Re: [asterisk-users] The downside of Asterisk and least cost routing...
Chris Mason (Lists) wrote: The only thing I'd probably lose is the ability to do faxes! So I am going to investigate that further first! Havn't doen that in years - an online fax service sends me my faxes by email and I sent out faxes through them, not that I ever do that. The fax-to-e-mail services charge as much as the telco does for a business line, sometimes more (at least, the ones I can deal with in this area). Better to set-up hylafax, IMHO. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] function_db_read: DB requires an argument, DB(family/key)
Lee Jenkins wrote:
OK, so I tried this:
exten = _X.,1,Noop(CallerId is ${CALLERID(all)})
exten = _X.,n,Noop(blurp)
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
This now appears to execute the first Noop(), skip the second, and
then issue the no argument warning on the Set() call.
Try an Answer() first?
OK, tried that, didn't change anything.
What I still don't get is - why does reloading the app_queue module fix
this problem? The app_queue issue is another one, but I just can't see
how it would influence the workings of the DB() function.
/Per Jessen, Zürich
--
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[asterisk-users] Segmentation fault
Asterisk is crashing about once a day with segmentation fault.
This is the error..
/usr/sbin/safe_asterisk: line 111: 3482 Segmentation fault (core
dumped) nice –n $PRIORITY ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS}
/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
Is this information helpful?
Can anyone suggest anything?
Can I provide anymore useful information for troubleshooting?
Thanks!
Adam
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[asterisk-users] Re: Segmentation fault
Sorry, part of the email got chopped of.
Hardware specs are:
AMD A64/3500+ CPU: Desktop Athlon64
Asus A8N-SLI Deluxe Athlon™ 64 S939 NVIDIA nForce(r)4 SLI™ PCI
Corsair TWINX1024-3200XL 2x 512MB 1024MB Dual Channel Optimised Pair
Western Digital WD1200JD/PD-SATA 120GB Serial ATA 150Mbps 7200rpm HDD 8Mb Cache
Leadtek PX6200 TC TDH GeForce 6200 64MB PCI-E
LG GDR-8163B DVD-ROM Internal: 16x
Digium TDM400P 4x FXO
Software:
centos 4.4 32 bit
Asterisk 1.4.4
Zaptel 1.4.2.1
codec_g729a_v31_athlon
On 5/16/07, Adam Lovegrove [EMAIL PROTECTED] wrote:
Asterisk is crashing about once a day with segmentation fault.
This is the error..
/usr/sbin/safe_asterisk: line 111: 3482 Segmentation fault (core
dumped) nice –n $PRIORITY ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS}
/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
Is this information helpful?
Can anyone suggest anything?
Can I provide anymore useful information for troubleshooting?
Thanks!
Adam
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RE: [asterisk-users] The downside of Asterisk and least cost routing...
On Tue, 2007-05-15 at 22:29 +0200, Francesco Peeters (Asterisk) wrote: In NL you actually can ditch the telephony and keep the ADSL... My ISP even gives emergency access if you transfer your main number to their SIP service. Here in France you can also move to ADSL only, what I found really interesting was that you can even do that when your ISP is Orange a division of France Telecom the company that will loose the telephony business. It's not too well advertised though. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmentation fault
Adam Lovegrove wrote:
Asterisk is crashing about once a day with segmentation fault.
This is the error..
/usr/sbin/safe_asterisk: line 111: 3482 Segmentation fault (core
dumped) nice –n $PRIORITY ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS}
/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
Is this information helpful?
Can anyone suggest anything?
Can I provide anymore useful information for troubleshooting?
I think it would be useful if you could describe what the system is
doing when it crashes - if you know. Also, the core dump will probably
help someone diagnose the problem. (but don't send it to the list :-)
/Per Jessen, Zürich
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Re: [asterisk-users] The downside of Asterisk and least cost routing...
On Wed, 16 May 2007, Stephen Bosch wrote: Chris Mason (Lists) wrote: The only thing I'd probably lose is the ability to do faxes! So I am going to investigate that further first! Havn't doen that in years - an online fax service sends me my faxes by email and I sent out faxes through them, not that I ever do that. The fax-to-e-mail services charge as much as the telco does for a business line, sometimes more (at least, the ones I can deal with in this area). Better to set-up hylafax, IMHO. !!! They can be had for free (or sometimes a one-off setup fee, usually less than a tenner) in the UK. The numbers assigned, while not premium numbers are charged to the caller at local or national rates, and there is a tiny kick-back to the provider from the fees which covers running costs, depending on the number assigned (ie. 0845 or 0870 - but there's a big resistance to 0870 numbers in the UK - see www.saynoto0870.co.uk) You can pay more for fancy numbers - eg. ones that map to names on the dial-pad. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] Asterisk is not showing the correctIncomming CallerID
Yes national and internation prefix was 1 and +1 there. Thank you very much Francois BERGERET. Regards Farooq Quoting [EMAIL PROTECTED]: Hi Farook and the list, You have may be forgotten to input that in the misdn.conf file : nationalprefix=0 internationalprefix=00 dialplan=0 localdialplan=0 cpndialplan=0 Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Farooq Ahmed Envoyé : mercredi 16 mai 2007 06:14 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Asterisk is not showing the correctIncomming CallerID I forgot to give the asterisk logs pbx*CLI -- Executing Set(mISDN/2-2, FROM_DID=3722) in new stack -- Executing Gosub(mISDN/2-2, app-blacklist-check|s|1) in new stack -- Executing LookupBlacklist(mISDN/2-2, ) in new stack -- Executing GotoIf(mISDN/2-2, 0?blacklisted) in new stack -- Executing Return(mISDN/2-2, ) in new stack -- Executing Goto(mISDN/2-2, ext-group|1|1) in new stack -- Goto (ext-group,1,1) -- Executing Macro(mISDN/2-2, user-callerid|) in new stack -- Executing NoOp(mISDN/2-2, user-callerid: 1416222888) in new stack -- Executing GotoIf(mISDN/2-2, 0?report) in new stack -- Executing GotoIf(mISDN/2-2, 0?start) in new stack -- Executing Set(mISDN/2-2, REALCALLERIDNUM=1416222888) in new stack -- Executing NoOp(mISDN/2-2, REALCALLERIDNUM is 1416222888) in new stack -- Executing Set(mISDN/2-2, AMPUSER=) in new stack -- Executing Set(mISDN/2-2, AMPUSERCIDNAME=) in new stack -- Executing GotoIf(mISDN/2-2, 1?report) in new stack -- Goto (macro-user-callerid,s,11) -- Executing NoOp(mISDN/2-2, TTL: ARG1: ) in new stack -- Executing GotoIf(mISDN/2-2, 0?continue) in new stack -- Executing Set(mISDN/2-2, _TTL=64) in new stack -- Executing GotoIf(mISDN/2-2, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(mISDN/2-2, Using CallerID 1416222888) in new stack -- Executing Set(mISDN/2-2, modifiedcallerid=1416222888) in new stack -- Executing Set(mISDN/2-2, CALLERID(number)=1416222888) in new stack -- Executing GotoIf(mISDN/2-2, 1?skipdb) in new stack -- Goto (ext-group,1,4) -- Executing Set(mISDN/2-2, __NODEST=) in new stack -- Executing Set(mISDN/2-2, __BLKVM_OVERRIDE=BLKVM/1/mISDN/2-2) in new stack -- Executing Set(mISDN/2-2, __BLKVM_BASE=1) in new stack -- Executing Set(mISDN/2-2, DB(BLKVM/1/mISDN/2-2)=TRUE) in new stack -- Executing Set(mISDN/2-2, RRNODEST=) in new stack -- Executing Set(mISDN/2-2, __NODEST=1) in new stack -- Executing GotoIf(mISDN/2-2, 1?REPCID) in new stack -- Goto (ext-group,1,14) -- Executing NoOp(mISDN/2-2, CALLERID(name) is ) in new stack -- Executing Set(mISDN/2-2, RecordMethod=Group) in new stack -- Executing Macro(mISDN/2-2, record-enable|903-909|Group) in new stack -- Executing GotoIf(mISDN/2-2, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI(mISDN/2-2, recordingcheck|20070516-140757|1179288477.1037) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(mISDN/2-2, No recording needed) in new stack -- Executing Set(mISDN/2-2, RingGroupMethod=hunt) in new stack -- Executing Macro(mISDN/2-2, dial|10||903-909) in new stack -- Executing DeadAGI(mISDN/2-2, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'unknown' number is '1416222888' dialparties.agi: Methodology of ring is 'hunt' dialparties.agi: USE_CONFIRMATION: 'FALSE' dialparties.agi: RINGGROUP_INDEX: '' -- dialparties.agi: Added extension 903 to extension map -- dialparties.agi: Added extension 909 to extension map -- dialparties.agi: Extension 903 cf is disabled -- dialparties.agi: Extension 909 cf is disabled -- dialparties.agi: Extension 903 do not disturb is disabled -- dialparties.agi: Extension 909 do not disturb is disabled dialparties.agi: extnum: 903 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: exthascfu: 0 dialparties.agi: extcfu: dialparties.agi: extnum: 909 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: exthascfu: 0 dialparties.agi: extcfu: dialparties.agi: NODEST: 1 adding M(auto-blkvm) to dialopts: M(auto-blkvm) -- AGI Script
[asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4
Hi, has anyone managed to get hudlite server working on a Debian Etch based installation of Asterisk 1.4? So far I managed to eliminate all error messages, but the process is killed directly after starting the hudlite server without showing any error messages. I would be very happy if anyone can give me some hints or point me to a installation guide. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
aaah... I'm running asterisk in a Xen kernel, but not on a virtual machine (DomU), only on Dom0, so it's supposed to be running on the physical server (no PCI frontend device, ...). I had seen possible problems with older versions of Xen, but only with ztdummy timing and on DomU virtual machines. On Xen mailing list, they all advise to use a digium PCI card to remove those problems. I will test and report what happens with a normal kernel, but meanwhile doesn't anyone know of a possible possibility to make it work with this setting playing for example with IRQ priorities or something, or isn't there any hope at all? Thanks again, François. Stephen Bosch wrote: François Delawarde wrote: Thanks Michael, I've already been through all that unfortunately, and I have a SATA drive, so no UDMA mode 2 as far as I know. I'm currently trying everything again anyway, but i doubt it will work if nothing worked the first time. Anyone would know of issues with XEN or SMP (or both) kernel? Do dual core AMD64 processors have issues? Uh, yeah... Xen has many, many problems with interrupt handling and is utterly unsuitable for running anything that depends on hardware peripherals. I speak from very painful experience. There is no way, under any circumstance, that I would try to run Asterisk with interface cards in a Xen environment. It's too bad you wasted so much time trying to fix it -- it's never going to work. Try ripping Xen out and doing it directly on the physical server. I think you'll find your problems will go away. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write data to astdb?
