[asterisk-users] Disable MoH for certain phones

Hi, Is it possible to configure asterisk so it doesn't play MoH from certain phones? Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Zap channels unavailable?

of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jan.sarin at securia.se Sent: Tuesday, July 17, 2007 9:44 AM

[asterisk-users] How to change Zap channel negotiation/exclusive etc..?

Hi, I just spoke with my telco about a problem I have with some zap channels getting stuck in PRI flags: Resetting when we have a heavy load (lots of calls). The technican I spoke with told me that this is most likely because asterisk says the zap channel should be exclusive and this causes

[asterisk-users] PRI Change Channel Identification from Exclusive to Preferred or Negotiation?

Hi, Does anyone know how to change the channel identification on a PRI line on our asterisk from 'Exclusive' to 'Preferred' or 'Negotiation'? Is this even possible? Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com--

[asterisk-users] Zap channels unavailable?

Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really

Re: [asterisk-users] Zap channels unavailable?

because of the heavy load on the server (cpu running at max 20% with 40-50 simultaneous calls, so why would it be this?). Regards, Jan -- Have you tried setting resetinterval=never in zapata.conf? On Tue, 2007-07-17 at 15:43 +0200, jan.sarin at securia.se wrote: Hi, Lately

Re: [asterisk-users] Zap channels unavailable?

it be this?). Regards, Jan -- Have you tried setting resetinterval=never in zapata.conf? On Tue, 2007-07-17 at 15:43 +0200, jan.sarin at securia.se wrote: Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60

[asterisk-users] QUEUE_WAITING_COUNT

Hi, I'm playing around with the QUEUE_WAITING_COUNT function but it always seems to return zero? I've tried everything. I suspect that this feature is not implemented in 1.2.7 which I am running.. Does anyone know in which version this function was added? Regards, Jan

[asterisk-users] Random all circuits busy now message

Hi, We have quite a large setup working just fine most of the time. We have 60 outgoing lines on PRI and we never use all of these lines. But sometimes we get the all circuits busy now message, seemingly random. Sometimes we get it before the call even goes through to PSTN. Sometimes after 5 or

[asterisk-users] Set(CALLERID(all) not working with 'unknown' call?

Hi, This is really strange (but probably simple solution). The CALLERID(all) setting doesn't seem to work when the incomming callerid is 'unknown'. Dialplan looks like this: exten = _3072,1,Answer exten = _3072,n,Set(CALLERID(all)=DIRECT 0850553072) exten =

SV: [asterisk-users] Set(CALLERID(all) not working with 'unknown'call?

Hi Chris, Yes the call was from PSTN and your solution worked great! I've read about SetCallerPres earlier but I didn't connect the dots this time. Thanks alot! :) Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Christoph Fürstaller

[asterisk-users] Asterisk logging everything?

Hi, Is it possible to keep asterisk from logging exactly everything? I can do the logger rotate and keep the files small enough, but I think it's unneccesary to log exactly all data. File grows by about 5 gb per month! Thanks! Regards, Jan ___

[asterisk-users] Manage 'full' log file

Hi, I need some help on how to manage the full log file. It's getting quite large now and I'd like to clear it. Is there any simple command for this or should I just delete the file (need to be sure this won't affect the system). Also - how do I keep the log file from growing so large? Thanks!

SV: [asterisk-users] Manage 'full' log file

Thanks for the quick response! I read about logrotate at voip-info.org but I didn't quite understand it. I'm no asterisk/linux expert unfortunately. First of all. What exactly does happen when I run: /usr/sbin/asterisk -rx 'logger rotate' Does it clear the file and create a new one? Can I run

SV: [asterisk-users] Manage 'full' log file

Super! Thanks! Now I see how the script works a bit more clearly. :) I still don't understand what happens if I run: /usr/sbin/asterisk -rx 'logger rotate' Can I run the above without having the script? What will the command do? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL

[asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip?

Hi, I have lately noticed that we sometimes get choppy sound when recieving calls from the PSTN (on a TE410P-card) that get sent to an external SIP extension (over the internet) who has a somewhat bad connection. The strange thing is that it still sounds good when calling internally to the

SV: [asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip?

