[asterisk-users] Re: FW: Realtime Voicemail Password Change Not Working
On Jan 17, 2007, at 11:00 AM, [EMAIL PROTECTED] wrote: Date: Wed, 17 Jan 2007 09:55:50 -0700 From: David Thomas [EMAIL PROTECTED] Subject: Re: [asterisk-users] FW: Realtime Voicemail Password Change Not Working To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 1/17/07, JR Richardson [EMAIL PROTECTED] wrote: I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, enter new password ok, re-enter new password ok, password has been changed There are no entries in the mysql.log setting the new password in the database. How can I isolate between asterisk, realtime driver, and mysql? I updated to asterisk 1.2.14 and add-ons 1.2.5 with no luck. I still don't see any update statement in the mysql.log when I change a password. I built a vmbox in the voicemail.conf file and can change that password just fine. Any suggestions? JR, I'm just pulling things out of the air here, but if realtime voicemail works like realtime users/peers, loading everything into memory from MySQL, then there would need to be some type or prune command to force the re-read of the voicemail table, this is asuming you change the password via MySQL and not on the handset. Maybe something like DBput would work to update astdb as well. Again just throwing out ideas... It sounds like you are using the handset to update the password. Is this correct? I can confirm that I also have this issue. For the record I can confirm that the PIN is not changed in the Mysql DB. I think, however, that I don't have a uniqueid column. Maybe check this out: http://bugs.digium.com/view.php?id=8758 Thanks, - Jesse ___ --Bandwidth 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] ARA extensions ordering
Hello List, I am curious how the ordering of the extensions are determined for an ARA dial-plan. For example, if I have these: _9X. _9011. Which is selected first? Any number dialed starting with 9011 is matched by either rule here and I don't remember seeing any ORDER BY clauses when I had debugged the ARA queries. I'm sure I just missed some critical documentation here. Thoughts? THanks, - Jesse ___ --Bandwidth 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: Question about Mitel phones
Yes, the Mitel phones do have a Web interface for configuration. They also support mass-deployment scenarios with TFTP HTTP. You may want to check out these: http://sipdnld.mitel.com/ http://edocs.mitel.com/DB/5212_5224/WebConfigHelp_Admin_en_CA/WebHelp/ WebConfig.htm Thanks, - Jesse On Nov 10, 2006, at 10:35 AM, [EMAIL PROTECTED] wrote: Message: 9 Date: Fri, 10 Nov 2006 17:03:29 +0100 From: Christian [EMAIL PROTECTED] Subject: [asterisk-users] Question about Mitel phones To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Hi all, Does anyone know if the Mitel phone features a webintreface for configuring the phone? Many thanks, Christian -- Jesse Peterson [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] Mitel 5224/SIP no MWI
Does anybody know if this is supposed to work and if so, what, if any, workaround is needed? I have other phones (Snom, Polycom) MWI working with this system fine. 6.0.0.19 (latest) Mitel SIP firmware is loaded. Thanks for your time, - Jesse -- Jesse Peterson [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] Re: Mitel 5224/SIP no MWI [RESOLVED]
I've resolved this issue. It appears that somewhere between the firmware I was running and the latest firmware Mitel introduced multiple registrations User List Configurations in Mitel speak. It is here that the phone must have its Voice Mail Server configured - not in the global phone-message button area that I was configuring before and expecting to work. Thanks for your time, - Jesse On Tue, 10 Oct 2006 10:20:57 -0700 Jesse Peterson [EMAIL PROTECTED] wrote: Does anybody know if this is supposed to work and if so, what, if any, workaround is needed? I have other phones (Snom, Polycom) MWI working with this system fine. 6.0.0.19 (latest) Mitel SIP firmware is loaded. Thanks for your time, - Jesse -- Jesse Peterson [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] PRI TON/pridialplan digit prefixing
Hello all, I have a situation where I receive international calls from a legacy PBX into Asterisk. These calls have a PRI TON of International and do not have the (US) international prefix 011 in the number. I need to be able to either act on (in the dial plan) the PRI TON or be able to have the 011 re-added (prefixed) onto the called number so that I may route it properly. Is there a way to do this? This post: http://lists.digium.com/pipermail/asterisk-users/2005-May/102837.html Appears to suggest that this might be possible, but I was not able to get it working as suggested. Thanks for your time, - Jesse -- Jesse Peterson [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] Concurrents calls on asterisk with H323
Look for the recent 'capacity testing' thread here. We've had some discussions on it, but so far the bottom line sounds like you won't be able to run more than 20 - 25 decent quality calls before asterisk dies. jesse -Original Message-From: Cesar Rico [mailto:[EMAIL PROTECTED]Sent: Monday, January 19, 2004 09:28To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Concurrents calls on asterisk with H323 Hi all, I`ve installed succesfully asterisk wiht h323 protocol, I need kwon how many concurrenst call support asterisk working with h323 clients. My other questions is: I have a sound file in g.723.1 format in sound directory, my h323 clients have the g.723.1 codec but when I make a playback with that file (i.e. exten = 100,Playback(file.g7231)) I can`t hear nothing, in the handbook for asterisk say that asterisk is transparent for that codec, so I don`t know why no works well. I`ll appreciate you support Att. César Augusto Rico. image001.jpg
RE: [Asterisk-Users] Residential services
It will suite you well to fumble around with asterisk for several days and to keep
reading all the documentation tidbits you can find. That will really help get you
aquainted with asterisk and the support/documentation that is available. Most of the
good info I've found has come from the wiki and from reading through list archives.