On Wednesday 16 May 2007 03:47, C F wrote: I use asterisk -rx database put value if you are trying to batch it from windows you can use plink This will be VERY slow. Other options might be writing to the asterisk socket (I heard it's not that reliable). But again, this will be a problem on remote scenarios. What I have been using is creating a asterisk-manager connection to Asterisk, which is very reliable and fast. The downside is that you must have a user configured in manager.conf (all others do not need this, a simple root account is good enough). ___ --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] PRI got event
Hi all, I have 1 Card Digium TE412P and 2PRI E1. I have more problems with drops lines. The asterisk log is this: May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 May 16 10:52:26 WARNING[4465]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 My Interruptions are: 0: 576052004 0IO-APIC-edge timer 1: 9 0IO-APIC-edge i8042 2: 0 0 XT-PIC cascade 12: 58 0IO-APIC-edge i8042 14: 14 0IO-APIC-edge ide0 25: 38568228 0 IO-APIC-level eth0 51:1744474 0 IO-APIC-level cciss0 78: 575931860 0 IO-APIC-level wct4xxp My zaptel version is 1.2.12 and asterisk 1.2.13 zapata.conf [trunkgroups] [channels] language=es usecallerid=yes hidecallerid=yes callerid=asreceived cidsignalling=dtmf switchtype=euroisdn signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=yes callwaiting=yes immediate=no context=default rxgain=5.5 txgain=5.5 musiconhold=default faxdetect=both group = 1 context=primario channel = 1-15,17-31 channel = 32-46,48-62 zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,2,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 Thanks. _ Descubre la descarga digital con MSN Music. Más de un millón de canciones. http://music.msn.es/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The downside of Asterisk and least cost routing...
On Wed, 16 May 2007, Stephen Bosch said something to this effect: The fax-to-e-mail services charge as much as the telco does for a business line, sometimes more (at least, the ones I can deal with in this area). Better to set-up hylafax, IMHO. Not necessarily, except perhaps in cases of very high volumes. http://www.ureach.com/home3/ufax_overview.htm http://www.efax.com/en/efax/twa/productOverview -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voice recording on legacy PBX
Hi, Is it possible to use Asterisk to record or monitor all conversation on standard PSTN PBX ? ASLAY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The downside of Asterisk and least cost routing...
Stephen Bosch wrote: The fax-to-e-mail services charge as much as the telco does for a business line, sometimes more (at least, the ones I can deal with in this area). Better to set-up hylafax, IMHO. http://www.maxemail.com/fax/fax-lite.html $24/annum. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voice recording on legacy PBX
On Wed, 16 May 2007, [EMAIL PROTECTED] said something to this effect: Is it possible to use Asterisk to record or monitor all conversation on standard PSTN PBX ? I imagine with enough work you could use it as a pass-through tap by taking FXO trunks, barging into the media and then turning around and handing another leg off as FXO again, but I don't know how well it would work, scale, or, depending on the intended application, be transparent to users in terms of PBX features / not breaking things / funny sounds / echo / perceptible rx/tx gain differences, etc. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SRTP certificates
Hello all,
I want to use Asterisk with the SRTP patch from
http://bugs.digium.com/view.php?id=5413 .
I'm confused to create the certificates for it.
Can anybody help in such question?
P. S. I've created the pem files and renamed it to
* ${astetcdir}/asterisk.crt
* ${astetcdir}/asterisk.key
* ${astetcdir}/ca-certificates.crt
but the asterisk got segmentation fault error at startup.
--
Thanks in advance
Alexander Olekhnovich
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RE: [asterisk-users] voice recording on legacy PBX
Yes. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, 16 May 2007 6:06 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voice recording on legacy PBX Hi, Is it possible to use Asterisk to record or monitor all conversation on standard PSTN PBX ? ASLAY ___ --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] Re: Asterisk SRTP certificates
Thank you all, I've got it.
On 5/16/07, Alexandr Olekhnovich [EMAIL PROTECTED] wrote:
Hello all,
I want to use Asterisk with the SRTP patch from
http://bugs.digium.com/view.php?id=5413 .
I'm confused to create the certificates for it.
Can anybody help in such question?
P. S. I've created the pem files and renamed it to
* ${astetcdir}/asterisk.crt
* ${astetcdir}/asterisk.key
* ${astetcdir}/ca-certificates.crt
but the asterisk got segmentation fault error at startup.
--
Thanks in advance
Alexander Olekhnovich
--
Best Regards
Alexander Olekhnovich
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RE: [asterisk-users] voice recording on legacy PBX
Hi, Connect you Asterix box in the middle of call-flow and you will be able to record all calls. PSTN - Asterisk - Legacy PBX - Phones Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com VoIP ServicesSolutions MOR - FREE Open Source billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, May 16, 2007 1:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voice recording on legacy PBX Hi, Is it possible to use Asterisk to record or monitor all conversation on standard PSTN PBX ? ASLAY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WaitExten not responding on key presses
Hi,
I have the problem that WaitExten is not responding to key presses. Here
are the sections from my extensions.conf:
[globals]
incoming_call=0
menu=0
announce=0
[internal]
exten = 777,1,Goto(hotline,${EXTEN},1)
[hotline]
exten = _X.,1,Set(CALLERID(name)=Hotline)
exten = _X.,n,Set(original_extension=${EXTEN})
exten = _X.,n,GotoIf($[${announce}=1]?4:10)
exten = _X.,n,Answer
exten = _X.,n,NoOp(Ansage: Das Problem XYZ ist bereits bekannt und wird
bearbeitet)
exten = _X.,n,NoOp(Ansage: Druecken Sie die Taste 1 falls Sie wegen
einem anderen Problem anrufen)
exten = _X.,n,NoOp(Ansage: Ansonsten druecken Sie eine andere Taste
oder legen Sie bitte auf)
exten = _X.,n,WaitExten(5)
exten = _X.,n,Goto(18)
exten = _X.,n,Set(menu=1)
exten = _X.,n,NoOp(Ansage: Das Gespraech wird aus Qualitaetsgruenden
aufgezeichnet)
exten = _X.,n,NoOp(Ansage: Falls Sie damit nicht einverstanden sind
druecken Sie bitte die Taste 1)
exten = _X.,n,WaitExten(5)
exten = _X.,n,MixMonitor(test.wav)
exten = _X.,n,SayDigits(123)
exten = _X.,n,Queue(hotline|t|||120)
exten = _X.,n,StopMonitor()
exten = _X.,n,Hangup
exten = _[0-9],1,Goto(menu,${EXTEN},1)
exten = i,1,Goto(invalid,${EXTEN},1)
exten = t,1,Goto(timeout,${EXTEN},1)
[menu]
exten = 1,1,GotoIf($[${menu}=0]?2:4)
exten = 1,n,Set(menu=1)
exten = 1,n,Goto(hotline,${original_extension},11)
exten = 1,n,Goto(hotline,${original_extension},16)
exten = _[02-9*#],1,Hangup
The CLI output is:
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/202-b6d08708, hotline|777|1)
in new stack
-- Goto (hotline,777,1)
-- Executing [EMAIL PROTECTED]:1] Set(SIP/202-b6d08708,
CALLERID(name)=Hotline) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/202-b6d08708,
original_extension=777) in new stack
-- Executing [EMAIL PROTECTED]:3] GotoIf(SIP/202-b6d08708, 0?4:10) in
new stack
-- Goto (hotline,777,10)
-- Executing [EMAIL PROTECTED]:10] Set(SIP/202-b6d08708, menu=1) in
new stack
-- Executing [EMAIL PROTECTED]:11] NoOp(SIP/202-b6d08708, Ansage: Das
Gespraech wird aus Qualitaetsgruenden aufgezeichnet) in new stack
-- Executing [EMAIL PROTECTED]:12] NoOp(SIP/202-b6d08708, Ansage: Falls
Sie damit nicht einverstanden sind druecken Sie bitte die Taste 1) in
new stack
-- Executing [EMAIL PROTECTED]:13] WaitExten(SIP/202-b6d08708, 5) in
new stack
= here is the point where I press a digit but nothing happens:
-- Timeout on SIP/202-b6d08708, continuing...
