Re: [asterisk-users] Fun with virtual asterisks ...
In article pine.lnx.4.64.1002261731010.11...@unicorn.drogon.net, Gordon Henderson gordon+aster...@drogon.net wrote: So I've been testing asterisk under LXC for a few days now and am very happy with the results. My test server is a 1.8GHz Celeron with 256KB cache and 512MB RAM and I have 20 containers each running asterisk (and apache/php,sendmail and a few other minor things) Hi Gordon, sounds good fun. I hadn't heard of LXC until you mentioned it recently. Have you done any experiments with Meetme yet? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fun with virtual asterisks ...
On Sat, 27 Feb 2010, Tony Mountifield wrote: In article pine.lnx.4.64.1002261731010.11...@unicorn.drogon.net, Gordon Henderson gordon+aster...@drogon.net wrote: So I've been testing asterisk under LXC for a few days now and am very happy with the results. My test server is a 1.8GHz Celeron with 256KB cache and 512MB RAM and I have 20 containers each running asterisk (and apache/php,sendmail and a few other minor things) Hi Gordon, sounds good fun. I hadn't heard of LXC until you mentioned it recently. Have you done any experiments with Meetme yet? Will be doing some later today when I get more phones to try it with (using Page()) I could setup something that creates 20 extensions on each PBX and 19 SIP trunks to register to every other PBX, then have them all do a meetme, but I don't fancy hand-editing 20 sets of config files - yet. Maybe I'll see if I can write a set of scripts to do it! LXC only has one kernel on the processor, so one copy of dhadi_dummy loaded - it really is just like running X copies of asterisk on the same processor with their network isolated from each other (and a few other things) - so there's no overhead of a hypervisor schedulling multiple kernels each schedulling their own jobs. I did run 20 copies of dhadi_test and they were all in the 99.996% range for a few hours. Asterisk doesn't seem to be able to set real-time mode (-p option) so I'm not sure if that will have much effect. It's probably an artifact of the way the LXC containers are setup, but there's not much documentation on it. I'll need to do a bit more reading/testing before I put this into production - which I fully intend to do now. Cheers, Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fun with virtual asterisks ...
On Fri, 26 Feb 2010, Gordon Henderson wrote:
More for fun than anything else, I've tried daisy-chaining instances
together - so 20 asterisks running on the same host, 0 calls 1, 1 calls 2,
2 - 3... 19 calls 0 - which plays music on hold.
Put a call into 0 - setup through the chain seems to take less than half a
second and I hear MoH... RTP is passing through each instance as
canreinvite is set to no. CPU usage was about 6%.
More plumbing fun:
Phone - pbx0 - pbx1 - ... pxb19 -
pbx0 - pbx1 - ... pbx19 -
pbx0 - pbx1 - ... pbx19 -
pbx0 - MoH
That's it's limit. when I loop it once more I get audio breakup. So 20
Asterisks handling 3 calls + 1; 61 calls, each handling full media.
dialplan on node 19:
exten = 666,1,Noop(Thrash Test - Turn ${thrashCounter})
exten = 666,n,GotoIf($[${thrashCounter} = 2]?gameOver)
exten = 666,n,SetGlobalVar(thrashCounter=$[${thrashCounter}+1])
exten = 666,n,Dial(SIP/6...@dsx0-out)
exten = 666,n,Hangup()
exten = 666,n(gameOver),Dial(SIP/*...@dsx0-out)
channels on 0:
dsx0*CLI show channels
Channel Location State Application(Data)
SIP/200-004d *...@internal:4 Up MusicOnHold(default)
SIP/dsx1-out-004 (None) Up AppDial((Outgoing Line))
SIP/200-004b 6...@internal:2 Up Dial(SIP/6...@dsx1-out)
SIP/dsx1-out-004 (None) Up AppDial((Outgoing Line))
SIP/200-0049 6...@internal:2 Up Dial(SIP/6...@dsx1-out)
SIP/dsx1-out-004 (None) Up AppDial((Outgoing Line))
SIP/299-0047 6...@internal:2 Up Dial(SIP/6...@dsx1-out)
7 active channels
4 active calls
CPU usage (From top) on the host node:
Cpu(s): 26.7%us, 40.8%sy, 0.0%ni, 0.6%id, 0.3%wa, 0.0%hi, 31.6%si, 0.0%st
So not much actual user CPU, but a lot in the system and handling
interrupts. This is likely the networking layer and just the overhead of
running 20 live programs I suspect. I'm using Ethernet Bridging to do the
underlying network plumbing - each node sees an eth0 as it would do if
it were on a standalone server (and having used the bridge code in the
past for a transparent traffic shaper, I think it's very efficient, but
obviously there will be overhead)
I am a little dissapointed that it wouldn't do more, but I suspect that's
the overhead of simply running 20 virtual asterisks rather than one - I
know I can exceede that number of calls handling full media on a single
500MHz CPU... However, this is an old 1.8GHz Celeron... I'm going to order
up a pair of nice fast dual core server boxes for production use so will
see how it behaves there.