Sorry. It sould say SIP-to-Zap not the other way around. Meaning that the Zap user is heard fine, but the external-SIP user is choppy when calling out on Zap (not when calling SIP-to-SIP though). -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL

SV: [asterisk-users] Help debugging strange asterisk behaviour

Ok. I have an update! When all the problems begin (described below) the 'show queues' command doesn't work either!! The queues have dissappeared (or asterisk is unable to read them?)! What the heck is going on? Why are the queues gone by themselves? When I restart they're back. Queues.conf in

SV: [asterisk-users] Help debugging strange asterisk behaviour (update)

Allright. I think I've located the problem. It's reported here: http://bugs.digium.com/view.php?id=7604 I'm not however using 'show queues'. It stops responding anyway. Maybe because we use freepbx and flash operator panel. Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED]

[asterisk-users] Encoding recorded queue calls to mp3

Hi, What do I need to add to the dialplan BEFORE a caller enter a queue so that the recorded call (generated by queue monitor) is encoded to mp3? I'm defining the monitor save destination with: exten = 1,n,Set(MONITOR_FILENAME=/new/monitor/queues/${TIMESTAMP}-${CALLERID(nam

SV: [asterisk-users] Help debugging strange asterisk behaviour

I think I'm using native since I don't recall installing anything else (except lame codec). How do I check which I am using? I'm unfortunately no asterisk expert that's why I need your help! ;) My musiconhold.conf (I have no musiconhold_additional.conf): ; ; Music on hold class definitions ;

SV: [asterisk-users] Help debugging strange asterisk behaviour

I'm thinking this could be a queue problem? But I still don't understand why the hell it just flips out after a few hours. Now it all ran for about 12 hours since last reboot (longest so far). And this config worked on my old install of asterisk... Problem description (one of them): Incoming

[asterisk-users] chan_zap.c: Failed to read gains: Invalid argument

Hi, I'm recieving the following error in my asterisk log (when starting *): chan_zap.c: Failed to read gains: Invalid argument Why? Attaching my zapata.conf and zaptel.conf. Using TE405P. Thanks! zaptel.conf: span=1,1,0,ccs,hdb3

[asterisk-users] Help debugging strange asterisk behaviour

Hi, I'm one of those types who want to know what the heck is wrong when something is wrong. I just installed a new server (see config below) and it all works fine for a few hours. But after 3-5 hours asterisk starts behaving VERY strangely for no apparent reason... 1) MoH stops playing 2) Some

[asterisk-users] SV: Help debugging strange asterisk behaviour

Actually I found one error now after a reboot..Although I don't think it has anything to do with the strange behaviour. Could someone please tell me what this means? Aug 1 16:59:25 DEBUG[6771] chan_zap.c: Failed to read gains: Invalid argument Where is the invalid argument? I've set the gains

[asterisk-users] Compiling zaptel on CentOS x86_64

Hi, I've been trying to compile zaptel on a CentOS x86_64 (4.3) for a couple of days now. I've read probably 10 different fixes to make it work but none of them seem to... Has ANYONE successfully compiled zaptel on the above - if so, what did you do to succeed? Help would be MUCH appreciated!

[asterisk-users] Calls waiting announcement with two or more queues?

Hi, I'm wondering how the calls waiting announcement works when you have several queues? We have different people answering different kind of calls and we have three queues setup because of this... If I where to use queue-callswaiting - how would it behave? Would it only prompt the caller the

[asterisk-users] Queue hold position in other language?

Hi, I am wondering how I can change the language of queue hold position. This is probably pretty simple (yes I know I have to record my own soundfiles). What I don't get is where to set the numbers? In queues.conf there are settings for: queue-youarenext = queue-youarenext

SV: [asterisk-users] Queue hold position in other language?

Okay, thanks! I already have set language to 'se' in indications.conf. Next question. If asterisk where to play a digit - does it look in /sounds/se/digits or /sounds/digits/se ? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Marco Mouta

[Asterisk-Users] HP Proliant server?

Has anyone had any experience running asterisk on a dual-xeon HP Proliant server. Have you had any experience setting up digium cards on this? We've only used Dell before and are thinking about upgrading to a hp ProLiant ML350 G4p. ANY comments (positive/negative) would be appreciated! Regards,

SV: [Asterisk-Users] HP Proliant server?

Thanks! I was looking for that page. I know I've seen it before. I guess we're lucky that we're running asterisk on a PowerEdge 600SC today (since many PowerEdge seem unsupported). ;) But I'm also interested in hands-on experience on running asterisk and digium cards on HP proliants. So if

SV: [Asterisk-Users] HP Proliant server?