-- Executing [EMAIL PROTECTED]:14] MixMonitor(SIP/202-b6d08708,
test.wav) in new stack
-- Executing [EMAIL PROTECTED]:15] SayDigits(SIP/202-b6d08708, 123) in
new stack
-- SIP/202-b6d08708 Playing 'digits/1' (language 'de')
== Begin MixMonitor Recording SIP/202-b6d08708
-- SIP/202-b6d08708 Playing 'digits/2' (language 'de')
-- SIP/202-b6d08708 Playing 'digits/3' (language 'de')
-- Executing [EMAIL PROTECTED]:16] Queue(SIP/202-b6d08708,
hotline|t|||120) in new stack
[May 16 11:37:00] WARNING[8400]: translate.c:163 framein: no samples for
alawtolin
-- Started music on hold, class 'default', on SIP/202-b6d08708
-- Stopped music on hold on SIP/202-b6d08708
-- User disconnected from queue hotline while waiting their turn
== Spawn extension (hotline, 777, 16) exited non-zero on
'SIP/202-b6d08708'
== End MixMonitor Recording SIP/202-b6d08708
The real strange thing is that when I change the value of the global
variable announce to 1 WaitExten is working as expected:
[globals]
incoming_call=0
menu=0
announce=1
CLI output:
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/202-081bb9f8, hotline|777|1)
in new stack
-- Goto (hotline,777,1)
-- Executing [EMAIL PROTECTED]:1] Set(SIP/202-081bb9f8,
CALLERID(name)=Hotline) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/202-081bb9f8,
original_extension=777) in new stack
-- Executing [EMAIL PROTECTED]:3] GotoIf(SIP/202-081bb9f8, 1?4:10) in
new stack
-- Goto (hotline,777,4)
-- Executing [EMAIL PROTECTED]:4] Answer(SIP/202-081bb9f8, ) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(SIP/202-081bb9f8, Ansage: Das
Problem XYZ ist bereits bekannt und wird bearbeitet) in new stack
-- Executing [EMAIL PROTECTED]:6] NoOp(SIP/202-081bb9f8, Ansage:
Druecken Sie die Taste 1 falls Sie wegen einem anderen Problem anrufen)
in new stack
-- Executing [EMAIL PROTECTED]:7] NoOp(SIP/202-081bb9f8, Ansage:
Ansonsten druecken Sie eine andere Taste oder legen Sie bitte auf) in
new stack
-- Executing [EMAIL PROTECTED]:8] WaitExten(SIP/202-081bb9f8, 5) in
new stack
== CDR updated on SIP/202-081bb9f8
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/202-081bb9f8, menu|1|1) in
new stack
-- Goto (menu,1,1)
-- Executing [EMAIL PROTECTED]:1] GotoIf(SIP/202-081bb9f8, 1?2:4) in new
stack
-- Goto (menu,1,2)
-- Executing [EMAIL PROTECTED]:2] Set(SIP/202-081bb9f8, menu=1) in new
stack
-- Executing [EMAIL PROTECTED]:3] Goto(SIP/202-081bb9f8, hotline|777|11) in
new stack
-- Goto (hotline,777,11)
-- Executing [EMAIL PROTECTED]:11]
[asterisk-users] asterisk manager interface stability
There are many past posts stating that AMI is not stable when multiple client computers are allowed to connect, particularly when connections are dropped. Has much progress been made on this? Is it more stable now than in the past? As of what versions were these issues improved? Is it feasible to connect a large number of windows computers directly to AMI for the purpose of initiating calls from software? ___ --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 client registers then unregisters
I have a remote user with Eyebeam on a laptop. Internet connectivity seems good, there is no packet loss to that location from the PBX. Everytime the user starts eyebeam, the application tries to register. Asterisk accepts the registration but the reply never gets to the client application, so it thinks it has not been accepted and times out. Then Asterisk unregisters the extension. -- Registered SIP '881' at 212.248.xxx.xxx port 26605 expires 300 -- Unregistered SIP '881' Anyone got any ideas how to debug and fix this? -- Chris Mason -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 Queue Problem - Automatic Call Distribution
Hi all, I am seeing a strange problem with Asterisk queue. I am not sure if it's my configuration which is wrong or there's something with Asterisk. I am using Asterisk 1.4.2 and i have a queue with one MGCP member. When i tried to call the extension number directing to the queue, the MGCP phone is not ringing. However, it is fine to call the MGCP phone directly. The strange thing is after i've called the MGCP phone directly, calling the extension number directing to the queue works fine. I wonder what could be wrong. Any comment and help is very much appreciated. The following is the configuration: queues.conf [queue1] strategy=roundrobin member=MGCP/[EMAIL PROTECTED] mgcp.conf [101] host=dynamic context=default canreinvite=no callerid=101101 line=101 extension.conf exten=601,1,Queue(queue1) Regards, Chong ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate and ForkCDR()
Thank you! I solved my problems and thanks to your useful link I have now a better understanding of how Local channels work. Federico -- Federico Cabiddu RD Software Engineering Abbeynet S.p.A. - www.abbeynet.com http://www.abbeynet.com phone: +39 070 2339349 http://www.federico_cabiddu.sitofono.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
Re: [asterisk-users] Feasibility Request
Hello Jeremy, We have implemented HA systems in the past for numerous clients, call centers, etc, where reliability matters. We can definitely offer you a bid on this, but I would like to speak with you a bit first to nail down the requirements. What would be a convenient time and number to reach you? Chris Jeremy Mann wrote: I have a ton of Nortel MICS/CICS phone systems and am looking for an easy way to integrate them. Two questions arise: 1.Is it feasible to use asterisk as a Man in the Middle for a T1 PRI system? The idea is to intercept outbound calls from the Nortel PBX and redirect them via VoIP to another asterisk box at another branch transparently(thus saving the LD cost). Otherwise I'd pass the call on to the T1 for outbound processing. Our Nortel is already PRI equipped, the PRI would just come from the Asterisk box instead of the Telco directly. 2. Is it feasible to use asterisk as a Man in the Middle for Analog lines? I'd be using anywhere from 4-12 lines depending on location size. I'd like to do the same feature as above(intercept outbound calls and redirect them using VoIP if they are inter-office calls. a. I'd also like the VoIP trunks to be used for outbound calls in the case of PSTN downtime or busy. For example, all 4 outgoing lines are in use, person 5 wants to make an outbound call and it gets redirected to one of my T1 offices. I'd attach their outbound caller ID to make it appear as the call came from that location. My inevitable hope is to reduce my analog presense in smaller communities to 1 primary Line for 911/emergency calling, and to get a published presense in the community. I'd then beef up my T1 locations to handle more VoIP based calls. Currently we're using on the order of 30k minutes a month of LD just intercompany, about 10k external (IntraLATA). I'd also like any insight or suggestions on uptime. We're a healthcare organization so 5-9's is what we'll require. Any suggestions on hardware configs(or better yet, Bids!) would be appreciated as well. I don't need VoIP capable phones yet, but if the system works well enough we'd probably startup our next location(averaging 3-6 per quarter) with a pure VoIP system with Nortel fallback(again, 5-9's is critical). I'm located in Dallas, TX for any bids that might include installation. We have a presense up to about 400 miles west of here. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by *MailScanner* http://www.mailscanner.info/, and is believed to be clean. ___ --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] Busy tone with the different length tone
Hi, I have a problem with busy tone detection. the problem is busy tone with the different length tone and silence! Means: Busy tone = 400/400,0/345,400/230,0/520 400 on 345 off 230 on 520 off Repeat I try in Zapata.conf to enable busy tone detection by this way busydetect=yes callprogress=no busycount=3 busypattern=400,345 But the problem busypattern take only one on and one off. But I have now two on and two off. So what I do? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with CDR and DeadAGI
Hello, I've a Problem with my CDRs. The clid and src Fields are empty, if a hangup inside my DeadAGI Program occurs. It makes no difference if the AGI Program or the caller initiated the hangup. The Problem doesn't exists, when i use the normal AGI Application. Further, this Problem started after the update from 1.4.2 to 1.4.3 (1.4.4 is also concerned). I hope, that someone has a solution for this Problem ? Arnd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI got event
On 5/16/07, Oscar Atienza [EMAIL PROTECTED] wrote: Hi all, I have 1 Card Digium TE412P and 2PRI E1. I have more problems with drops lines. The asterisk log is this: May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 May 16 10:52:26 WARNING[4465]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 This indicates an unstable d-channel. Try changing dchan in zaptel.conf to hardhdlc. If that fixes it, you are missing interrupts for one reason or another. I would also advise that you call Digium's tech support. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] function_db_read: DB requires an argument, DB(family/key)
Per Jessen wrote:
Lee Jenkins wrote:
OK, so I tried this:
exten = _X.,1,Noop(CallerId is ${CALLERID(all)})
exten = _X.,n,Noop(blurp)
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
This now appears to execute the first Noop(), skip the second, and
then issue the no argument warning on the Set() call.
Try an Answer() first?
OK, tried that, didn't change anything.
What I still don't get is - why does reloading the app_queue module fix
this problem? The app_queue issue is another one, but I just can't see
how it would influence the workings of the DB() function.
