[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: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channels unavailable?
Hi, I was talking to a technican at our telco yesterday and he told me that this problem was most likely caused by our PBX sending channel identification Exclusive when we dial out. If there's a heavy load and someone is dialing in on the same time on the same channel that we try to dial out from - it causes a deadlock. He said some Cisco PBXs have the same problem. Now, I'm no asterisk expert and I don't quite understand what this means. I've emailed the list asking if this can be changed to Preferred or Negotiation as the technican told me to. But I got no response yet. I did however solve the problem by reversing the channels that we dial out from (so now it tries the last channel first and then backwards to the first). Since all of our incoming calls come from the first to the last this minimizes the risk of a collision of the incoming/outgoing calls. This is of cource no long-term solution but anyway. I need to know if it's possible to change channel identification (whatever that is) to preferred or negotiation. Regards, Jan Martin Smith wrote: Hello Jan, We have also been seeing this issue, and we are running Asterisk 1.2.17/Zaptel 1.2.16/LibPRI 1.2.4-r2. We have been informed by our PRI provider that a 3rd party vendor has applied firmware to some hardware along our path, and that it has an unfortunate bug of hanging B-channels in the PRI flags resetting state. We have been assured that the vendor has been given a 30-day deadline (starting maybe 2 weeks ago) to fix the problem, and that it will go away soon. In the mean time, we've also had to restart Asterisk to free our B-channels for use, and any link-level event potentially re-hangs them. Keep us posted if you find out anything! Martin Smith, Systems Developer martins at bebr.ufl.edu Bureau 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 To: asterisk-users at lists.digium.com Subject: [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 was no solution (?). I'm not sure if this problem has occured now because of more load on the pbx but the machine should take it just fine (2x3,0 ghz xeon with 1 gb ram etc). Restarting asterisk makes the zaps' available again but they get locked later again. It seems it's always the same channels that are unavailable too? This one is unavailable and not being used... It's been in PRI Flags state resetting for hours now. Channel: 1 File Descriptor: 11 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 702821667 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook If anyone can help me with this I'd be really glad. Thanks. Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 problems. I need to try to change this to 'negotiation' or something else, but I don't understand where I do this. Please help. Our zapata.conf below: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=se context=from-pstn switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callerid=asreceived usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=400 jitterbuffers=8 rxgain=-1.0 txgain=-1.5 group=0 callgroup=1 pickupgroup=1 immediate=no overlapdial=no channel = 1-15,17-31,32-46,48-62 group=1 channel = 94-108,110-124 group=2 context=from-internal signalling=pri_net channel = 63-77,79-93 ;Include genzaptelconf configs #include zapata-auto.conf Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 was no solution (?). I'm not sure if this problem has occured now because of more load on the pbx but the machine should take it just fine (2x3,0 ghz xeon with 1 gb ram etc). Restarting asterisk makes the zaps' available again but they get locked later again. It seems it's always the same channels that are unavailable too? This one is unavailable and not being used... It's been in PRI Flags state resetting for hours now. Channel: 1 File Descriptor: 11 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 702821667 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook If anyone can help me with this I'd be really glad. Thanks. Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channels unavailable?
Hi, No I havn't tried that. That entry wasn't even in there so I'll try it. I'll let you know if it helped. The odd thing is that this problem started yesterday. And our asterisk has been running for +1 year without these kind of problems. So either our telco has changed something OR it's 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 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 was no solution (?). I'm not sure if this problem has occured now because of more load on the pbx but the machine should take it just fine (2x3,0 ghz xeon with 1 gb ram etc). Restarting asterisk makes the zaps' available again but they get locked later again. It seems it's always the same channels that are unavailable too? This one is unavailable and not being used... It's been in PRI Flags state resetting for hours now. Channel: 1 File Descriptor: 11 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 702821667 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook If anyone can help me with this I'd be really glad. Thanks. Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channels unavailable?