Your dialplan question isn't too hard. I'll leave it up to you to make sure your
contexts are included/excluded correctly. Basically all you need to do is make sure
that
ignorepat = 9 is commented out (so ;ignorepat = 9) in any context that should use
normal dialing. Then you have 2 extensions, either this for 10 digit dialing
exten = _1NXXNXX,1, Dial, ${OUTTRUNK}
exten = _NXXNXX,1, Dial, ${OUTTRUNK}
or this for 7 digit:
exten = _1NXXNXX,1, Dial, ${OUTTRUNK}
exten = _NXX,1, Dial, ${OUTTRUNK}
of course you are also on your own to make sure ${OUTTRUNK} is working correctly.
I haven't done outbound to the PSTN with a Cisco setup, but I have with a Lucent setup
(mvam as gk and tnts for pstn access). I've had to use OH323 channel instead of the
included H323 channel. To get Asterisk to register w/ my gk as a gateway instead of a
terminal. Basically, I did this:
1) setup a gateway called 'ASTERISK' on my gk (mvam)
2) in oh323.conf:
gatekeeper = 192.168.0.50
[register]
alias = ASTERISK ;or whatever name you used on your gatekeeper
prefix = 999000;oh323 was complaining there were no prefixes registered... so
a bogus one fixed that. my gatekeeper seems to just ignore it anyway.
3) in my extensions.conf, I dial to OH3223/BYEXTENSION, so in my above 10digit
dialing example, it would be:
exten = _1NXXNXX,1, Dial, OH3223/BYEXTENSION
exten = _NXXNXX,1, Dial, OH3223/BYEXTENSION
the 'BYEXTENSION' part just take the number as matched by the exten option and passed
it out to the gatekeep OH323 has registered with and routes based on it's response.
Now, as far as residential services. Test, test, test. Then test a little more before
you decide you can run production services. In some non-formal testing, myself and
another fella using approx. the same setup (basically * as a sip-h323 gateway) have
found we could not run more than about 20-25 simultaneous calls. With typical telecom
oversubscription rates, you're not going to get very many customers on an asterisk box.
If you do manage to get it to handle a decent number of calls, I'm sure many people
would be interested in your configuration.
jesse
-Original Message-
From: Jeremy Jones [mailto:[EMAIL PROTECTED]
Sent: Monday, January 19, 2004 11:52
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Residential services
Hi folks,
The obligatory newbie disclaimer:
Hi, I'm new to Asterisk and I have a couple questions...
OK, now that that's over with:
I've just started working for a small CLEC, and I'm trying to sell * to
my boss as a replacement for some expensive/inflexible/closed-source
software he's been using to provide residential dialtone with for a
couple years now. Presently, we have:
1) a cluster of sun boxes running propriatary IP-PBX software
2) a cisco 3640 h323 gatekeeper
3) a cisco as5300 pstn gateway
I'd like to use sip between an asterisk box and that as5300 (which right
now is only speaking h323), and I'd like to be able to use sip, h323,
mgcp, or skinny for residential customers. This ought to be no problem,
right? I'm coming up with pretty much nil on documentation regarding
as5300 - asterisk configuration, however. And, while I'm sure I could
fumble through it for a couple days, I thought there just might be
someone out there who has a working configuration using an as5300 as a
pstn gateway with asterisk (either with sip, or with h323 via a cisco
gatekeeper).