/Per Jessen, Zürich
pulls cid off the line
Set(CALLERID(name)=${CALLERID(name)})
pulls cidname (cid rewriting from astdb)
Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
pulls cidnum
Set(CALLERID(num)=${CALLERID(num)})
you could try reordering the dialplan so it's _X,1 _X,2 _X,3 instead of _X,1
_X,n _X,n as
well
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[asterisk-users] G729 Transcoding problems
Hi I've bought the Digium g729 codec and have installed it correctly (I think) voip*CLI show g729 0/0 encoders/decoders of 3 licensed channels are currently in use if I do an echo test either from a sip Cisco 7960 or another hard phone (unbranded) using g729 it sometimes works and sometimes the announcement about the test (echo-test.gsm) fails part way though but the test continues to work ie I can hear the echo test - if Asterisk doesn't crash first! If I place a call from the Cisco phone or other phone using g729 / SIP into my * server and then out to my service provider using IAX2 / GSM asterisk restarts and the call fails I'm running a Trixbox 2.0 system but I have mannually patched it to Asterisk 1.2.18 Zaptel 1.2.17.1 Digium HPEC 8 (9 is not working right) Sangoma A200 with the wanpipe-3.1.0.p21-zaptel-patched drivers installed I've tried recompiling all these too. Since installing the g729 I've tried the different cpu types (the machine is a P3 650MHz) and found that the i686 works most reliably. any help much appreciated as I've search the Internet and this mailing list and am still stuck Harvey___ --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] FW: Play a file on a channel from the Manager API
Anyone have any ideas how this might work? Any experience doing this? Is there any way to play a file on a channel from the Manager API (other than from Originate)? This question was asked by someone else on the ast-dev list and the only advice given was that Redirect was the solution. I find myself with the same problem now but I don't understand the response. The situation: I need to play a file from the Asterisk Manager on a channel that is currently in a call. I don't want to break them out of the call to play the message and I only want one specific channel to hear the message. In effect I want to ChanSpy the channel but to play a message instead of speak to the person on the channel. How does Redirect provide a solution? Thanks again, George ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.2.1 and TE212P
On 15 May 2007, at 20:05, Matthew Fredrickson wrote: Are you sure that you set the T1/E1 jumpers on the board correctly for E1 mode? Matthew, Ah .. that old chestnut ! , thanks - I did not actually install the card myself (however I should have checked - schoolboy error.) Thanks again. Regards Matt Brown ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk manager interface stability
If it's not stable what needs to be done to improve this? What are the issues? What are the alternatives (eg is Adhearsion an alternative here) I am about to start looking into a project that requires every user to have AMI access so looking to fund development in this space. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Wednesday, 16 May 2007 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk manager interface stability There are many past posts stating that AMI is not stable when multiple client computers are allowed to connect, particularly when connections are dropped. Has much progress been made on this? Is it more stable now than in the past? As of what versions were these issues improved? Is it feasible to connect a large number of windows computers directly to AMI for the purpose of initiating calls from software? image001.gif___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
2007/5/15, George Pajari [EMAIL PROTECTED]: If you have the clipping issue, make sure you get HPEC version 8.2 from Digium. Note, however, that we have observed stability issues with HPEC 8.2 (two kernel panics in two weeks since installing HPEC). (The 9.00 version had such severe clipping that we could not run it long enough to determine if it was more stable). g. So, what's going on ? Should someone file a bug report, so that we can check progress on this ? We have customers waiting for echo cancellation improvement. As we already tried to use HPEC, it would be very hard to either try something different nor to wait without deadline. So ? ___ --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] GSM Cards for Asterisk (UK)
Hi, I am currently building a 1.4.4 Asterisk box for a client and they are interested in GSM functionality. Does anyone have any experience with a GSM card, preferably Quad Span (4 GSM modules or higher) for use in the UK. I have seen the Junghanns* version but I am not keen on the limitation of having to use a BriStuffed version of Asterisk. Do Digium make one ? as I am unable to find on their website or is it possible to compile the ztgsm parts into the current zaptel source ? *Junghanns if you are on list, please do not take the wrong way - the cards are fine, we use a QuadBri in our very own PBX - but it does mean we are having to run the experimental version from your website for asterisk 1.2, where as we would prefer to be using 1.4 :-) Regards Matt Brown ___ --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 INVITE failing and AgentCallBackLogin()
Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite: Failed to authenticate on INVITE to 'Ex 301 sip:[EMAIL PROTECTED] ;tag=as4e18cbb4' I have a peer setup in the box doing the AgentCallBackLogin() with insecure=very, ive also tried insecure=invite as well, no luck!! Asterisk 1.2.13 I am using on both boxes. Can anyone provide any help on this? I think is rellly weird invites are failing when im telling * to ignore them basically!!! Phones are Snom 360's as well. Thanks! Ron ___ --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] HUDlite Server on Debian Etch/Asterisk 1.4
Hi, has anyone managed to get hudlite server working on a Debian Etch based installation of Asterisk 1.4? So far I managed to eliminate all error messages, but the process is killed directly after starting the hudlite server without showing any error messages. I would be very happy if anyone can give me some hints or point me to a installation guide. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue Problem - Automatic Call Distribution
On 5/16/07, TienSen Chong [EMAIL PROTECTED] wrote: Hi all, I am seeing a strange problem with Asterisk queue. I am not sure if it's my configuration which is wrong or there's something with Asterisk. I am using Asterisk 1.4.2 and i have a queue with one MGCP member. When i tried to call the extension number directing to the queue, the MGCP phone is not ringing. However, it is fine to call the MGCP phone directly. The strange thing is after i've called the MGCP phone directly, calling the extension number directing to the queue works fine. I wonder what could be wrong. Any comment and help is very much appreciated. The following is the configuration: queues.conf [queue1] strategy=roundrobin member=MGCP/[EMAIL PROTECTED] I don't think that's a valid interface string as far as app_queue is concerned at present. I'll have to take a look at that. I think a workaround would probably be to define a Local channel as the queue member and then dial to the MGCP phone in the exten you're defining for the Local channel. -- 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
Re: [asterisk-users] PRI got event
On 5/16/07, William Moore [EMAIL PROTECTED] wrote: On 5/16/07, Oscar Atienza [EMAIL PROTECTED] wrote: Hi all, I have 1 Card Digium TE412P and 2PRI E1. I have more problems with drops lines. The asterisk log is this: May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 May 16 10:52:26 WARNING[4465]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 This indicates an unstable d-channel. Try changing dchan in zaptel.conf to hardhdlc. If that fixes it, you are missing interrupts for one reason or another. I would also advise that you call Digium's tech support. I've seen this be a problem with the LBO value being wrong in the zaptel.conf. ___ --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] Which KDE editor to edit Asterisk config files ?
Hi, New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor with which I could easily edit Asterisk config files. It seems Kate provide this type of service but I couldn't find anything specific to Asterisk (unlike vim) What's your advice ? Best regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk manager interface stability
The issue has more to do with the sheer amount of data passed to the client from within the Asterisk application when you have 50-100+ clients connected to the AMI on full output mode. Running a system with FreePBX/Trixbox especially generates vast amounts of output that has to be generated on every AMI connection for every client. This is not trivial and can result in lockups very easily, although this has gotten much better since the early 1.0 versions. The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. As for how to go about doing this, I can't help you there. I did build a very specialized version of something like this 4 years ago for the astGUIclient project called the Asterisk Central Queue System(ACQS) It is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is limited in what it does, but it does scale much better than using direct AMI connections. http://astguiclient.sourceforge.net/acqs.html MATT--- On 5/16/07, Dean Collins [EMAIL PROTECTED] wrote: If it's not stable what needs to be done to improve this? What are the issues? What are the alternatives (eg is Adhearsion an alternative here) I am about to start looking into a project that requires every user to have AMI access so looking to fund development in this space. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph [image: Call Button]http://click.mexuar.com/webuser/click/7/userurl/Cognation -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Damon Estep *Sent:* Wednesday, 16 May 2007 7:32 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] asterisk manager interface stability There are many past posts stating that AMI is not stable when multiple client computers are allowed to connect, particularly when connections are dropped. Has much progress been made on this? Is it more stable now than in the past? As of what versions were these issues improved? Is it feasible to connect a large number of windows computers directly to AMI for the purpose of initiating calls from software? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Cards for Asterisk (UK)
On Wed, 16 May 2007, Matt Brown wrote: Hi, I am currently building a 1.4.4 Asterisk box for a client and they are interested in GSM functionality. Does anyone have any experience with a GSM card, preferably Quad Span (4 GSM modules or higher) for use in the UK. I have seen the Junghanns* version but I am not keen on the limitation of having to use a BriStuffed version of Asterisk. The only thing I've used is a GSM phone terminal - they are designed for use in remote areas - box with antennae and a phone socket on it. You plug in an analogue phone and off you go - I've used them in asterisk boxes on analogue cards and they work OK - nothing special. ~£150. I never got incoming callerId to work... (claimed to be bell compatable - from a UK built unit too!!!) Probably not what you want if you're looking for 4+ though! Theres this: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html I've also seen a PCI card which can take 4 SIMs, from an Italian company I think. when I made enquiries about these some months back I reckoned them being in the £1400 range... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Passing dialstatus back through an IAX chain ..
I feel I'm doing something obviously wrong here and will kick myself when I see
the answer!!!
The scenario:
SIP phone - Asterisk1 - IAX - Asterisk2 - IAX - Asterisk3 - PSTN
So I place a call from the SIP phone. A1 picks it up and forwards it to A2
which forwards it to A3. A3 sends the call to the PSTN. I control A1 and A2,
but not A3.
When a call fails (for either unavalable or busy), A2 sees the failure code
back from A3. A2 doesn't do anything with it other than reach the end of the
dialplan segment for that call. A1 carries on after it had waited for it's
Dial() to A2 to complete, however A1 doesn't then see the status code - it
always gets CONGESTION.
Output from the monitor on A2 looks like:
-- Executing NoOp(IAX2/a1-3, Call failled. Result code is: CHANUNAVAIL)
in new stack
== Auto fallthrough, channel 'IAX2/a1-3' status is 'CHANUNAVAIL'
This is good - I had dialled an invalid number.
And on A1 it gets:
-- IAX2/a2-16384 is circuit-busy
-- Hungup 'IAX2/a2-16384'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing NoOp(SIP/101-081563a0, Dialling out via VoIP Trunk failed:
CONGESTION) in new stack
Congestion and not Unavail... Where did I go wrong?
A1 has this in extensions.conf:
exten = _0.,n,Dial(IAX2/a2/${EXTEN},WTo)
exten = _0.,n,Noop(Dialling out via VoIP Trunk failed: ${DIALSTATUS})
and A2 has:
exten = _[0-9].,n,SetCallerID(0123456789)
exten = _[0-9].,n,Set(CDR(accountcode)=a1)
exten = _[0-9].,n,Dial(IAX2/trunk1/${EXTEN})
exten = _[0-9].,n,Noop(Call failled. Result code is: ${DIALSTATUS})
So how can I get the original status code that A2 sees be returned to A1 ? I
feel I'm simply missing a simple flag, parameter, variable to set or
something Any clues?