Okay, I've got an update on the resetinterval=never... same thing even though i added the line to zapata.conf and restarted the server. Now the load wasn't even high, maybe 6-7 calls. I think I just might call my telco, feels like it's their issue, but if anyone has any other suggestions let me know and I'll try them! Channel: 7 File Descriptor: 17 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 708307496 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook Regards, Jan -Ursprungligt meddelande- Från: Jan Sarin Skickat: den 17 juli 2007 16:57 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: SV: [asterisk-users] Zap channels unavailable? Hi, No I havn't tried that. That entry wasn't even in there so I'll try it. I'll let you know if it helped. The odd thing is that this problem started yesterday. And our asterisk has been running for +1 year without these kind of problems. So either our telco has changed something OR it's 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 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 was no solution (?). I'm not sure if this problem has occured now because of more load on the pbx but the machine should take it just fine (2x3,0 ghz xeon with 1 gb ram etc). Restarting asterisk makes the zaps' available again but they get locked later again. It seems it's always the same channels that are unavailable too? This one is unavailable and not being used... It's been in PRI Flags state resetting for hours now. Channel: 1 File Descriptor: 11 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 702821667 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook If anyone can help me with this I'd be really glad. Thanks. Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 6 rings etc. It seems that the carrier is signalling something and asterisk always interprets this as all circuits busy now whilst a old-fashioned exchange might say busy or wrong number or hangup etc. What might cause asterisk to handle signals like this? Thanks! Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 = _3072,n,Dial(SIP/2001SIP/2002SIP/2003SIP/2004SIP/2201SIP/2202SIP/2 203SIP/2205,30,r) exten = _3072,n,Wait(1) exten = _3072,n,Goto(custom-incoming-3070,1,1) exten = _3072,n,Hangup() Now, it works if the incomming caller id is NOT 'unknown'. Does anyone understand why? We're running Asterisk 1.2.7. Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Skickat: den 29 mars 2007 15:29 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Set(CALLERID(all) not working with 'unknown'call? -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jan, Is this call from PSTN? Probably the Nr is prohibited in PSTN, then asterisk doesn't set the CALLERID. Try this: exten = _3072,1,Answer exten = _3072,n,SetCallerPres(allowed) exten = _3072,n,Set(CALLERID(all)=DIRECT 0850553072) Look here: http://www.voip-info.org/wiki-Asterisk%20cmd%20SetCallerPres chris... [EMAIL PROTECTED] schrieb: 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 = _3072,n,Dial(SIP/2001SIP/2002SIP/2003SIP/2004SIP/2201SIP/2202SIP /2 203SIP/2205,30,r) exten = _3072,n,Wait(1) exten = _3072,n,Goto(custom-incoming-3070,1,1) exten = _3072,n,Hangup() Now, it works if the incomming caller id is NOT 'unknown'. Does anyone understand why? We're running Asterisk 1.2.7. Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - -- Dipl.-Ing. Kurt Krenn - IT-Beratung Franz-Josef-Strasse 33/4/43, 5020 Salzburg Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel: +43 780 kkrenn (557366) Email: [EMAIL PROTECTED] sip: [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGC78gR0exH8dhr/YRAqf+AJsHuGgk1Ei6czT7+Q08I4wZ1F4DzACfe8V0 Y841CYDBAn518nnYMCbFC1E= =+l5m -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 this manually without
any interruption in the system?
And what does the script do? I understand it rotates the logs. But does it
delete the old files? Where do I put the script? How do I run it? As you can
see I'm really a newbie on this. Unfortunately the docs for asterisk are often
with the expectation that you know everything... :)
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Ex Vitorino
Skickat: den 8 januari 2007 13:10
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] Manage 'full' log file
We've been using logrotate without any issue... We're using
the below quoted configuration. Notice the invocation of
Asterisk's CLI logger reload command so as to close the
old files and open new ones.
Cheers,
--
Ex Vito
/var/log/asterisk/messages /var/log/asterisk/queue_log
/var/log/asterisk/event_log {
weekly
rotate 52
dateext
compress
delaycompress
nocreate
missingok
sharedscripts
postrotate
/usr/sbin/asterisk -rx logger reload
endscript
}
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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 PROTECTED] [mailto:[EMAIL PROTECTED] För Lenz Skickat: den 8 januari 2007 13:13 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Manage 'full' log file Hi Jan, You should use the logrotate in order to delete the log on periodic intervals. This article is meant to do exactly the opposite :) http://astrecipes.net/index.php?n=205 but you get an idea of how to setup log file rotation and how to notify Asterisk that it should open a new file after the log rotation. Hope this helps l. On Mon, 08 Jan 2007 13:00:52 +0100, [EMAIL PROTECTED] wrote: 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! Regards, Jan -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 SIP-to-SIP. Is there any simple answer to why Zap-to-SIP (external) sounds bad when there is a bad connection, but SIP-to-SIP doesn't? The problem (I think) is not with the card or drivers since the problem only occurs when the connection is bad and never on our phones that are on the same internal network with the server. Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 PROTECTED] Skickat: den 21 augusti 2006 15:15 Till: asterisk-users@lists.digium.com Ämne: [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 SIP-to-SIP. Is there any simple answer to why Zap-to-SIP (external) sounds bad when there is a bad connection, but SIP-to-SIP doesn't? The problem (I think) is not with the card or drivers since the problem only occurs when the connection is bad and never on our phones that are on the same internal network with the server. Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 previous e-mail. Please! Someone help. I even reinstalled the
machine with asterisk 1.2.10 stable. Same problem. This config worked fine on
my previous machine.
Thanks.