Now, regarding residential services in particular...
The configuration files examples I've found all assume a business
environment, where you'd dial a 9 for outside lines. Anyone have an
example config where an endpoint gets dumped directly to the pstn when
they pick up the phone?
Thnaks,
Jeremy Jones
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RE: [Asterisk-Users] capacity testing
Title: RE: [Asterisk-Users] capacity testing
1)
Yes, I did get that. I've never seen a segmentation fault message, but that
should be b/c I've been running the process in the background since it is
obviously seg-faulting. I believe you are also correct that most people are not
trying to put the load on it that we are.
2) I
always see 'safe_asterisk' and 'asterisk -vvvg' running. my monitoring was
always done with top, but I've checked w/ ps a couple times and I believe only
ever see 1 of each of those processes. I may have to do some tests again to
double check that. My CPU problems did not come until the last 10 - 30 seconds
before asterisk crashed. This is still odd that our memory processor
observations are opposite... the next thing I'm going to try is a dual xeon pIII
800 or 1ghz machine to see what happens.
3) I'm
running oh323. It was the one I could get to register w/ my gatekeeper as a
gateway - that made it much easier for me to do call routing on both sides. I
have also noticed some inconsistencies in the call flows like you mention, but
haven't taken the time yet to pinpoint exactly what and when they are
happening.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of T.
ChanSent: Thursday, January 15, 2004 22:54To:
[EMAIL PROTECTED]Cc: Alan ChanSubject: RE:
[Asterisk-Users] capacity testing
Hi
all, and Jesse
1.
So, you did get the experience of crashing all of a sudden with the
"Disconnected from Asterisk server" error message. I got both this and the
segmentation error when crashing. I am running the version of asterisk, libpri
and zaptel updated to about 5 days ago, but I have had tested Asterisk for
more than a month already and needless to say I have had this experience since
Day 1, meaning it has always been a problem even in the previous revisions.
Henceforth, I feel that it is an intrinsic Asterisk problem, rather than just
the problem with specific versions / revisions. I have posted this problem a
few times before, I feel that this is a major problem but surprisingly, I was
not getting any feedback at all. I have this feeling that more than 90% of the
Asterisk community is using the system for PBX application rather than VOIP,
may be, just may be, Asterisk has not been tested with a good number of
simultaneous calls.
2. I
am using Xeon 2.6G chip, much more powerful than yours, I have not got any
problem with CPU usage, at least not during the time that I was watching. The
thing is when I start 'safe_asterisk' , I could see when doing a PID, 1
"safe_asterisk" PID session and at least 10 (or more especially when there are
more calls) "asterisk -vvvg -c" PID session. Each session takes up about 18M
to 20M RAM, when that is why I am seeing all very high memory usage. How many
sessions of Asterick do you see running after you loaded it?
3.
Are you running H323 (Jeremy) and OH323 (Michael)? I am running Jeremy's and
have had this inbound H323 problem. I tried OH323 (Michael) as well, but for
some reasons, I am getting this false connect signal, that is, I made an
outbound H323 call to a CiscoAS5300 for example, I heard the ring and
immediately on my "Asterisk", it showed call answered when it was still
ringing. Do you have that experience?? What setting you have if you do not
have that experience?
4.
Lets talk off list at [EMAIL PROTECTED].
Thanks
Tom
-Original Message-From: Jesse Peterson
[mailto:[EMAIL PROTECTED]On Behalf Of Jesse
PetersonSent: Thursday, January 15, 2004 8:21 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users]
capacity testing
Sorry for the malformed mail. My responses are marked with
'***' below.
jesse
==
Hi,I am a newbie in Asterisk as well, intending to
use it in a similar way asyou are, communicating with AS5300 as well as
other gateways includingMAXTNT.I have had similar, but yet
different experiences than yours.1. Asterisk does crash with the
number of calls, but in my case, about orless than 20 calls, then I
would get either a Segmentation Error and thencrashed OR it would just
crash saying "Disconnected from Asterisk server"all of a
sudden.