Thanks,
Gordon
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Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
It will take me a few hours to write a syntax highlighter for kate. But if I do, I can commit this for KDE 3.5.8 and KDE4. On Wednesday 16 May 2007 17:12, Olivier wrote: Hi, New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor with which I could easily edit Asterisk config files. It seems Kate provide this type of service but I couldn't find anything specific to Asterisk (unlike vim) What's your advice ? Best regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI got event
On May 16, 2007, at 7:43 AM, William Moore wrote: On 5/16/07, Oscar Atienza [EMAIL PROTECTED] wrote: Hi all, I have 1 Card Digium TE412P and 2PRI E1. I have more problems with drops lines. The asterisk log is this: May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 May 16 10:52:26 WARNING[4465]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 This indicates an unstable d-channel. Try changing dchan in zaptel.conf to hardhdlc. If that fixes it, you are missing interrupts for one reason or another. I would also advise that you call Digium's tech support. I don't think hardhdlc will fix this problem. This indicates that there is an alarm on the span. He needs to find the source of the alarms to fix this. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with conferences, Vlada, Pancevo
the forst problem you have, you need to los the meetme module, and second one is a timer, for that you can use ztdummy, compiling the zaptel driver. Regards On 5/7/07, Ronaldo [EMAIL PROTECTED] wrote: Hi, I'm not sure, but MeetMe needs some timer module from zaptel project. Try read about timers for MeetMe application. Ronaldo. Vladimir Kovacevic wrote: Hi, I have problem with setting up a conferences. When I dial the reserved conference number from xlite the line gets hunged up and on a console there is a following message: WARNING[5924]: pbx.c:1783 pbx_extension_helper: No application 'MeetMe' for extension (internal, 1234, 3) exten = 1234,1,Answer() exten = 1234,4,MeetMe(1234|Md) exten = 1234,101,HangUp() meetme.conf: [general] [rooms] conf = 1234 What I did wrong? Thx, Vlada ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk manager interface stability
why not do it via an snmp interface? If you spend the time building an solid snmp base you would open up for an easier world of custom gui's as well as possibly some cleaner ties into an nms infrastructure. On May 16, 2007, at 10:14 AM, Matt Florell wrote: The issue has more to do with the sheer amount of data passed to the client from within the Asterisk application when you have 50-100 + clients connected to the AMI on full output mode. Running a system with FreePBX/Trixbox especially generates vast amounts of output that has to be generated on every AMI connection for every client. This is not trivial and can result in lockups very easily, although this has gotten much better since the early 1.0 versions. The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. As for how to go about doing this, I can't help you there. I did build a very specialized version of something like this 4 years ago for the astGUIclient project called the Asterisk Central Queue System(ACQS) It is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is limited in what it does, but it does scale much better than using direct AMI connections. http://astguiclient.sourceforge.net/acqs.html MATT--- On 5/16/07, Dean Collins [EMAIL PROTECTED] wrote: If it's not stable what needs to be done to improve this? What are the issues? What are the alternatives (eg is Adhearsion an alternative here) I am about to start looking into a project that requires every user to have AMI access so looking to fund development in this space. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] ] On Behalf Of Damon Estep Sent: Wednesday, 16 May 2007 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk manager interface stability There are many past posts stating that AMI is not stable when multiple client computers are allowed to connect, particularly when connections are dropped. Has much progress been made on this? Is it more stable now than in the past? As of what versions were these issues improved? Is it feasible to connect a large number of windows computers directly to AMI for the purpose of initiating calls from software? ___ --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 -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --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] Outside lines are *STILL* just not happening...
From yesterday:
-Original Message-
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0
context=internal
signalling=fxo_ks
channel = 1-4
context=external
signalling=fxs_ks
channel = 5-8
A snippet from /etc/asterisk/extensions.conf:
[internal]
ignorepat = 9
exten = _9NXX,1,Dial(Zap/5/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.
Problem B: Dialing out. From the SIP phone, if I dial out, here's the
transcript:
-- Executing Dial(SIP/102-081854e0, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-081854e0
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.
I know this has been long, and wordy...hope someone can help. We're
newbs around here, and trying to get things working. My boss is *very*
impressed with the menus and such I've got set up and working, and we've
used soft phones via VPN and it works great...now we just need our
outside lines working!
Thanks a million!
J. David Bavousett
System Administrator
Abilene Library Consortium
One other symptom: I never, ever hear any ringing on the SIP phone
while it's dialing out. No indication is ever given that it's ringing
on the other end.
Can *anyone* shed some light on this?
--David
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Re: [asterisk-users] GSM Cards for Asterisk (UK)
We use instead sip gsm gateway, with 4 gsm modules from 2N. It should work in UK as well. Jorge Matt Brown wrote: Hi, I am currently building a 1.4.4 Asterisk box for a client and they are interested in GSM functionality. Does anyone have any experience with a GSM card, preferably Quad Span (4 GSM modules or higher) for use in the UK. I have seen the Junghanns* version but I am not keen on the limitation of having to use a BriStuffed version of Asterisk. Do Digium make one ? as I am unable to find on their website or is it possible to compile the ztgsm parts into the current zaptel source ? *Junghanns if you are on list, please do not take the wrong way - the cards are fine, we use a QuadBri in our very own PBX - but it does mean we are having to run the experimental version from your website for asterisk 1.2, where as we would prefer to be using 1.4 :-) Regards Matt Brown ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The downside of Asterisk and least cost routing...
Chris Mason (Lists) wrote: Stephen Bosch wrote: The fax-to-e-mail services charge as much as the telco does for a business line, sometimes more (at least, the ones I can deal with in this area). Better to set-up hylafax, IMHO. http://www.maxemail.com/fax/fax-lite.html $24/annum. I suppose I should have added the qualifier that we need to be able to port our existing fax number over to the new service :) Would this still be possible? (All these services have numbers in remote area codes or have 800 numbers). Can anybody suggest one that will take a ported number (in Canada)? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk manager interface stability
Quoting Bryan Laird [EMAIL PROTECTED]: why not do it via an snmp interface? If you spend the time building an solid snmp base you would open up for an easier world of custom gui's as well as possibly some cleaner ties into an nms infrastructure. you have my vote on that implementation method. snmp really is simple, but it seems to be a neglected protocol that has been around for a long time. On May 16, 2007, at 10:14 AM, Matt Florell wrote: The issue has more to do with the sheer amount of data passed to the client from within the Asterisk application when you have 50-100 + clients connected to the AMI on full output mode. Running a system with FreePBX/Trixbox especially generates vast amounts of output that has to be generated on every AMI connection for every client. This is not trivial and can result in lockups very easily, although this has gotten much better since the early 1.0 versions. The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. As for how to go about doing this, I can't help you there. I did build a very specialized version of something like this 4 years ago for the astGUIclient project called the Asterisk Central Queue System(ACQS) It is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is limited in what it does, but it does scale much better than using direct AMI connections. http://astguiclient.sourceforge.net/acqs.html MATT--- On 5/16/07, Dean Collins [EMAIL PROTECTED] wrote: If it's not stable what needs to be done to improve this? What are the issues? What are the alternatives (eg is Adhearsion an alternative here) I am about to start looking into a project that requires every user to have AMI access so looking to fund development in this space. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] ] On Behalf Of Damon Estep Sent: Wednesday, 16 May 2007 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk manager interface stability There are many past posts stating that AMI is not stable when multiple client computers are allowed to connect, particularly when connections are dropped. Has much progress been made on this? Is it more stable now than in the past? As of what versions were these issues improved? Is it feasible to connect a large number of windows computers directly to AMI for the purpose of initiating calls from software? ___ --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 -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Cards for Asterisk (UK)
On Wed, May 16, 2007 at 02:17:11PM +0100, Matt Brown wrote: I am currently building a 1.4.4 Asterisk box for a client and they are interested in GSM functionality. Does anyone have any experience with a GSM card, preferably Quad Span (4 GSM modules or higher) for use in the UK. I have seen the Ensure the client is ONLY using for their own use (i.e. they're not handling ANY 3rd party calls through their system) or their operating in an illegal manner. Ofcom does allow GSM gateways to be used for your own use. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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
Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
This might be of some assistance: http://www.voip-info.org/wiki/view/vim+syntax+highlighting - sf Olivier wrote: Hi, New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor with which I could easily edit Asterisk config files. It seems Kate provide this type of service but I couldn't find anything specific to Asterisk (unlike vim) What's your advice ? Best regards !DSPAM:1020,464b158e638175802679531! ___ --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:1020,464b158e638175802679531! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outside lines are just not happening...