Are you running just one or a few simultaneous calls or do you have any experience running many (maybe 30-50) simultaneous calls? Thanks! Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Luca Corti Skickat: den 5 juli 2006 11:10 Till: Asterisk

[Asterisk-Users] Intel E7220 chipset?

According to http://www.digium.com/en/docs/misc/compatibility_notes.php the Intel E7221 chipset and Intel E7525 chipset is somewhat incompatible with Digium hardware. Does anyone know about Intel E7520? Because that's what we're thinking about using. Again. Any common server recomendations to run

[Asterisk-Users] Running 40 active calls (too m uch för CPU?)

Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 of these where running

SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)

I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls. Mvh, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 juli 2006 09:41 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Running 40

SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)

Hello again, I read this interesting article about the TE405P card. How do I check what firmware version my card has? http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html ... And how do I update it if it's an old one? Regards, Jan -Ursprungligt meddelande- Från:

SV: SV: [Asterisk-Users] Running 40 active calls (to o much för CPU?)

Phones are not behind NAT. Every client is on the sameinternal network as the asterisk pbx (nothing is sent throughthe internet). It's not the network since I tested this by calling asterisk from an outside phone (cell) and let asterisk play a message for me. Same "cutting" and "chopping"

SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)

Phones are not behind NAT. Every client is on the sameinternal network as the asterisk pbx (nothing is sent throughthe internet). It's not the network since I tested this by calling asterisk from an outside phone (cell) and let asterisk play a message for me. Same "cutting" and "chopping"

[Asterisk-Users] TE420P/TE415P?

Hi, I just read a pressrelease from VON that Digium will soon be releaseing a couple of new cards. What got me interested was: The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels. Does anyone know when thease will be released and what

[Asterisk-Users] USB headsets?

Hi, What USB headset would you recomend? We have some laptop soundcards that are really bad and I would be glad if you could share your experiences when changing to a USB headset instead of using the built in soundcard in your computer. Thanks! Regards, Jan

SV: [Asterisk-Users] USB headsets?

I don't quite follow you? There are USB headsets that don't require a soundcard at all. They have a built in soundcard which (I suppose) could be better than the crap they build into most laptops. Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]

SV: [Asterisk-Users] USB headsets?

I have a logitech USB headset and a labtec USB headset, and love both. The Logitech has better audio though, so when using it to listen to music, etc., you'd better be looking at something similar. Or get a USB audio-device with input/output jacks, so you can plug in whatever you want...

[Asterisk-Users] Reading queue_logs

Hi, Are there any good free win32 apps for reading queue_logs? Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Echo cancel voip channel?

Hi, Is it possible to echo cancel a voip (sip) channel/trunk in asterisk somehow? If not, this function would be neat since some providers really suck at echocancelling when you call out on pstn. Regards, Jan ___ --Bandwidth and Colocation provided by

SV: [Asterisk-Users] Polycom 501 - Disable DND feature?

Solved! In sip.cfg: keys key.scrolling.timeout=1 key.IP_500.9.function.prim=Null/ Thanks to Derek for this solution! Regards, Jan Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 maj 2006 15:46 Till:

[Asterisk-Users] Polycom 501 - Disable DND feature?

Hi, Is it possible to disable the DND feature on a Polycom 501? Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

SV: [Asterisk-Users] Polycom 501 - Disable DND feature?

Well, yes and no. I tested that before and it causes a silent ring instead of a call rejection. I actually want to disable the entire feature. So the phone always rings unless you're actually on the phone. Thanks for the reply though! Regards,Jan Från: [EMAIL PROTECTED] [mailto:[EMAIL

[Asterisk-Users] Phone UNREACHABLE: Plays agent-incorrect to Queue-caller ??

Hi, I just encountered a very strange problem. When some of our phones that connect to asterisk through the Internet went down - the callers on the queue got the agent-incorrect message played to them as soon as asterisk tried to call the extention. Why? The agents where logged on via

[Asterisk-Users] SIP trunk ring tone

Hi, I'm wondering what I need to change to get the swedish type ring on a SIP-trunk. When I make an inbound call i still have the US-type of ring on my SIP trunks. I need help on changing this. However I've successfully changed this on the Zap interface for all inbound calls. Thanks in advance!

[Asterisk-Users] Receive fax (libtiff problem?)