*** The crashesI experienced were fairly transparent. When I had
the console (asterisk -r) running, I saw the 'Disconnected' message you
mention.2. I am using Pentium Xeon chip and hence more powerful than
yours with 512MRAM, my CPU usage has always been low, however, I have
not had a chance tolook at the CPU usage just before crashing, but all
the time that I waslooking, it has been low. Rather the MEMORY has
always remained high at 450Musage even with no calls. This is a
different experience as compared toyours.*** A Xeon of the same
speed (
[Asterisk-Users] capacity testing
Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip-h323 gateway to tie SIP H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past week (maybe 01/09/04 and 01/14/04) and have not been able to get them to work semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 has supported those ok. Primarily test cases have involved sending ip phone calls via SIP to Asterisk and having Asterisk route the calls using h323 via a gatekeeper to my TNT network which then sends it out the PSTN... and the opposite path, PSTN-TNT-Asterisk-SIP Phone. Another test has been sending a call from a AS5300 using SIP to Asterisk, out H323 to a TNT. Both of those have worked very well with the voice quality being excellent (actually better than a SIP-ISDN T1 hardware solution we've been working with - audiocodes mediant 2k for those interested). This is the test case I describe below as it was the one the allowed me to load Asterisk up with the most calls. Anyway, I know that what I'm doing is not exactly the intended primary use of Asterisk. That said, here's what I found. Voice quality was very good until I had approx. 25 calls up. At that point there were intermittent issues with garbled voice, a little echo, etc. When it reached a little over 30 calls, Asterisk just died (oops). During the test, I was trying to keep an eye on proc. memory util. Memory never seemed to be an issue - even right before the crash the Asterisk process was not using more than 20 - 25MB. Processor utilization was interesting to watch though. I couldn't make any direct/firm correlation, but it seemed like my spikes were coming when Asterisk was doing call setup. Even up to about 25 calls, utilization didn't spike to more the 25% for long, and with ~25 calls seemed to 'idle' around 15%. Above the 25 (when also started noticing voice quality issues), the proc. util. seemed to start going wacky - spikes up to 40, 50, even 60%. Then it went to 99% for a moment, voice quality was horrible if you could hear anything, and Asterisk crashed. I did not find anything in the logs to inidicate any problems, though I've found that to be the case pretty much everytime Asterisk crashes. I saw a list thread in which a developer asked for some gdb output... in it, he said this: Run asterisk with -vvvcg. Do your test (core file generated). Run gdb /usr/sbin/asterisk core_filename From within gdb run bt and send me the output of it. if it is of use, here it is (from asterisk v.0.5.0) - (gdb) bt #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504 #2 0x0805884f in ast_write (chan=0x8214488, fr=0x5de5c4a8) at channel.c:1385 #3 0x0805afa1 in ast_channel_bridge (c0=0x5de5c4a8, c1=0x0, flags=0, fo=0x6ef20e50, rc=0x6ef20e54) at channel.c:2262 #4 0x418bdd7a in ast_bridge_call (chan=0x5de5ed98, peer=0x8214488, allowredirect_in=0, allowredirect_out=0, allowdisconnect=0) at res_parking.c:224 #5 0x41d6bfeb in dial_exec (chan=0x5de5ed98, data=0x41d6d19b) at app_dial.c:668 #6 0x08061a5a in pbx_exec (c=0x5de5ed98, app=0x80f0f98, data=0x6ef216e8, newstack=1) at pbx.c:396 #7 0x08068c61 in pbx_extension_helper (c=0x5de5ed98, context=0x5de5eeec longdistance, exten=0x8214488 H323:8257, priority=2, callerid=0x5de10048 \Jesse Peterson\ 2474766, action=1104606132) at pbx.c:1150 #8 0x0806392c in ast_pbx_run (c=0x41d6f3b4) at pbx.c:1634 #9 0x08069321 in pbx_thread (data=0x84a5038) at pbx.c:1855 #10 0x40026484 in start_thread () from /lib/tls/libpthread.so.0 - If anyone has tried something like this or has any comments, I'd be interested in hearing from them. jesse ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] capacity testing
I did initially, but I was having problems (possibly just in thinking it through) getting the provided h323 driver to either a) register as a gateway with my gatekeeper - that just does not seem to be and option (please correct me if I'm wrong!!!) or b) setup a 'variable' extension (yes, extensions.conf) that would allow me to route any number to it. jesse -Original Message- From: Alastair Maw [mailto:[EMAIL PROTECTED] Sent: Thu 1/15/2004 5:17 PM To: [EMAIL PROTECTED] Cc: Subject: Re: [Asterisk-Users] capacity testing On 15/01/04 19:39, Jesse Peterson wrote: #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504 Do you experience the same problems when you use the other (bundled) h323 driver? (asterisk/channels/h323/README for instructions) Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
RE: [Asterisk-Users] capacity testing