J. David Bavousett wrote: Problem A: Dialing in. If I call from my cell, the FXO picks right up, and sends me to the voice menu that I have at the top of the [external] context. So far so good, but if the SIP that I get in touch with hangs up, the FXO stays off-hook for more than a minute before dropping the POTS line. If I pick that SIP phone back up, and dial an outside number, I can reconnect to the dangling call, which will hear the tones after the 9... The outside caller will finally get dropped after about a minute of waiting. You need to tell the telco to change the disconnect supervision or CPC (Calling Party Control) parameters for your switch. This can be tricky if your number is a residential number, depending on the telco. If it's a business line, *demand* that they be changed. There are still lots of people using key systems with analog lines, and they need the same kind of disconnect supervision. Here's what I do: I call the telco repair number and I open a trouble ticket. I say I am having trouble with my equipment because of an improper setting on the telco side; then I say that I need the disconnect interval on calling party disconnect set to 5 seconds, and I ask that the battery drop be extended to 500 ms (just to be safe). Once those changes are made, it works really very well. If you get dumbfounded responses ask to speak to someone in the programming group (unless they are a tiny little phone company, they will have one). If you open a ticket, it usually means they will escalate the problem, even if the agent you are speaking with has no idea what you are talking about. Best to be friendly with them! 98% of modern switches can do this (otherwise, how would it even know to disconnect after one whole minute? It has to have some way of knowing that the remote port has dropped; on most switches, 65 seconds is the default; it's just a matter of changing that interval for your port) and your biggest challenge will be finding someone who has a clue, but if you do it right, it's not a big deal. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
François Delawarde wrote: aaah... I'm running asterisk in a Xen kernel, but not on a virtual machine (DomU), only on Dom0, so it's supposed to be running on the physical server (no PCI frontend device, ...). I had seen possible problems with older versions of Xen, but only with ztdummy timing and on DomU virtual machines. On Xen mailing list, they all advise to use a digium PCI card to remove those problems. You had seen? Did you have these problems personally, or are you going by mailing list postings? I will test and report what happens with a normal kernel, but meanwhile doesn't anyone know of a possible possibility to make it work with this setting playing for example with IRQ priorities or something, or isn't there any hope at all? We abandoned Xen (recent versions too!) after serious interrupt problems (it doesn't matter if you are in domU or dom0, by the way) that caused the entire *system*, with all the VMs, to lock up *hard* whenever we started to push significant amounts of data through anywhere, be it an Ethernet controller or a SCSI adapter. It is in need of a lot of work. Their efforts to commercialize it are premature. We had to learn this hard way, unfortunately. If you need virtualization that badly, you might want to consider going with VMware Server, which is now freely available. My experience with VMware has been better. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
Olivier wrote: 2007/5/15, George Pajari [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: If you have the clipping issue, make sure you get HPEC version 8.2 from Digium. Note, however, that we have observed stability issues with HPEC 8.2 (two kernel panics in two weeks since installing HPEC). (The 9.00 version had such severe clipping that we could not run it long enough to determine if it was more stable). g. So, what's going on ? Should someone file a bug report, so that we can check progress on this ? We have customers waiting for echo cancellation improvement. As we already tried to use HPEC, it would be very hard to either try something different nor to wait without deadline. So ? Um -- Digium is need of sound captures and debug output for this problem, because they are having trouble reproducing it in the laboratory. Perhaps you could help out by making a capture of the clipping phenomenon using the latest Zaptel (1.2.17) and ztmonitor to capture audio on the affected channel. All our cards went into production machines so I can't do this right now. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk manager interface stability
Matt Florell wrote: The issue has more to do with the sheer amount of data passed to the client from within the Asterisk application when you have 50-100+ clients connected to the AMI on full output mode. Running a system with FreePBX/Trixbox especially generates vast amounts of output that has to be generated on every AMI connection for every client. This is not trivial and can result in lockups very easily, although this has gotten much better since the early 1.0 versions. The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. I definitely agree here personally. Clients could connect to this proxy and subscribe to only the events that are interesting or applicable. As for how to go about doing this, I can't help you there. I did build a very specialized version of something like this 4 years ago for the astGUIclient project called the Asterisk Central Queue System(ACQS) It is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is limited in what it does, but it does scale much better than using direct AMI connections. I've been considering writing something like this for a project that I'm thinking about doing that would require potentially high number of concurrent clients to consume AMI services. From your experience, does the software that you wrote require significant CPU to cache and then doll out the kind of volume of messages that AMI can send? If I end up writing something myself, I'll release it as OS... -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
Do you mean nobody has ever done this before (as I thought before asking this question to the list) ? So which tool KDE users are using for this ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voice recording on legacy PBX
hi Sir, Thank very much for your suggestion. But I hope you don't mind giving me more detail information. Imagine that I have a pbax with 3 incoming PSTN line and I have 10 extentions using stand phone. The requirement is to capture all conversation either internal or external Regards ASLAY Hi, Connect you Asterix box in the middle of call-flow and you will be able to record all calls. PSTN - Asterisk - Legacy PBX - Phones Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com VoIP ServicesSolutions MOR - FREE Open Source billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, May 16, 2007 1:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voice recording on legacy PBX Hi, Is it possible to use Asterisk to record or monitor all conversation on standard PSTN PBX ? ASLAY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HPEC audio clipping
Octasic SoftEcho works very well for me. _ From: Olivier [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 16, 2007 6:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HPEC audio clipping 2007/5/15, George Pajari [EMAIL PROTECTED]: If you have the clipping issue, make sure you get HPEC version 8.2 from Digium. Note, however, that we have observed stability issues with HPEC 8.2 (two kernel panics in two weeks since installing HPEC). (The 9.00 version had such severe clipping that we could not run it long enough to determine if it was more stable). g. So, what's going on ? Should someone file a bug report, so that we can check progress on this ? We have customers waiting for echo cancellation improvement. As we already tried to use HPEC, it would be very hard to either try something different nor to wait without deadline. So ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outside lines are *STILL* just not happening...
In your zaptel.conf you need to use fxsks rather than fxoks
hth
Harvey
- Original Message -
From: J. David Bavousett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 16, 2007 3:59 PM
Subject: [asterisk-users] Outside lines are *STILL* just not happening...
From yesterday:
-Original Message-
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0
context=internal
signalling=fxo_ks
channel = 1-4
context=external
signalling=fxs_ks
channel = 5-8
A snippet from /etc/asterisk/extensions.conf:
[internal]
ignorepat = 9
exten = _9NXX,1,Dial(Zap/5/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.
Problem B: Dialing out. From the SIP phone, if I dial out, here's the
transcript:
-- Executing Dial(SIP/102-081854e0, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-081854e0
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.
I know this has been long, and wordy...hope someone can help. We're
newbs around here, and trying to get things working. My boss is *very*
impressed with the menus and such I've got set up and working, and we've
used soft phones via VPN and it works great...now we just need our
outside lines working!
Thanks a million!
J. David Bavousett
System Administrator
Abilene Library Consortium
One other symptom: I never, ever hear any ringing on the SIP phone
while it's dialing out. No indication is ever given that it's ringing
on the other end.
Can *anyone* shed some light on this?
--David
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Re: [asterisk-users] The downside of Asterisk and least cost routing...
On Wed, 16 May 2007, Stephen Bosch said something to this effect: Would this still be possible? (All these services have numbers in remote area codes or have 800 numbers). Can anybody suggest one that will take a ported number (in Canada)? That's just something you have to contact the provider and ask about. Some of them use North American carriers that will port your number, even if they don't provide ready DID inventory in your LATA. What is your objection to an 800 number? -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue Problem - Automatic Call Distribution
Hi Chong, I have no experience with MGCP, but do you see anything in the Asterisk CLI or the full log while the terminal is supposedly being called by the ACD? Another thing you could do is to create a Local/[EMAIL PROTECTED] channel, whee as the queue memebre you have Local/[EMAIL PROTECTED] and that does the actual calling to the MGCP terminal - you should see something in the CLI at this point. Hope this helps, l. On Wed, 16 May 2007 13:45:19 +0200, TienSen Chong [EMAIL PROTECTED] wrote: Hi all, I am seeing a strange problem with Asterisk queue. I am not sure if it's my configuration which is wrong or there's something with Asterisk. I am using Asterisk 1.4.2 and i have a queue with one MGCP member. When i tried to call the extension number directing to the queue, the MGCP phone is not ringing. However, it is fine to call the MGCP phone directly. The strange thing is after i've called the MGCP phone directly, calling the extension number directing to the queue works fine. I wonder what could be wrong. Any comment and help is very much appreciated. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Video Door Phone
I have a customer that has a campground. Wants to see who's at the gate, remotely, via camera, and talk to that person through a traditional squawk box and be able to open the gate remotely from that phone. He doesn't want to have a separate camera feed, etc, he wants to do it all on one phone. Does such a way to do this exist by using Asterisk and some kind of relay system / Video phone ? R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outside lines are *STILL* just not happening...
in zaptel.conf use fxsks
- Original Message -
From: J. David Bavousett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 16, 2007 3:59 PM
Subject: [asterisk-users] Outside lines are *STILL* just not happening...
From yesterday:
-Original Message-
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0
context=internal
signalling=fxo_ks
channel = 1-4
context=external
signalling=fxs_ks
channel = 5-8
A snippet from /etc/asterisk/extensions.conf:
[internal]
ignorepat = 9
exten = _9NXX,1,Dial(Zap/5/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.
Problem B: Dialing out. From the SIP phone, if I dial out, here's the
transcript:
-- Executing Dial(SIP/102-081854e0, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-081854e0
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.
I know this has been long, and wordy...hope someone can help. We're
newbs around here, and trying to get things working. My boss is *very*
impressed with the menus and such I've got set up and working, and we've
used soft phones via VPN and it works great...now we just need our
outside lines working!
Thanks a million!
J. David Bavousett
System Administrator
Abilene Library Consortium
One other symptom: I never, ever hear any ringing on the SIP phone
while it's dialing out. No indication is ever given that it's ringing
on the other end.
Can *anyone* shed some light on this?