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[asterisk-users] New patch for app_queue to show all call attempts, even failing ones
Hi, I've just uploaded a patch here: https://issues.asterisk.org/view.php?id=16925 This patch introduces a new parameter; congestion to both RINGNOANSWER in queue_log and AgentRingNoAnswer AMI event, which is set to 1 if the call failed to go through because of technical difficulties. And it also is more verbose than app_queue has been earlier, since app_queue usually silently ignores channel problems with its agent members. With this patch, it is easier to make statistics out of queue_log with information about problems with an agent. For example if an agent has a faulty line, or your telco/dahdi connection is having problems. Please come with comments about this patch, and help test it if you agree with the idea. Regards, Håkon Nessjøen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems installing dahdi : kernel sources
On Thu, Feb 25, 2010 at 04:30:03PM +0100, jonas kellens wrote: Hello list, when installing Dahdi, the following error comes up : You do not appear to have the sources for the 2.6.18-164.11.1.el5xen kernel installed. make[1]: *** [modules] Error 1 The running kernel version : -bash-3.2# uname -a Linux vds.hosting.net 2.6.18-164.11.1.el5xen #1 SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux -bash-3.2# ls /usr/src/kernels/ 2.6.18-164.11.1.el5-x86_64 Isn't the kernel the same as the sources ?? Package kernel-devel-2.6.18-164.11.1.el5.x86_64 already installed and latest version Package kernel-headers-2.6.18-164.11.1.el5.x86_64 already installed and latest version No. Install kernel-xen-devel -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Increasing the dahdi chunk size with Sangoma cards
Hi, does anyone run non HWEC Sangoma PRI cards with an increased dahdi chunk size? I tested it at 2ms and it seemed fine with no noticeable loss in audio quality, and it reduced the interrupt processing to 50%. Regards Lee -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk AUTHENTICATE Command
I was surprised to find that you can not override the default voice prompts when using the Authenticate Command. I have viewed the source and the prompt file names are hard coded. I am developing an application that use the Authenticate command to use one use PINs located in the Asterisk Database. I had all of the recordings for the application recorded by a professional voice talent. It seems odd that I can not specify an option when calling the Authenticate command to provide my own recordings. The only 2 work arounds that I have come up with are - change the system recordings (which will change them for any application that uses the same recordings). Or I could write my own application to authenticate the PINs and not use the authenticate command. Does anyone else think that this feature would be useful or am I looking at this the wrong way? Thanks. Matthew A. Kolberg www.justask.net mkolb...@justask.net ___ Just ask for ASK Taking the hassle out of technology so you can run your business. 1-877-ASK-4-ASK Did you know you can chat online with us? http://www.justask.net/support Follow ASK on twitter at: http://twitter.com/justasknet-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redirect call based on CLI???
Or, alternatively using the 's' priority...