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]
Skickat: den 2 augusti 2006 13:25
Till: asterisk-users@lists.digium.com
Ämne: 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 call gets answered and processed all the way to exten =
1,n,Queue(1000|tTn|||300). But it seems like the queue doesn't pick up the
call, nothing happens. No MoH, no nothing, just silence. Caller is not sent to
free agent. It just hangs there.
Example of incomming call that gets sent to queue (from extensions_custom.conf):
exten = 1,1,Macro(custom-callerid,${CALLERIDNUM},SPARR)
exten = 1,n,Wait(2)
exten = 1,n,Set(QUEUE_PRIO=10)
exten =
1,n,Set(MONITOR_FILENAME=/new/monitor/queues/${TIMESTAMP}-${CALLERID(name)}-${CALLERID(num)}-${UNIQUEID})
exten = 1,n,Queue(1000|tTn|||300)
exten = 1,n,Macro(failover-alarm,SPARR,custom-incoming-3000,1,4)
My queues_custom.conf:
[1000]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60
member=Agent/1001 ; Agent 1001
[1001]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60
member=Agent/1001 ; Agent 1001
[1002]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60
member=Agent/1008 ; Agent 1008
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]
Skickat: den 2 augusti 2006 11:52
Till: asterisk-users@lists.digium.com
Ämne: 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
; This is using the new 1.2 config file format, and will not work with 1.0 ;
based Asterisk systems ; [default] mode=files
directory=/var/lib/asterisk/mohmp3
#include musiconhold_additional.conf
Thanks very much for your time!
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Mojo with Horan
Company, LLC
Skickat: den 1 augusti 2006 23:20
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] Help debugging strange asterisk behaviour
Are you using mpg123 for MoH or native? What's in your musiconhold.conf?
[EMAIL PROTECTED] wrote:
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 calls are not hung up from Zap-side
3) Flash Operator Panel starts showing all kind of random letters.
4) Agents are unable to login/logout.
..and so on. But the strange thing is that some things seem to work
perfectly fine as usual. Inbound calls are getting playbacks() but no
MoH when sent to queue, and caller is not sent to an agent. Outgoing
sip and zap calls work fine (until all zapchans are filled because of
the above hangup problem which is NOT consistent).
I've tried to debug the asterisk
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] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]
Skickat: den 4 augusti 2006 12:20
Till: asterisk-users@lists.digium.com
Ämne: 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 previous e-mail. Please! Someone help. I even reinstalled the
machine with asterisk 1.2.10 stable. Same problem. This config worked fine on
my previous machine.
Thanks.
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]
Skickat: den 2 augusti 2006 13:25
Till: asterisk-users@lists.digium.com
Ämne: 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 call gets answered and processed all the way to exten =
1,n,Queue(1000|tTn|||300). But it seems like the queue doesn't pick up the
call, nothing happens. No MoH, no nothing, just silence. Caller is not sent to
free agent. It just hangs there.
Example of incomming call that gets sent to queue (from extensions_custom.conf):
exten = 1,1,Macro(custom-callerid,${CALLERIDNUM},SPARR)
exten = 1,n,Wait(2)
exten = 1,n,Set(QUEUE_PRIO=10)
exten =
1,n,Set(MONITOR_FILENAME=/new/monitor/queues/${TIMESTAMP}-${CALLERID(name)}-${CALLERID(num)}-${UNIQUEID})
exten = 1,n,Queue(1000|tTn|||300)
exten = 1,n,Macro(failover-alarm,SPARR,custom-incoming-3000,1,4)
My queues_custom.conf:
[1000]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60
member=Agent/1001 ; Agent 1001
[1001]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60
member=Agent/1001 ; Agent 1001
[1002]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60
member=Agent/1008 ; Agent 1008
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]
Skickat: den 2 augusti 2006 11:52
Till: asterisk-users@lists.digium.com
Ämne: 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
; This is using the new 1.2 config file format, and will not work with 1.0 ;
based Asterisk systems ; [default] mode=files
directory=/var/lib/asterisk/mohmp3
#include musiconhold_additional.conf
Thanks very much for your time!
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Mojo with Horan
Company, LLC
Skickat: den 1 augusti 2006 23:20
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] Help debugging strange asterisk behaviour
Are you using mpg123 for MoH or native? What's in your musiconhold.conf?
[EMAIL PROTECTED] wrote:
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 calls are not hung up from
[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
e)}-${CALLERID(num)}-${UNIQUEID})
I saw it was possible to add a line after that that told asterisk to
encode an mp3 and remove the wav using lame. But now I can't find the
damn link.
Thanks very much for your time.