--David
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Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
I use Bluefish, and have developed a syntax-highlighting template for Asterisk conf files, if you're interested. CP Steve Finkelstein wrote: This might be of some assistance: http://www.voip-info.org/wiki/view/vim+syntax+highlighting - sf Olivier wrote: Hi, New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor with which I could easily edit Asterisk config files. It seems Kate provide this type of service but I couldn't find anything specific to Asterisk (unlike vim) What's your advice ? Best regards !DSPAM:1020,464b158e638175802679531! ___ --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:1020,464b158e638175802679531! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Iaxy clicking
Hi, Can any one suggest if asterisk-users is the best mailing list for questions on Digium Iaxy (S101I) hardware, or a different one if not? I found this link on Digium's site: http://kb.digium.com/entry/15/120/ However, I assume that if this was the case, all of my Iaxys would click, and only one of mine does. Is Digium referring to clicking coming from the FXO/FXS hardware or the Iaxy device? My constant clicking is coming out of the Iaxy, whether or not it's connected to the VOIP network. Thanks, Matthew Yingling -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Yingling Sent: Thursday, May 10, 2007 5:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Iaxy clicking Hi, I have three Iaxy devices (s101i) parts. Two of them seem to work fine. The third plays a loud repeating click sound when an analog phone is plugged in. I can provision all of them, and make calls to all of them. The clicking one will blink when a call is incoming, but no audio from the call can be heard on the handset, and the caller only hears silence. The same handset works on the other Iaxys, and other handsets have the same clicking issue. Resetting the Iaxy doesn't seem to fix the problem. Does anyone have any ideas on how to fix this problem, or whether the Iaxy is broken and unfixable (for me as an end-user). Thanks, Matthew Yingling ___ --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] SIP Hardware Phone
Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY ___ --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] Microsoft's Move Into IP PBX Market
From c|net News: On Monday,Microsoft and nine leading phone manufacturers--Asustek Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007. http://news.com.com/8301-10784_3-9719931-7.html?part=rsssubj=newstag=2547-1_3-0-20 -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
Hi Olivier, I would guess that most people aren't running any type of GUI on their Asterisk box. Running an X server plus some type of window manager adds a lot of overhead that's completely unnecessary for a server. I just SSH into the server and use VI to edit the files - The server doesn't run any type of GUI, there's no reason for it to. Alex On 5/16/07, Olivier [EMAIL PROTECTED] wrote: Do you mean nobody has ever done this before (as I thought before asking this question to the list) ? So which tool KDE users are using for this ? ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP INVITE failing and AgentCallBackLogin()
On Wed, May 16, 2007 at 06:21:30AM -0700, Ron McCarthy wrote: Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite: Failed to authenticate on INVITE to 'Ex 301 sip:[EMAIL PROTECTED] ;tag=as4e18cbb4' I have a peer setup in the box doing the AgentCallBackLogin() with insecure=very, ive also tried insecure=invite as well, no luck!! I'm not sure what the link you have here between SIP and Agents? Agents use chan_agent Dial(Agent/nnn) but SIP calls use chan_sip, so the two don't interact in the dialplan. (SIP User 301 is not equivalent to Agent 301, they are completely separate.) The error you have pasted here looks like either type= problem or the extension 301 doesn't exist in sip.conf of the box that the invite is being placed to. (or the IP address for the peer is wrong, etc.) I would not rely on AgentCallBackLogin(). chan_agent has limited use, which introduces a few strange problems, unchangable assumptions about how you want to handle calls, and the AgentCallBackLogin() feature has been (annoyingly) been deprecated by digium as of 1.4 The suggestion is to replicate the AgentCallBackLogin() functionality with dialplan logic, and dynamic queue members. This is possible, but very complicated (you do NOT want to see my extensions.conf!) and there is no neat way to handle hints for blf keys when you do this, as you lose the ability to dynamically track Agents in the hints config, and I haven't found a way to dynamically update the hints that doesn't crash asterisk. If you don't want BLF keys, this won't cause a problem. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip: [EMAIL PROTECTED] - inoc-dba: 5459*710- tel: +44 (0)20 7645 3510 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
Olivier wrote: Do you mean nobody has ever done this before (as I thought before asking this question to the list) ? So which tool KDE users are using for this ? I use vi(m). /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
On Wednesday 16 May 2007 11:47 am, Olivier wrote: Do you mean nobody has ever done this before (as I thought before asking this question to the list) ? So which tool KDE users are using for this ? I am a KDE user, although on Slackware. Have been for many, many years. Typically you will find that those who wish to use their GUIs to manipulate Asterisk will do so through one of the available GUIs. Those who want to work on the text files will use vim or emacs. I develop embedded systems; I use kdevelop for coding for the most part, and once in a while I'll use Kate to edit config files, but 99% of my time manipulating text files is done in vim. Even as I type this I have kdevelop open for the source and html, but I have three konsole tabs open: one to a screen session to a server I IRC from, one to a screen session to my development box in the server room (which has two login sessions going), one to a telnet session to the board I'm developing for, and finally one to a serial port server which the serial console of the development box is connected to. Kate's open, but contains a little textfile I append to which has todo lists and notes for the development project. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk manager interface stability
On Wednesday 16 May 2007 11:43, Lee Jenkins wrote: Matt Florell wrote: The issue has more to do with the sheer amount of data passed to the client from within the Asterisk application when you have 50-100+ clients connected to the AMI on full output mode. Running a system with FreePBX/Trixbox especially generates vast amounts of output that has to be generated on every AMI connection for every client. This is not trivial and can result in lockups very easily, although this has gotten much better since the early 1.0 versions. The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. I definitely agree here personally. Clients could connect to this proxy and subscribe to only the events that are interesting or applicable. As for how to go about doing this, I can't help you there. I did build a very specialized version of something like this 4 years ago for the astGUIclient project called the Asterisk Central Queue System(ACQS) It is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is limited in what it does, but it does scale much better than using direct AMI connections. I've been considering writing something like this for a project that I'm thinking about doing that would require potentially high number of concurrent clients to consume AMI services. From your experience, does the software that you wrote require significant CPU to cache and then doll out the kind of volume of messages that AMI can send? If I end up writing something myself, I'll release it as OS... You might be interested in a python server script I wrote (called ProxyMan) that does this kind of thing. It is part of my EventMonitor package but runs fine on its own. #A multi-threaded server which connects to an Asterisk Manager #and logs all events # #Connects to the Asterisk Manager and listens for all events #Optionally listens on socket and accepts client connections # proxies all client commands to the Asterisk Manager Interface # sends all data received from the manager to all connected clients #Optionally prints data as received (also in optional hex dump format) #Optionally logs all data to a MySQL database table You can get it here: http://www.micpc.com/eventmonitor earl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
Stephen Bosch wrote: François Delawarde wrote: aaah... I'm running asterisk in a Xen kernel, but not on a virtual machine (DomU), only on Dom0, so it's supposed to be running on the physical server (no PCI frontend device, ...). I had seen possible problems with older versions of Xen, but only with ztdummy timing and on DomU virtual machines. On Xen mailing list, they all advise to use a digium PCI card to remove those problems. You had seen? Did you have these problems personally, or are you going by mailing list postings? Sorry for not being Scottish anymore, my English is not what it used to be a few hundred years ago. I meant that I'm mainly going by mailing list postings. I will test and report what happens with a normal kernel, but meanwhile doesn't anyone know of a possible possibility to make it work with this setting playing for example with IRQ priorities or something, or isn't there any hope at all? We abandoned Xen (recent versions too!) after serious interrupt problems (it doesn't matter if you are in domU or dom0, by the way) that caused the entire *system*, with all the VMs, to lock up *hard* whenever we started to push significant amounts of data through anywhere, be it an Ethernet controller or a SCSI adapter. It is in need of a lot of work. Their efforts to commercialize it are premature. We had to learn this hard way, unfortunately. If you need virtualization that badly, you might want to consider going with VMware Server, which is now freely available. My experience with VMware has been better. Thanks again for your help, and sorry if I was not 'that' convinced on your first answer and sent a mail to Xen user mailing list to check if they knew that issue (no answer yet). Now I almost believe you a lot. If I understand well I have two options, recode Xen or abandon it. I'll probably go for the 2nd choice and start looking at other solutions, KVM seems to be a good choice and shouldn't interfere much with Asterisk (again: as far as mailing lists say). François. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
2007/5/16, Stephen Bosch [EMAIL PROTECTED]: Um -- Digium is need of sound captures and debug output for this problem, because they are having trouble reproducing it in the laboratory. Perhaps you could help out by making a capture of the clipping phenomenon using the latest Zaptel (1.2.17) and ztmonitor to capture audio on the affected channel. All our cards went into production machines so I can't do this right now. -Stephen- Our last trial was so conclusive (every call was affected), we step back to previous situation without HPEC. We will do our best to help to solve this (gathering audio captures for instance) though it will be very hard for me to convince our customer to try. Being able to to limit HPEC to given channels would have certainly helped to create conditions for customer acceptance. As far as I can tell, no bug is opened on this HPEC audio, though a long thread, a couple of weeks ago, proved us we were not the ones affected. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voice recording on legacy PBX
You would need two 4-port FXO cards. One to take the 3 outside POTS lines, and one to generate the 3 FXO lines toward the legacy PBX pretending to be the far end. Produce a simple dial plan that basically forwards nearly everything in and out indiscriminately and run MixMonitor() on all of the bridged calls. http://www.voip-info.org/wiki/view/MixMonitor Caveats, as I said in my original response, may include unintended breakage that is perceptible to end-users, ranging from malfunctioning features to echo issues to poor audio quality to unnaturally high or low volume, depending on the circumstances and the interoperability of the equipment. Also, I don't know about Malaysia, but here in the United States recording voice in that manner is categorically illegal without the consent of both (or more) parties to the call, or a court order in a law enforcement capacity. This is the reason that the queue announcements / IVRs of most customer service centers that may or may not potentially record your call say something about, For quality control purposes, this call may be recorded. It is a legal requirement that they do so. So, if you are doing this without the knowledge of either the PBX owners or the outside endpoints, do so at your own peril unless you are aware of local statute or are acting in a law-enforcement capacity. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk manager interface stability
You guys all sound like you're talking about AstManProxy. See: http://www.voip-info.org/tiki-index.php?page=AstManProxy I'm not saying it is the solution to your problem per se, but I can't help but think of it when I read the descriptions of what people want (you even use the word proxy!). Figured I'd send this out in case someone hadn't seen it. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Subject: Re: [asterisk-users] asterisk manager interface stability The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. I definitely agree here personally. Clients could connect to this proxy and subscribe to only the events that are interesting or applicable. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The downside of Asterisk and least cost routing...
Alex Balashov wrote: On Wed, 16 May 2007, Stephen Bosch said something to this effect: Would this still be possible? (All these services have numbers in remote area codes or have 800 numbers). Can anybody suggest one that will take a ported number (in Canada)? That's just something you have to contact the provider and ask about. Some of them use North American carriers that will port your number, even if they don't provide ready DID inventory in your LATA. What is your objection to an 800 number? The significant pre-investment in marketing materials that have our current number on it :) Not to mention an established customer base that knows our current number. (I also have little interest in paying for the privilege of receiving fax spam from all over the continent) -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
Olivier wrote: Do you mean nobody has ever done this before (as I thought before asking this question to the list) ? So which tool KDE users are using for this ? This KDE user is using vim :P (What's wrong with vim?) -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video Door Phone
You could probably make something work, but instead of trying to pound a nail with a wrench.buy a door entry control system. * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * Smith, Rick wrote: I have a customer that has a campground. Wants to see who's at the gate, remotely, via camera, and talk to that person through a "traditional squawk box" and be able to open the gate remotely from that phone. He doesn't want to have a separate camera feed, etc, he wants to do it all on one phone. Does such a way to do this exist by using Asterisk and some kind of relay system / Video phone ? R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Outside lines are *STILL* just not happening...
Really, Harvey?
Lines 1-4 are FXS ports, lines 5-8 are FXO... My outside line is
plugged into port 8. Ports 1-4 are inside, and work fine.
--David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wireless
Sent: Wednesday, May 16, 2007 10:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Outside lines are *STILL* just not
happening...
In your zaptel.conf you need to use fxsks rather than fxoks
hth
Harvey
- Original Message -
From: J. David Bavousett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 16, 2007 3:59 PM
Subject: [asterisk-users] Outside lines are *STILL* just not
happening...
From yesterday:
-Original Message-
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0
context=internal
signalling=fxo_ks
channel = 1-4
context=external
signalling=fxs_ks
channel = 5-8
A snippet from /etc/asterisk/extensions.conf:
[internal]
ignorepat = 9
exten = _9NXX,1,Dial(Zap/5/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.
Problem B: Dialing out. From the SIP phone, if I dial out, here's the
transcript:
-- Executing Dial(SIP/102-081854e0, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-081854e0
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.
I know this has been long, and wordy...hope someone can help. We're
newbs around here, and trying to get things working. My boss is *very*
impressed with the menus and such I've got set up and working, and we've
used soft phones via VPN and it works great...now we just need our
outside lines working!