exten = 845,1,Verbose(3, Incoming call from ${CALLERID(all)}) ; this will
be priority 1
exten = 845/12345678,n,Goto(blacklist) ; the n will make this priority 2
exten = 845/23456789,s,Goto(blacklist) ; the s will make this also priority
2
exten = 845/34567890,s,Goto(blacklist) ; again, priority 2
exten = 845/09876543,s,Goto(whitelist) ; and again
exten = 845/98765432,s,Goto(whitelist) ; and again
exten = 845/87654321,s,Goto(whitelist) ; and again
exten = 845,s,Verbose(3, CLI (${CALLERID(num)}) is neither blacklisted or
whitelisted) ; last s, so last priority 2, this time with no pattern
exten = 845,n,Dial(SIP/somewhere,180) ; this will be priority 3
exten = 845,n,Hangup() ; priority 4
exten = 845,n(blacklist),Verbose(3, CLI (${CALLERID(num)}) is blacklisted)
; priority 5
exten = 845,n,Hangup(21) ; priority 6, cause code 21 = rejected
exten = 845,n(whitelist),Verbose(3, CLI (${CALLERID(num)}) is whitelisted)
; priority 7
exten = 845,n,Dial(SIP/somewhereelse,180) ; priority 8
exten = 845,n,Hangup() ; priority 9
You can see the priorities from the asterisk cli by doing a 'dialplan show
context'
d
On 26 February 2010 01:33, Mark Hulber asterisk.ad...@hulber.com wrote:
Since you are using 'n' notation, you might not have your statements
aligned. You can label your statements as below:
exten = s,n,Answer
exten = s,n,GotoIf($[${CALLERID(name)} != UNAVAILABLE]?ans)
exten = s,n,Set(CALLERID(name)=${CALLERID(number)})
exten = s,n(ans),NoOp
; Banned
exten = s/708857500X,ans+1,Goto(banned,1) ;
exten = s/9044898017,ans+1,Goto(banned,1) ;
exten = s/8883222785,ans+1,Goto(banned,1) ;
; Specifically routed
exten = s/9165553456,ans+1,Goto(markivr,1) ; Allowed
exten = s/19165553456,ans+1,Goto(markivr,1) ; Allowed
; Default
exten = s,ans+1,Goto(mainmenu,s,1)
On 2/25/2010 10:11 AM, Brian wrote:
On Thu, 2010-02-25 at 03:00 -0800, Kyle Kienapfel wrote:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
Has example
exten = s,1,Answer
exten = s/9184238080,2,Set(CALLERID(name)=EVIL BASTARD)
exten = s,2,Set(CALLERID(name)=Good Person)
exten = s,3,Dial(SIP/goodperson)
for white list
exten = s/123123123,1,Dial(SIP/phoneA)
exten = s/456456456,1,Dial(SIP/phoneA)
exten = s,1,Dial(SIP/phoneB)
Thanks Kyle.
I tried the example given but I could not get this to work - basically
if I dial it from any phone that does not match 0800800800 (for
illustration) it hangs up the channel with an error.
exten = 845/0800800800,n,Set(CALLERID(name)=EVIL BASTARD)
Auto fallthrough, channel 'SIP/1000-0017' status is 'UNKNOWN'
I'm struggling to work out the logic here of a non-match, but this was
not caught by i or s in error, so I'm probably missing some brain
connection here.
However, I've managed to do what I want using gotoif statements matching
caller id - but I'd be interested to work out how the above is meant to
branch on a non-match.
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[asterisk-users] Conference Calling
Hey All, I want to implement a conference calling scenario. Conference Call Procedure:User1 dial the User2. When call is connected put the current call on Hold and dial User3. When the call is connected between User1 and User3 join the User2 in a conference room!How I can implement this scenario. What are generic steps to do so! Thanks=Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference Calling
Here is where to get you start with this. http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO -Tri From: Faheem faheem_...@yahoo.com To: asterisk-users@lists.digium.com Sent: Sat, February 27, 2010 12:08:24 PM Subject: [asterisk-users] Conference Calling Hey All, I want to implement a conference calling scenario. Conference Call Procedure: User1 dial the User2. When call is connected put the current call on Hold and dial User3. When the call is connected between User1 and User3 join the User2 in a conference room! How I can implement this scenario. What are generic steps to do so! Thanks = Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk n-way DTMF detection
