it. My calls compare graphic shows most of the on the calls at first hours past midnight, and it never logs anything after lunch time. This is wrong, my calls are made on business hours. The call log lists those calls at the right time. Is there something I should set on the graphic engine, like a timezone or something? I´m on Brazil, timezone GMT -3. Tried to email the author directly but got no answer. Thanks, Gelson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disabling agent call logging
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Anyone know how to stop agent calls from being logged? It appears that the options in /etc/asterisk/agents.conf do not work with respect to call logging. All calls are logged in /var/spool/asterisk/monitor/something-agent#-something else.wav, even with the option to log disabled. It records 3 files for each call: an incoming file, an outgoing file, and another file, which I assume is the merging of incoming and outgoing data (haven't actually listened to any of these files, yet). The only reason I'm complaining is that it ate up all the hard drive space on my first Asterisk system before I realized what was happening (2.5GB free space ... gone). I didn't see a need to keep the files, so I've just been deleting them, but I'd like to not have them recorded in the first place (or is this something that isn't legal?). Thanks. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAoTJnuYsUrHkpYtARAgILAJ9/emsvzLP4Q3sHCBg4s8K9xh/ltgCgg3bQ qq9LYjTi+otsVch7AcVSV00= =W8vW -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible TICKING sound
Ive found this in audio apps on other boxes when the power supply is really loaded down hard. Just one more maybe for you to check. Have you blown the dust out of the P/S lately? Dirt and temp variations seem to affect it as well...found this with audio equipment at a broadcast station. (Streaming server on a Linux box) Not saying its your situation, but wont hurt to check. If it does help, a beefier power supply might help here. It cured my case. Marc At 10:45 5/11/2004, you wrote: I've fought with this problem on and off. Number 1 thing to check is /proc/interrupts to ensure that your card isn't sharing an interrupt with something else. Number 2 is a bit of an unknown variable - my guess is either electrical noise, or perhaps vibrations affecting your card inside your box. I find that carefully remounting my tdm400p/x100p so that nothing at all is touching it (no wires, no plastic, nothing - except at the mount point) will make the problem go away the majority of the time. If it doesn't go away, try re-mounting again. It's a little scary, especially when you're working with a small-form-factor machine. Ryan On 10-May-04, at 8:43 PM, Paul Mahler wrote: i'm getting a tick every second or so on all my calls. All channels are zap channels. Does anyone know how to fix this? Thanks! Paul Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.signate.com/ Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] midget packet received
Hello, I just upgraded to the latest cvs and I am seeing these messages in my logs now. They are almost always 2 minutes apart, is this a problem? I checked the source, appears to be saying the network is not sending the right amount of data (recvfrom) when compared to the iax2 mini struct. Seems really odd that it happens every 2 minutes, and only after I upgraded to the newest cvs. Network interface does not show incrementing errors, carrier transitions etc. Sound like a problem with the source not the network, can anyone here set me straight? May 11 16:06:27 WARNING[81926]: File chan_iax2.c, Line 3929 (socket_read): midget packet received (0 of 4 min) May 11 16:08:28 WARNING[81926]: File chan_iax2.c, Line 3929 (socket_read): midget packet received (0 of 4 min) May 11 16:10:28 WARNING[81926]: File chan_iax2.c, Line 3929 (socket_read): midget packet received (0 of 4 min) -- Brian J. Schrock Anistone Technologies, LLC Phone: 614-798-9106 ext. 2 Cell: 614-537-2817 E-Mail: [EMAIL PROTECTED] http://www.anistonetech.com http://www.vipcomtel.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