Thanks a million!
J. David Bavousett
System Administrator
Abilene Library Consortium
One other symptom: I never, ever hear any ringing on the SIP phone
while it's dialing out. No indication is ever given that it's ringing
on the other end.
Can *anyone* shed some light on this?
--David
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Re: [asterisk-users] SIP Hardware Phone
[EMAIL PROTECTED] wrote: Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY ___ --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 We swear by the SNOM hardphones (we have a good number of 320s, 190s, and 360s about). Clear speakers, good microphone pickup, good volume control. The Budgetones are cheap, and they sound cheap. Good price, though. N. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardware Phone
On Thu, 17 May 2007, [EMAIL PROTECTED] wrote: Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft There are dozens if not 100's to choose from. Just use google to find them. I like Grandstream GXP2000's - they're cheap cheerful and do everything I want, but I also like the Siemens CP460IP DECT phones. Others here will rant rave on about their own favourite, just search from SNOM, Polycom, Linksys, to name but a few. This link http://www.voiptalk.org/products/VoIP+Phones will give you some idea what they look like features... You'll just have to find local distributors if you like any of them... Then there's ATAs - which let you connect an ordinary analogue phone to a SIP account... Good luck finding something that suits you ( your pocket!) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardware Phone
[EMAIL PROTECTED] wrote: Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY Polycoms seem great to me and according to much feedback I've read on this list. I had a budgetone 100 when I first started playing with Asterisk, but once I was sure I wanted to use it/learn it, I got a polycom. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
On May 16, 2007, at 8:14 AM, Olivier wrote: 2007/5/15, George Pajari [EMAIL PROTECTED]: If you have the clipping issue, make sure you get HPEC version 8.2 from Digium. Note, however, that we have observed stability issues with HPEC 8.2 (two kernel panics in two weeks since installing HPEC). (The 9.00 version had such severe clipping that we could not run it long enough to determine if it was more stable). g. So, what's going on ? Should someone file a bug report, so that we can check progress on this ? We have customers waiting for echo cancellation improvement. As we already tried to use HPEC, it would be very hard to either try something different nor to wait without deadline. So ? I know that ADT (the company that makes the echo cancellation algorithm) has received some of the sample audio files and are working on trying to fix it. We are working hard to solve this problem. All things considered, I have heard (with the new echo canceller) that if you set echocancelwhenbridged=no it makes the problem go away. Have you tried that? Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft's Move Into IP PBX Market
George Pajari wrote: From c|net News: On Monday,Microsoft and nine leading phone manufacturers--Asustek Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007. Yawn. -s ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk manager interface stability
On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote: Matt Florell wrote: The issue has more to do with the sheer amount of data passed to the client from within the Asterisk application when you have 50-100+ clients connected to the AMI on full output mode. Running a system with FreePBX/Trixbox especially generates vast amounts of output that has to be generated on every AMI connection for every client. This is not trivial and can result in lockups very easily, although this has gotten much better since the early 1.0 versions. The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. I definitely agree here personally. Clients could connect to this proxy and subscribe to only the events that are interesting or applicable. As for how to go about doing this, I can't help you there. I did build a very specialized version of something like this 4 years ago for the astGUIclient project called the Asterisk Central Queue System(ACQS) It is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is limited in what it does, but it does scale much better than using direct AMI connections. I've been considering writing something like this for a project that I'm thinking about doing that would require potentially high number of concurrent clients to consume AMI services. From your experience, does the software that you wrote require significant CPU to cache and then doll out the kind of volume of messages that AMI can send? One of the great parts about removing the broadcasting of AMI events outside of the Asterisk process is that the broadcast server process can exist on a separate physical server removing any kind of overhead on the Asterisk server. In my experience doing the proxy on the same machine uses less CPU resources than the same number of AMI connected clients, and doesn't have any of the deadlock issues that can happen with a lot of direct AMI connections. For my application(ACQS) I use MySQL as a storage engine for all of the recent events received and sent so that they can be independantly queried by any client apps that need to see them. MATT--- If I end up writing something myself, I'll release it as OS... -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardware Phone
I like and use aastra sets. Very good quality - build and sound wise. If you're in the GTA (Toronto) area, I could show you D Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft's Move Into IP PBX Market
How sad, cnet misspelled Polycom and Cisco didn't make the cut. On 5/16/07, George Pajari [EMAIL PROTECTED] wrote: From c|net News: On Monday,Microsoft and nine leading phone manufacturers--Asustek Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007. http://news.com.com/8301-10784_3-9719931-7.html?part=rsssubj=newstag=2547-1_3-0-20 -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
On Wednesday 16 May 2007 1:00 pm, François Delawarde wrote: Thanks again for your help, and sorry if I was not 'that' convinced on your first answer and sent a mail to Xen user mailing list to check if they knew that issue (no answer yet). Now I almost believe you a lot. If I understand well I have two options, recode Xen or abandon it. I'll probably go for the 2nd choice and start looking at other solutions, KVM seems to be a good choice and shouldn't interfere much with Asterisk (again: as far as mailing lists say). Let me try to understand this: Xen is a (far) more mature virtualization technology than KVM, and it's been said that it's commercialization was rushed. So you're going to try KVM, which is still under heavy development, as a stable solution? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voice recording on legacy PBX
On Wednesday 16 May 2007 1:07 pm, Alex Balashov wrote: You would need two 4-port FXO cards. One to take the 3 outside POTS lines, and one to generate the 3 FXO lines toward the legacy PBX pretending to be the far end. Produce a simple dial plan that basically forwards nearly everything in and out indiscriminately and run MixMonitor() on all of the bridged calls. Uh, you'd need 3 FXS and 3 FXO. You need to generate ring to the legacy system, which requires FXS ports. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
Hi Olivier - Our last trial was so conclusive (every call was affected), we step back to previous situation without HPEC. We will do our best to help to solve this (gathering audio captures for instance) though it will be very hard for me to convince our customer to try. Being able to to limit HPEC to given channels would have certainly helped to create conditions for customer acceptance. As far as I can tell, no bug is opened on this HPEC audio, though a long thread, a couple of weeks ago, proved us we were not the ones affected. Do you have an old machine and a spare TDM card you could use? Maybe you could connect just one POTS line to another machine and grab some audio captures. Or maybe you can test during off-hours? - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk manager interface stability
How does the Events cache in AstManProxy work?(is there a cache?) MATT--- On 5/16/07, Martin Smith [EMAIL PROTECTED] wrote: You guys all sound like you're talking about AstManProxy. See: http://www.voip-info.org/tiki-index.php?page=AstManProxy I'm not saying it is the solution to your problem per se, but I can't help but think of it when I read the descriptions of what people want (you even use the word proxy!). Figured I'd send this out in case someone hadn't seen it. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Subject: Re: [asterisk-users] asterisk manager interface stability The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. I definitely agree here personally. Clients could connect to this proxy and subscribe to only the events that are interesting or applicable. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardware Phone
I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Polycom, Snom, Cisco, Aastra ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iaxy clicking
Hi Matthew - Can any one suggest if asterisk-users is the best mailing list for questions on Digium Iaxy (S101I) hardware, or a different one if not? Digium support is the best resource for this. How old is the affected IAXy? - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft's Move Into IP PBX Market
From c|net News: On Monday,Microsoft and nine leading phone manufacturers--Asustek Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007. Now we can look forward to the same great reliability in a phone system as we experience with their operating systems. Lovely. -Kenneth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
François Delawarde wrote: Stephen Bosch wrote: François Delawarde wrote: aaah... I'm running asterisk in a Xen kernel, but not on a virtual machine (DomU), only on Dom0, so it's supposed to be running on the physical server (no PCI frontend device, ...). I had seen possible problems with older versions of Xen, but only with ztdummy timing and on DomU virtual machines. On Xen mailing list, they all advise to use a digium PCI card to remove those problems. You had seen? Did you have these problems personally, or are you going by mailing list postings? Sorry for not being Scottish anymore, my English is not what it used to be a few hundred years ago. I meant that I'm mainly going by mailing list postings. No, there was nothing wrong with the grammar -- I was just suggesting that you not take the Xen mailing list postings too seriously. Better get it from someone with personal experience (and it sounds like you've had plenty of your own personal experience with this issue already). I don't know of anybody who is using it in a serious production environment anymore, for the reasons I've already noted. You'll notice that the release schedule has been, *ahem*, very slow; the mailing lists are littered with pleading posts from users reporting crashes and freeze-ups, sometimes catastrophic. Just have a look at the archives. (The Xensource people are also very quiet, which suggests to me that they themselves don't know how to address some of problems.) For hardware, it's just not good enough. It was an interesting academic project once, but like many such projects, its transition into the applied world has been rocky at best. I would love to be proven wrong, preferably sooner than later :) Thanks again for your help, and sorry if I was not 'that' convinced on your first answer and sent a mail to Xen user mailing list to check if they knew that issue (no answer yet). Now I almost believe you a lot. If I understand well I have two options, recode Xen or abandon it. I'll probably go for the 2nd choice and start looking at other solutions, KVM seems to be a good choice and shouldn't interfere much with Asterisk (again: as far as mailing lists say). Hey -- it's no skin off my nose if you want to keep using Xen. I just think you'll be wasting a lot of your time, which I'm sure is valuable. Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voice recording on legacy PBX
Responses to your inquiry have addressed recording external calls only. If you want to record internal calls as well, I recommend that you replace the existing PBX with an Asterisk system as you have a relatively small installation. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, May 16, 2007 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] voice recording on legacy PBX hi Sir, Thank very much for your suggestion. But I hope you don't mind giving me more detail information. Imagine that I have a pbax with 3 incoming PSTN line and I have 10 extentions using stand phone. The requirement is to capture all conversation either internal or external Regards ASLAY Hi, Connect you Asterix box in the middle of call-flow and you will be able to record all calls. PSTN - Asterisk - Legacy PBX - Phones Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com VoIP ServicesSolutions MOR - FREE Open Source billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, May 16, 2007 1:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voice recording on legacy PBX Hi, Is it possible to use Asterisk to record or monitor all conversation on standard PSTN PBX ? ASLAY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