Regards,
Jan
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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 ; This is using the new 1.2 config file format, and will not work with 1.0 ; based Asterisk systems ; [default] mode=files directory=/var/lib/asterisk/mohmp3 #include musiconhold_additional.conf Thanks very much for your time! Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Mojo with Horan Company, LLC Skickat: den 1 augusti 2006 23:20 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Help debugging strange asterisk behaviour Are you using mpg123 for MoH or native? What's in your musiconhold.conf? [EMAIL PROTECTED] wrote: 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 calls are not hung up from Zap-side 3) Flash Operator Panel starts showing all kind of random letters. 4) Agents are unable to login/logout. ..and so on. But the strange thing is that some things seem to work perfectly fine as usual. Inbound calls are getting playbacks() but no MoH when sent to queue, and caller is not sent to an agent. Outgoing sip and zap calls work fine (until all zapchans are filled because of the above hangup problem which is NOT consistent). I've tried to debug the asterisk log but there are NO ERRORS! I have asterisk installed on a Dell 2850 server with dual Xeon CPU's. I'm running CentOS 4.3 x86_64 and asterisk SVN-branch-1.2-r38611M with freepbx-2.1.1 ontop of it all. I would really appreciate some thoughts on this. Please ask me for furhter info if needed since I'm no debugger. It's a hell of a task to reinstall the whole server so I'd like to know what went wrong this time first. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:500,44cf6f0c41131882367086! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 call gets answered and processed all the way to exten =
1,n,Queue(1000|tTn|||300). But it seems like the queue doesn't pick up the
call, nothing happens. No MoH, no nothing, just silence. Caller is not sent to
free agent. It just hangs there.
Example of incomming call that gets sent to queue (from extensions_custom.conf):
exten = 1,1,Macro(custom-callerid,${CALLERIDNUM},SPARR)
exten = 1,n,Wait(2)
exten = 1,n,Set(QUEUE_PRIO=10)
exten =
1,n,Set(MONITOR_FILENAME=/new/monitor/queues/${TIMESTAMP}-${CALLERID(name)}-${CALLERID(num)}-${UNIQUEID})
exten = 1,n,Queue(1000|tTn|||300)
exten = 1,n,Macro(failover-alarm,SPARR,custom-incoming-3000,1,4)
My queues_custom.conf:
[1000]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60
member=Agent/1001 ; Agent 1001
[1001]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60
member=Agent/1001 ; Agent 1001
[1002]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60
member=Agent/1008 ; Agent 1008
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]
Skickat: den 2 augusti 2006 11:52
Till: asterisk-users@lists.digium.com
Ämne: 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
; This is using the new 1.2 config file format, and will not work with 1.0 ;
based Asterisk systems ; [default] mode=files
directory=/var/lib/asterisk/mohmp3
#include musiconhold_additional.conf
Thanks very much for your time!
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Mojo with Horan
Company, LLC
Skickat: den 1 augusti 2006 23:20
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] Help debugging strange asterisk behaviour
Are you using mpg123 for MoH or native? What's in your musiconhold.conf?
[EMAIL PROTECTED] wrote:
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 calls are not hung up from Zap-side
3) Flash Operator Panel starts showing all kind of random letters.
4) Agents are unable to login/logout.
..and so on. But the strange thing is that some things seem to work
perfectly fine as usual. Inbound calls are getting playbacks() but no
MoH when sent to queue, and caller is not sent to an agent. Outgoing
sip and zap calls work fine (until all zapchans are filled because of
the above hangup problem which is NOT consistent).
I've tried to debug the asterisk log but there are NO ERRORS!
I have asterisk installed on a Dell 2850 server with dual Xeon CPU's.
I'm running CentOS 4.3 x86_64 and asterisk SVN-branch-1.2-r38611M with
freepbx-2.1.1 ontop of it all.
I would really appreciate some thoughts on this. Please ask me for
furhter info if needed since I'm no debugger. It's a hell of a task
to reinstall the whole server so I'd like to know what went wrong this
time first.
Regards,
Jan
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
!DSPAM:500,44cf6f0c41131882367086!