Hi, I'm trying to receive faxes with asterisk. Everything works fine except the tif to pdf conversion. Even though the tif file is okay, the pdf always turns out to be empty (blank).. I read that this might be caused by incompatible libtiff and that I should install another version. But when

[Asterisk-Users] queue_log timestamp?

Hi, How do I read (make sense of) the timestamp in the queue_log? I'm probably just slow but I don't understand it. Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

[Asterisk-Users] Polycom 501 ACDlogin

Hi, I have several Polycom 501 connected to asterisk. The phone has an ACD-login function that I'd like to use. But I can't find find much information about this. I've read a post on [EMAIL PROTECTED] (http://bugs.digium.com/view.php?id=6119) about this function but I'm not really clear on if

SV: [Asterisk-Users] Polycom 501 ACDlogin

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -Ursprungligt

SV: [Asterisk-Users] Polycom 501 ACDlogin

Yes of cource. But that's not what I'm interested in. I want to be able to see on the phone if the agent is logged on or not. Automatic logon is not an option either. Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Douglas Garstang Skickat:

SV: [Asterisk-Users] BAD/GOOD Echo Cancel

Im curious. Does anyone have experienced echo-problems that later where solved by buying a hardware-echo canceller such as the Wildcard TE411P? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För James Harper Skickat: den 6 februari 2006 11:46

SV: [Asterisk-Users] Help on queues

What kind of help do you need then? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Zach A Skickat: den 6 februari 2006 18:31 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: RE: [Asterisk-Users] Help on queues There is

SV: [Asterisk-Users] Outbound Caller ID number on E1

How do you set the CallerID? Have you checked with your provider that they've enabled callerid? If yes, are you using a correct number that the provider allows? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Garth van Sittert Skickat: den 2

SV: [Asterisk-Users] delaying answer for a number of rings or anamount of time

http://lists.digium.com/pipermail/asterisk-users/2005-September/125146.html -Ursprungligt meddelande- Från: [EMAIL PROTECTED] genom Brian J. Murrell Skickat: to 2006-02-02 20:14 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] delaying answer for a number of rings or anamount

SV: SV: [Asterisk-Users] delaying answer for a number of ringsor anamount of time

From what I understand it means that the *hardware* in your computer *acknowledges* the call as soon as it is recieved and then sends it to asterisk dialplan for processing. You would essentially need to put the delay before the call ever reaches asterisk. So this problem isn't asterisk

SV: [Asterisk-Users] Re: CallerID Problem

Seems to me like the negotiation fails for some reason. Maybe you are trying to use a callerid that isn't allowed? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] genom Gary Richardson Skickat: on 2006-02-01 21:45 Till: Asterisk Users Mailing List - Non-Commercial

SV: [Asterisk-Users] Re: CallerID Problem

This is what i found on Cisco's site: Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a 488 message to an Invite message. Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec

SV: [Asterisk-Users] Set caller id on Swedish PRI (euroisdn)

Try setting the Callerpresentation to something else: http://www.voip-info.org/wiki/page_history.php?page_id=1682preview=2 SetCallerPres(prohib) actually worked! Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com --

[Asterisk-Users] Set caller id on Swedish PRI (euroisdn)

Hi, I have a problem with setting outgoing caller id to nothing (secret) on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID seems to work fine when connecting the same line to a Ericsson PBX - so something must be wrong in my settings, but I don't know what. I've tried:

[Asterisk-Users] Queue Autologoff over trunks

Hi, I have set-up two asterisk servers with an IAX trunk between them. There is a queue-system and callagents configured on one of them Agents on both servers logon to the one queuesystemI have set up, which works fine. But autologoff (agents.conf) only seems to work with agents connected

SV: SV: [Asterisk-Users] Queues and call waiting indication

Could you post an example of how you've solved it. I read something about this earlier but didn't quite figure it out. I already use AgentCallbackLogin... And I still don't understand why this behavior isn't standard for queues. Does this really fix the agent makes an outgoing call but still

SV: [Asterisk-Users] Queues and call waiting indication

Hi, This issue has been discussed probably a million times on every asterisk forum in the world and I have the same problem too. Another problem you would have with the agents is that when they make an outgoing call they are not regarded as busy by asterisk and it sends more calls to the

SV: SV: [Asterisk-Users] Queues and call waiting indication

My suggestion would be the one-line eyeBeam phone under development. Check out support.xten.com. //Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 18 oktober 2005 14:48 Till: Asterisk Users Mailing List -