Sorry for the malformed mail. My responses are marked with '***' below. jesse == Hi, I am a newbie in Asterisk as well, intending to use it in a similar way as you are, communicating with AS5300 as well as other gateways including MAXTNT. I have had similar, but yet different experiences than yours. 1. Asterisk does crash with the number of calls, but in my case, about or less than 20 calls, then I would get either a Segmentation Error and then crashed OR it would just crash saying Disconnected from Asterisk server all of a sudden. *** The crashes I experienced were fairly transparent. When I had the console (asterisk -r) running, I saw the 'Disconnected' message you mention. 2. I am using Pentium Xeon chip and hence more powerful than yours with 512M RAM, my CPU usage has always been low, however, I have not had a chance to look at the CPU usage just before crashing, but all the time that I was looking, it has been low. Rather the MEMORY has always remained high at 450M usage even with no calls. This is a different experience as compared to yours. *** A Xeon of the same speed (800mhz in my case) should certainly perform better - lower, I don't know. I find it a little odd that you experienced basically the opposite of my testing. What version are you running? 3. I have also noticed that with more calls, and after a certain random period of time, any H323 calls going into the Asterisk would fail, my AS5300 and MAXT TNT would get their calls all rejected from Asterisk. However, Asterisk was still running at the time and I could actually call in and out the zap interface and outbound H323 from Asterisk was not a problem. It seems that something got hung with H323, causing inbound H323 calls into Asterisk to all fail. In this situation, I would have to stop the Asterisk and rerun it to fix the problem. *** Interesting - I have not experienced that (yet...). 4. I have not tried the 0.7.0 version, but with existing version, I am not getting reliable and stable system, nothing close to Cisco and Lucent which are rock solid. However, I really love the power and the features of Asterisk, and I remain in good faith to see improvements. Any associate out there who can shed some lights into this? I am rather curious as to why I seem to be using up all memory although I am not running any unnecessary processes, or should I actually disable all modules, other than really necessary ones to support VOIP? *** Since you and I are working in what sounds to be a familiar environment, maybe we should communicate about our test scenarios, etc off list to both help each other and see if we can find some similarities to share with others. Thanks ! Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jesse Peterson Sent: Thursday, January 15, 2004 2:40 PM To: Asterisk-Users (E-mail) Subject: [Asterisk-Users] capacity testing Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip-h323 gateway to tie SIP H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past week (maybe 01/09/04 and 01/14/04) and have not been able to get them to work semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 has supported those ok. Primarily test cases have involved sending ip phone calls via SIP to Asterisk and having Asterisk route the calls using h323 via a gatekeeper to my TNT network which then sends it out the PSTN... and the opposite path, PSTN-TNT-Asterisk-SIP Phone. Another test has been sending a call from a AS5300 using SIP to Asterisk, out H323 to a TNT. Both of those have worked very well with the voice quality being excellent (actually better than a SIP-ISDN T1 hardware solution we've been working with - audiocodes mediant 2k for those interested). This is the test case I describe below as it was the one the allowed me to load Asterisk up with the most calls. Anyway, I know that what I'm doing is not exactly the intended primary use of Asterisk. That said, here's what I found. Voice quality was very good until I had approx. 25 calls up. At that point there were intermittent issues with garbled voice, a little echo, etc. When it reached a little over 30 calls, Asterisk just died (oops). During the test, I was trying to keep an eye on proc. memory util. Memory never seemed to be an issue - even right before the crash the Asterisk process was not using more than 20 - 25MB. Processor utilization was interesting to watch though. I couldn't make any direct/firm correlation, but it seemed like my spikes were coming when Asterisk was doing call setup. Even up to about 25 calls, utilization didn't spike to more the 25% for long, and with ~25 calls seemed to 'idle' around 15%. Above the 25 (when also started noticing voice quality
RE: [Asterisk-Users] capacity testing
I know, but as I mentioned in the inital post, I haven't been able to get the last 2 cvs versions I've pulled to run stable enough to test. I've seen a 0.7.0 version number mentioned. Is there newer, mostly stable version of code I should try that just hasn't been officially released? jesse -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Thu 1/15/2004 10:11 PM To: [EMAIL PROTECTED] Cc: Subject: Re: [Asterisk-Users] capacity testing Jesse Peterson wrote: I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. CVS UPDATE! That code is hardcore old. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