I have figured this out and it's working fine now. Here is the key if anyone has the same issue. featuredigittimeout = 5000 Increase the digit timeout or you have to press the key codes quick enough in order for Asterisk to detect the keys. -Tri From: Tri Tu mtr...@yahoo.com To: asterisk-users@lists.digium.com Sent: Wed, February 24, 2010 10:38:25 PM Subject: [asterisk-users] Asterisk n-way DTMF detection Hello, I have setup the n-way conferencing with Asterisk and it's working when I use with my budgetone 100 phone but it doesn't work for any of the voip software or other ATA that I have. When I turned the debug on, I see that the correct keys (*0) were entered but asterisk doesn't detect the signal to trigger the features event. I have set a test extension to get the input dtmf key and say the digit out. They are getting correctly on the IVR but when using n-way conferencing, it's not taking it. Here is the output of testing DTMF with IVR. v103*CLI v103*CLI -- Executing [...@from-internal:1] Read(SIP/-b6807538, digito||10) in new stack -- Accepting a maximum of 10 digits. * DTMF-relay event received: 8 * DTMF-relay event received: 5 * DTMF-relay event received: 2 -- User entered '852' -- Executing [...@from-internal:2] SayDigits(SIP/-b6807538, 852) in new stack -- SIP/-b6807538 Playing 'digits/8' (language 'en') -- SIP/-b6807538 Playing 'digits/5' (language 'en') -- SIP/-b6807538 Playing 'digits/2' (language 'en') -- Executing [...@from-internal:3] Hangup(SIP/-b6807538, ) in new stack == Spawn extension (from-internal, 88, 3) exited non-zero on 'SIP/-b6807538' -- Executing [...@from-internal:1] Macro(SIP/-b6807538, hangupcall) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/-b6807538, vw) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/-b6807538, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/-b6807538, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/-b6807538, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/-b6807538, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/-b6807538, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/-b6807538' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/-b6807538' v103*CLI bash-3.1# anh here is the console log of the Asterisk when pressing the key during callerA is on the phone with CallerB. v103*CLI v103*CLI * DTMF-relay event received: * * DTMF-relay event received: 0 v103*CLI Wondering that if anyone know what could be wrong here. My asterisk version is Asterisk 1.4.20. -Tri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X-Lite won't register
Turn debug on and watch on the console to see if the you see the x-lite extension talks to your asterisk box. CLI core set debug or CLI core set verbose 99 From: Girard, Jeffrey COL MIL USA jeffrey.gir...@us.army.mil To: asterisk-users@lists.digium.com Sent: Thu, February 25, 2010 6:35:52 AM Subject: [asterisk-users] X-Lite won't register Beginner to Asterisk, but not beginner to VoIP FreePBX front end running on a dell 1550 and XLite running on a different Woindows XP box Both boxes connected via switch on same subnet. No NAT involved On FreePBX I created a new extension 1001 with a SIP password of 1001 On Xlite, username is 1001, password is 1001, authorization user name is 1001, and domain is IP of Free PBX XLite tries to register then shows 408 error registration timeout Windows box pings Asterisk and firewall is disabled on XP machine What am I missing? Jeff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk AUTHENTICATE Command
On Sat, 27 Feb 2010, Matthew A Kolberg wrote:
I was surprised to find that you can not override the default voice
prompts when using the Authenticate Command. I have viewed the source
and the prompt file names are hard coded. I am developing an
application that use the Authenticate command to use one use PINs
located in the Asterisk Database. I had all of the recordings for the
application recorded by a professional voice talent. It seems odd that
I can not specify an option when calling the Authenticate command to
provide my own recordings.
The only 2 work arounds that I have come up with are - change the system
recordings (which will change them for any application that uses the
same recordings). Or I could write my own application to authenticate
the PINs and not use the authenticate command.
Does anyone else think that this feature would be useful or am I looking
at this the wrong way?
I have not tried this with the authenticate() application, but you can
trick Asterisk into using your own prompts by [ab]using the LANGUAGE()
function.
Something like:
exten = *,n,set(LANGUAGE()=my-pin-prompts)
or
exten = *,n,set(LANGUAGE()=my-pin-prompts/${CLIENT})
may work.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] No RTP from asterisk?