--
[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 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone=se defaultzone=se Zapata.conf [channels] language=se context=from-pstn switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callerid=asreceived usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=400 rxgain=-1.0 txgain=-1.5 group=0 callgroup=1 pickupgroup=1 immediate=no overlapdial=no channel = 1-15,17-31,32-46,48-62 group=1 channel = 94-108,110-124 group=2 context=from-internal signalling=pri_net channel = 63-77,79-93 Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 calls are not hung up from Zap-side 3) Flash Operator Panel starts showing all kind of random letters. 4) Agents are unable to login/logout. ..and so on. But the strange thing is that some things seem to work perfectly fine as usual. Inbound calls are getting playbacks() but no MoH when sent to queue, and caller is not sent to an agent. Outgoing sip and zap calls work fine (until all zapchans are filled because of the above hangup problem which is NOT consistent). I've tried to debug the asterisk log but there are NO ERRORS! I have asterisk installed on a Dell 2850 server with dual Xeon CPU's. I'm running CentOS 4.3 x86_64 and asterisk SVN-branch-1.2-r38611M with freepbx-2.1.1 ontop of it all. I would really appreciate some thoughts on this. Please ask me for furhter info if needed since I'm no debugger. It's a hell of a task to reinstall the whole server so I'd like to know what went wrong this time first. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 in zapata.conf to rxgain=-1.0 txgain=-1.5 Regards, Jan -Ursprungligt meddelande- Från: Jan Sarin Skickat: den 1 augusti 2006 17:12 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: 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 calls are not hung up from Zap-side 3) Flash Operator Panel starts showing all kind of random letters. 4) Agents are unable to login/logout. ..and so on. But the strange thing is that some things seem to work perfectly fine as usual. Inbound calls are getting playbacks() but no MoH when sent to queue, and caller is not sent to an agent. Outgoing sip and zap calls work fine (until all zapchans are filled because of the above hangup problem which is NOT consistent). I've tried to debug the asterisk log but there are NO ERRORS! I have asterisk installed on a Dell 2850 server with dual Xeon CPU's. I'm running CentOS 4.3 x86_64 and asterisk SVN-branch-1.2-r38611M with freepbx-2.1.1 ontop of it all. I would really appreciate some thoughts on this. Please ask me for furhter info if needed since I'm no debugger. It's a hell of a task to reinstall the whole server so I'd like to know what went wrong this time first. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 number of callers in the *current* queue? If yes, is it possible for asterisk to calculate on a group of queues? Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 queue-thereare = queue-thereare ..but no settings for one, two, three and so on. How do I do this? Do I have to overwrite the default files (which I don't want to do)? Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Skickat: den 19 juli 2006 18:12 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Queue hold position in other language? Location of the sound files Asterisk normally looks for a sound file with an extension used for the codec used. If a language is set for the channel with the SetLanguage() application, Asterisk first looks for countrycode/filename where countrycode is the language code (example:. 'fr' for french). Languages and special tones for that country or region are defined in indications.conf. http://www.voip-info.org/wiki-Asterisk%20sound%20files Hope it helps On 7/19/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: 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 queue-thereare = queue-thereare ..but no settings for one, two, three and so on. How do I do this? Do I have to overwrite the default files (which I don't want to do)? Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 anyone has had this experience - please share! We need to upgrade really quick and I need to know as possible. OR if you could recomend ANOTHER common server for in the $1000-$3000 pricerange that can handle minimum 50 simultaneous SIP(ulaw)-Zap calls. Ps. If someone wants a low-budget server for asterisk that can handle ~10-20 simultaneous SIP-Zap calls (depending on what you do) I can tell you that there's absolutly no problem running asterisk 1.2.1 with a TE405P card on a PowerEdge 600SC with 1-2 GB of ram. With more than 20 calls we're experiencing sound drops... Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Idris AVCI Skickat: den 5 juli 2006 10:56 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: RE: [Asterisk-Users] HP Proliant server? Check this on : http://www.digium.com/en/docs/misc/compatibility_notes.php -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 05, 2006 11:48 AM To: asterisk-users@lists.digium.com Subject: [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, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Users Mailing List - Non-Commercial Discussion Ämne: Re: [Asterisk-Users] HP Proliant server? On Wed, 2006-07-05 at 10:48 +0200, [EMAIL PROTECTED] wrote: 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? I'm running Asterisk on an HP Proliant DL360p. But no Digium hardware, this is a SIP only setup. Luca ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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 and digium hardware would be appreciated. Must handle minimum 50 SIP(ulaw)-Zap calls. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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 simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio tracks on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 active calls (too much 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 simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio tracks on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: [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 active calls (too much 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 simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio tracks on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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" when many SIP-clients where active in a call at the same time. Computer RAM is 2 gb. If the E1 is channelized or not I don't actually know. How would I know this and why would it affect the call quality when many people are in a call at the same time (same lines work fine with an Ericsson BusinessPhone Exchange)? Thanks! Regards, Jan Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]Skickat: den 4 juli 2006 15:55Till: Asterisk Users Mailing List - Non-Commercial DiscussionÄmne: Re: SV: [Asterisk-Users] Running 40 active calls (too much för CPU?) Are the phones behind a NAT? What is the processory memory size? Are the E1 channelized? -- Original message -- From: [EMAIL PROTECTED] 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 active calls (too much 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 BR ; of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio "tracks" on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: h ttp:// lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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" when many SIP-clients where active in a call at the same time. Computer RAM is 2 gb. If the E1 is channelized or not I don't actually know. How would I know this and why would it affect the call quality when many people are in a call at the same time (same lines work fine with an Ericsson BusinessPhone Exchange)? Thanks! Regards, Jan Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]Skickat: den 4 juli 2006 15:55Till: Asterisk Users Mailing List - Non-Commercial DiscussionÄmne: Re: SV: [Asterisk-Users] Running 40 active calls (too much för CPU?) Are the phones behind a NAT? What is the processory memory size? Are the E1 channelized? -- Original message -- From: [EMAIL PROTECTED] 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 active calls (too much 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 BR ; of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio "tracks" on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: h ttp:// lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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 they will cost when released? Thanks! http://pressroom.pulvermedia.com/digium/pr.php#0314c Regards, jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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] För El Flynn Skickat: den 24 maj 2006 10:17 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [Asterisk-Users] USB headsets? [EMAIL PROTECTED] wrote: 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. Well, IMO if the soundcards are already crap to start out with, there's no way a fancy-schmancy USB headset -- or any other headset, for that matter -- will sound good when plugged in to the laptop. Because, remember, it's the soundcard that generates the audio and sends it out the heaphone port. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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... What model headsets (name/number) do you have? Any recomendations on the USB audio-devices? Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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@lists.digium.com Ämne: 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 PROTECTED] För Jerry Jones Skickat: den 4 maj 2006 15:00 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [Asterisk-Users] Polycom 501 - Disable DND feature? Attribute Values Default Interpretation call.rejectBusyOnDnd 0, 1 1 If set to 1, reject all incoming calls with the reason busy if do-not-disturb is enabled. Have not used, but looks like it may ignore the key if this is 0 Let us know... On May 4, 2006, at 2:22 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: 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: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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: http://lists.digium.com/mailman/listinfo/asterisk-users
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 PROTECTED] För Jerry JonesSkickat: den 4 maj 2006 15:00Till: Asterisk Users Mailing List - Non-Commercial DiscussionÄmne: Re: [Asterisk-Users] Polycom 501 - Disable DND feature? Attribute Values Default Interpretation call.rejectBusyOnDnd 0, 1 1 If set to 1, reject all incoming calls with the reason busy if do-not-disturb is enabled. Have not used, but looks like it may ignore the key if this is 0 Let us know... On May 4, 2006, at 2:22 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: 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: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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 AgentCallbackLogin, but the phone itself was unreacable because of the internet connection problem. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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 trying to uninstall with rpm -e libtiff I get the following error: error: Failed dependencies: libtiff.so.3 is needed by (installed) cups-libs-1.1.22-0.rc1.9.8.i386 libtiff = 3.6.1 is needed by (installed) libtiff-devel-3.6.1-8.i386 I didn't find anything on google on this so I thought you guys might be able to help me. Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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 options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 this is actually working or not? Has anyone actually used the Polycom ACD-login function succesfully with asterisk? How did you do it? Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För BJ Weschke Skickat: den 23 februari 2006 13:44 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [Asterisk-Users] Polycom 501 ACDlogin On 2/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: 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 this is actually working or not? Has anyone actually used the Polycom ACD-login function succesfully with asterisk? How did you do it? It is indeed functional. It's in production at the present time at one of our call center clients. I believe there is a README in the docs of the branch that outlines setup of the functionality. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: den 23 februari 2006 16:13 Till: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Ämne: RE: [Asterisk-Users] Polycom 501 ACDlogin You don't need the Polycom ACD support in order to do ACD logins with Polycom phones. Just dial an extension and call AgentCallBackLogin(). You won't get any visual confirmation on the phone however of being logged in, but you will be. If you set the acd-login fields in the phone's xml, the phone will come up and log the agent in automatically. At least that's what I've found. Doug. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, February 23, 2006 7:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom 501 ACDlogin Can we patch the stable release with your SVN branch ? Regards Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : On 2/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: 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 meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För BJ Weschke Skickat: den 23 februari 2006 13:44 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [Asterisk-Users] Polycom 501 ACDlogin On 2/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: 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 this is actually working or not? Has anyone actually used the Polycom ACD-login function succesfully with asterisk? How did you do it? It is indeed functional. It's in production at the present time at one of our call center clients. I believe there is a README in the docs of the branch that outlines setup of the functionality. Are you using the code of Asterisk provided in the SVN branch for this feature? -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: RE: [Asterisk-Users] BAD/GOOD Echo Cancel virtually all software echo cancelers cannot get double echo removed completly. It can get the first one but not the second one. There are instances where you get a 2nd echo, so ... Asterisk is no exception from this afaik nothing software only based is. If you really want good echo cancelation a hardware solution is the way to go. Just an enquiring mind wanting to know, but how is a hardware solution different to a software solution? The echo cancellers in the Digium hardware presumably just use the same sort of algorithms as the software versions, so it is just that they are dedicated and perform better, that they are closer to the source of the echo, or some other thing that I've overlooked? Thanks james ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 no good help on wiki and voip-info.org, I've gone through it already. Zach -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Monday, February 06, 2006 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help on queues Yes. The wiki and voip-info.org --- Zach A [EMAIL PROTECTED] wrote: Hi, Is there any detailed guide/tutorial source online on queues? Zach ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 februari 2006 12:37 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: [Asterisk-Users] Outbound Caller ID number on E1 Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 of time I want Asterisk to delay answering the POTS line via a Wildcard (a Zap channel) by some period of time, either a number of rings or just a number of seconds. I have tried this: [from-pots] exten = s,1,Wait(30) exten = s,n,Answer ... exten = s,n,Dial(SIP/brianSIP/joe,10,H) exten = s,n,Voicemail(u2001) exten = s,n,Hangup exten = s,103,Voicemail(u2001) exten = s,104,Hangup exten = h,1,Hangup exten = i,1,Hangup but that doesn't work. It seems that as soon as the first or second ring is detected, Asterisk has decided it will answer the line, it just waits 30 seconds to do it. The problem I have is that the POTS line that Asterisk is on is shared by handsets, and if somebody picks up a handset soon enough, I don't want Asterisk to pick up the line. Yeah, Asterisk as an answering machine... until I can get at least one FXS interface, anyway. Any ideas? b. -- My other computer is your Microsoft Windows server. Brian J. Murrell winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 related... if I've understood your question and the answer I found correctly. Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Brian J. Murrell Skickat: den 2 februari 2006 22:37 Till: asterisk-users@lists.digium.com Ämne: Re: SV: [Asterisk-Users] delaying answer for a number of ringsor anamount of time On Thu, 2006-02-02 at 22:08 +0100, [EMAIL PROTECTED] wrote: http://lists.digium.com/pipermail/asterisk-users/2005-September/125146 .html OK. The hardware is a wildcard though. How does that answer apply? Isn't it asterisk itself that is picking that call up? Can't it delay the pick up? Maybe I am just misunderstanding your reference. b. -- My other computer is your Microsoft Windows server. Brian J. Murrell ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Discussion
Ämne: Re: [Asterisk-Users] Re: CallerID Problem
No, I'm not including the -- I was trying to show that it was
something that I removed from my example..
Thanks.
On 2/1/06, Bromont Quebec [EMAIL PROTECTED] wrote:
Are you actually putting the in there?
try:
exten = _9.,1,Set(CALLERID(number)=MAINNUMBER)
Hey,
I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
connects to it using SIP. The asterisk version is 1.2.0.
In my sip.conf, I set callerid=First Last exten
When I make a an outbound call with the following macro:
exten = _9.,1,Dial(SIP/${EXTEN}@ROUTER,,w)
exten = _9.,2,Congestion()
The caller id is set to the extension that's defined in sip.conf.
If I try something like:
exten = _9.,1,Set(CALLERID(number)=MAINNUMBER)
exten = _9.,2,Dial(SIP/${EXTEN}@ROUTER,,w)
exten = _9.,3,Congestion()
I get the following error:
-- Got SIP response 488 Not Acceptable Media back from ROUTER
It all works fine if I don't set the caller id.. Any ideas on why this
may be happening?
Thanks.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
winmail.dat___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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 and when the Session
Description Protocol (SDP) for the incoming Invite message does not have any
FMTP attribute line, which means that the default value, that is, the G279B
(annex B) codec, is used.
Workaround: There is no workaround.
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] genom Gary Richardson
Skickat: on 2006-02-01 21:45
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [Asterisk-Users] Re: CallerID Problem
No, I'm not including the -- I was trying to show that it was
something that I removed from my example..
Thanks.
On 2/1/06, Bromont Quebec [EMAIL PROTECTED] wrote:
Are you actually putting the in there?
try:
exten = _9.,1,Set(CALLERID(number)=MAINNUMBER)
Hey,
I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
connects to it using SIP. The asterisk version is 1.2.0.
In my sip.conf, I set callerid=First Last exten
When I make a an outbound call with the following macro:
exten = _9.,1,Dial(SIP/${EXTEN}@ROUTER,,w)
exten = _9.,2,Congestion()
The caller id is set to the extension that's defined in sip.conf.
If I try something like:
exten = _9.,1,Set(CALLERID(number)=MAINNUMBER)
exten = _9.,2,Dial(SIP/${EXTEN}@ROUTER,,w)
exten = _9.,3,Congestion()
I get the following error:
-- Got SIP response 488 Not Acceptable Media back from ROUTER
It all works fine if I don't set the caller id.. Any ideas on why this
may be happening?