I've got an asterisk installation of 1.4.30-RC2 running, and while I can register lines and get call setup to pass, for some reason no RTP is being generated or received by asterisk. Debug doesn't seem to give me too much of relevance about it, especially rtp debug. I had a few other small issues, like trying to negotiate G729 when it's not capable, but since then, I've changed everything back to G711. I have connected to it, a SIP trunk, 3 registered users and I'm at a loss as to how to troubleshoot this further. Can anyone point me in the right direction? Peter -- Peter Serwe http://truthlightway.blogspot.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference Calling
Muhammad It is not really your scenario but the scenario to setup a conference call with three numbers could be to generate two call files that points to a local channel/a context/extension that route the leg into the conference room and have your own leg routed into the conference room after the input is done This not the solution but one of the many possible. enter the numbers for setting up the conference call like number1*number2 (check Read() cmd for storing input into a variable) split the input in seperated numbers See http://www.voip-info.org/wiki/index.php?page=Asterisk+variables generate the call files for setting up the connection. Point to a context, extension, priority to route the lef into a conference room. See http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out move the call files to /var/spool/asterisk/outgoing (check System() cmd ) have your own leg routed into the conference room (check Goto() cmd ) Have a nice chat with the three of you ;-) Erik On 27 feb 2010, at 21:08, Faheem wrote: Hey All, I want to implement a conference calling scenario. Conference Call Procedure: User1 dial the User2. When call is connected put the current call on Hold and dial User3. When the call is connected between User1 and User3 join the User2 in a conference room! How I can implement this scenario. What are generic steps to do so! Thanks = Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redirect call based on CLI???
On Sun, 2010-02-28 at 04:00 +0800, D Tucny wrote:
Or, alternatively using the 's' priority...
exten = 845,1,Verbose(3, Incoming call from ${CALLERID(all)}) ; this
will be priority 1
exten = 845/12345678,n,Goto(blacklist) ; the n will make this
priority 2
exten = 845/23456789,s,Goto(blacklist) ; the s will make this also
priority 2
exten = 845/34567890,s,Goto(blacklist) ; again, priority 2
exten = 845/09876543,s,Goto(whitelist) ; and again
exten = 845/98765432,s,Goto(whitelist) ; and again
exten = 845/87654321,s,Goto(whitelist) ; and again
exten = 845,s,Verbose(3, CLI (${CALLERID(num)}) is neither
blacklisted or whitelisted) ; last s, so last priority 2, this time
with no pattern
exten = 845,n,Dial(SIP/somewhere,180) ; this will be priority 3
exten = 845,n,Hangup() ; priority 4
exten = 845,n(blacklist),Verbose(3, CLI (${CALLERID(num)}) is
blacklisted) ; priority 5
exten = 845,n,Hangup(21) ; priority 6, cause code 21 = rejected
exten = 845,n(whitelist),Verbose(3, CLI (${CALLERID(num)}) is
whitelisted) ; priority 7
exten = 845,n,Dial(SIP/somewhereelse,180) ; priority 8
exten = 845,n,Hangup() ; priority 9
You can see the priorities from the asterisk cli by doing a 'dialplan
show context'
Thanks. I've got it doing just what I want now :-) Appreciate your help.
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Re: [asterisk-users] No RTP from asterisk?
RTP is only firewall issue. Make sure that you can pass traffic from your client to the asterisk server. If it's on the same LAN, there shouldn't be any issue with RTP unless the Asterisk is setup with firewall to block RTP traffic (default is from 1 - 2 upd) Asterisk doesn't support G29 (pass-through is OK) but if you want to connect from your client to asterisk server with G729, you need to buy license. Using G711 is free and it taking about 68kbp. From: Peter Serwe peter.se...@gmail.com To: asterisk-users@lists.digium.com Sent: Sat, February 27, 2010 12:42:56 PM Subject: [asterisk-users] No RTP from asterisk? I've got an asterisk installation of 1.4.30-RC2 running, and while I can register lines and get call setup to pass, for some reason no RTP is being generated or received by asterisk. Debug doesn't seem to give me too much of relevance about it, especially rtp debug. I had a few other small issues, like trying to negotiate G729 when it's not capable, but since then, I've changed everything back to G711. I have connected to it, a SIP trunk, 3 registered users and I'm at a loss as to how to troubleshoot this further. Can anyone point me in the right direction? Peter -- Peter Serwe http://truthlightway.blogspot.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Server response time
Hey Guys, I am considering leasing a new server in Germany to run my Asterisk infrastructure and I was wondering how response time would affect the performance of the system. Right now I have a response time of around 60-70ms with my server in California. The server in Germany would have a response time of around 140ms (both ways). My DID/Termination providers are in Canada and the USA, and all my voip boxes are also in the USA. Any suggestions or recommendations? Thanks in advance! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