Thanks.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
winmail.dat___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Set caller id on Swedish PRI (euroisdn)
Try setting the Callerpresentation to something else: http://www.voip-info.org/wiki/page_history.php?page_id=1682preview=2 SetCallerPres(prohib) actually worked! Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Set caller id on Swedish PRI (euroisdn)
Hi,
I have a problem with setting outgoing caller id to nothing (secret)
on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID
seems to work fine when connecting the same line to a Ericsson PBX - so
something must be wrong in my settings, but I don't know what.
I've tried:
exten = _*70X.,1,Set(CALLERID(name)=) exten =
_*70X.,2,Set(CALLERID(num)=) exten =
_*70X.,3,Dial(Zap/g0/${EXTEN:3}|60|T)
But the result is always that the caller id is our main number
(A-number).
Here is an from zapata.conf:
[channels]
language=se
context=from-pstn
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=400
rxgain=1.0
txgain=-4.0
group=0
callgroup=1
pickupgroup=1
immediate=no
overlapdial=no
channel = 1-15,17-31,63-77,79-93
group=1
channel = 94-108,110-124
group=2
context=from-internal
signalling=pri_net
channel = 32-46,48-62
Regards,
Jan
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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 direcly to the asterisk with the queues. Autologoff is set to 20 seconds in agents.conf. The odd thing is that when the queue calls an agent over trunk, it rings for 30 seconds and then recieves a congestion (yes, I've set ring time to 30 secs). Shouldn't it ring for 20 seconds, log off the agent, and try next agent (which it actually does with agents that are not called over trunk)? Anyone had the same problem? Thanks, Jan / Sweden ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 recieves calls from the queue-problem? Thanks! Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Adam Goryachev Skickat: den 18 oktober 2005 15:08 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: SV: [Asterisk-Users] Queues and call waiting indication On Tue, 2005-10-18 at 14:35 +0200, [EMAIL PROTECTED] wrote: 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 agent if it has call waiting enabled. This behaviour is totally senseless since the whole purouse of queues is to _queue_ the callers until the agent is available. available usually means not on the phone -- whether or not it's an incoming or outgoing call. I solved this problem by using single-line clients and phones where you can turn off call wating. Actually this can simply be solved in your dialplan Just use the setgroup/checkgroup values, and use the AgentCallbackLogin instead of AgentLogin This is what I used, and it seems to work quite well so far... well, I haven't actually added the bits for the outbound calls yet on my own system, but I've done it on others, and they seem to be quite happy with it... Regards, Adam ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 agent if it has call waiting enabled. This behaviour is totally senseless since the whole purouse of queues is to _queue_ the callers until the agent is available. available usually means not on the phone -- whether or not it's an incoming or outgoing call. I solved this problem by using single-line clients and phones where you can turn off call wating. //Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 18 oktober 2005 14:14 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Queues and call waiting indication Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a beep beep (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a single line user agent, like firefly, still this behaviour does not make any sense to me. I tried using incominglimit and outgoinglimit in my sip.conf, even if they are deprecated: no luck. Here is a sample from my queues.conf, something wrong in my setup maybe ? Tnx for any help! [ingombranti] joinempty = strict maxlen=3 musiconhold = default announce = annuncio-ingombranti strategy = rrmemory servicelevel = 60 timeout = 15 announce-frequency = 15 eventwhencalled = yes member=SIP/401 member=SIP/402 member=SIP/403 member=SIP/404 member=SIP/405 member=SIP/406 member=SIP/407 member=SIP/408 member=SIP/409 member=SIP/410 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 - Non-Commercial Discussion Ämne: Re: SV: [Asterisk-Users] Queues and call waiting indication This behaviour is totally senseless since the whole purouse of queues is to _queue_ the callers until the agent is available. available usually means not on the phone -- whether or not it's an incoming or outgoing call. Agree! I solved this problem by using single-line clients and phones where you can turn off call wating. Can you suggest me a SIP or IAX phone with just one line that can also open url's passed by asterisk ? Tnx! //Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 18 oktober 2005 14:14 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Queues and call waiting indication Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a beep beep (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a single line user agent, like firefly, still this behaviour does not make any sense to me. I tried using incominglimit and outgoinglimit in my sip.conf, even if they are deprecated: no luck. Here is a sample from my queues.conf, something wrong in my setup maybe ? Tnx for any help! [ingombranti] joinempty = strict maxlen=3 musiconhold = default announce = annuncio-ingombranti strategy = rrmemory servicelevel = 60 timeout = 15 announce-frequency = 15 eventwhencalled = yes member=SIP/401 member=SIP/402 member=SIP/403 member=SIP/404 member=SIP/405 member=SIP/406 member=SIP/407 member=SIP/408 member=SIP/409 member=SIP/410 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
