Re: [asterisk-users] Asterisk V/s FreeSwitch

Thanks for reply and share your techniques, dialplans and knowledge on this thread. But my question was not related to load-balancing. I want to know , Why freeSwitch can preferred with compare to Asterisk(Call base , quality base)? And what is architecture difference between them. I am totally a

Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.

nobody facing any issue with this or nobody using real time architecture On Thu, Feb 9, 2012 at 10:54 AM, DHAVAL INDRODIYA wrote: > Hi Group. > > I am facing an issue with Peer registration in my asterisk server . > > I am using asterisk version 1.8.5.0 and using SIP real-time > architecture.when

Re: [asterisk-users] TE410P (1st) without cables always green

Shaun, > Just thinking out loud here but I'm guessing it may be fair to just set alarmdebounce to 0 by default on gen1 cards. > With dahdi-linux 2.2.0.2 does your card function if you set alarm debounce to 2500? /etc/init.d/dahdi stop sleep 3 modprobe dahdi modprobe wct4xxp debug=1 alarmde

[asterisk-users] Asterisk 10.1.2 Now Available

The Asterisk Development Team has announced the release of Asterisk 10.1.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 10.1.2 resolves several issues reported by the community and would have not been possible wi

[asterisk-users] Asterisk 1.8.9.2 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.9.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.2 resolves several issues reported by the community and would have not been possible

[asterisk-users] Turning off splash ring on PAP2T

Hi all, I'd like to know how I can turn off the "splash ring" voicemail waiting indication on a PAP2T from the provisioning XML file. I can do it from the web interface, but I need to do it on "a lot" of machines TIA, -- Take care and have fun, Mike Diehl. -- __

Re: [asterisk-users] Asterisk V/s FreeSwitch

On 02/09/2012 01:17 PM, Danny Nicholas wrote: If the MOH thing is really true, a more "realistic" test would be to run playback(demo-instruct). Since I know that I will eventually cross this bridge in real life/real time, I devised this test on my Asterisk 10.0 box Dialplan (in default context)

[asterisk-users] T.38 Incoming Fax Problem

Hello, I've installed the free (one user) Fax for Asterisk (FFA) license. Outgoing faxes, using T.38, to the PSTN work quite well. However, incoming faxes, do not seem to detect tones and certainly do not switch to T.38. The call drops as soon as the fax answers. Since I am using FreePBX (2.

Re: [asterisk-users] Asterisk V/s FreeSwitch

If the MOH thing is really true, a more "realistic" test would be to run playback(demo-instruct). Since I know that I will eventually cross this bridge in real life/real time, I devised this test on my Asterisk 10.0 box Dialplan (in default context) exten => 3366,1,answer() exten => 3366,n,playba

[asterisk-users] Problem with SIP phone outside local network

I am having a strange problem with an external SIP phone. It can register and receive calls but it cannot initiate any calls. A softphone on the same network works without problems. As far as I can notice the difference is that the hard phone is not sending the proper contact inf

Re: [asterisk-users] TE410P (1st) without cables always green

On Thu, Feb 09, 2012 at 02:11:12PM -0200, Marcio Gomes wrote: > Hello Shaun, > > 1) dahdi-linux-complete-2.2.0.2+2.2.0 , is generatiing 1K int/s > after patches.. > > 2) Looking in 2.2.1 code, i see to wct4xxp.c the alarmdebounce set > in 2500 and in 2.2.0 the alarmdebounce seted to 0, > > I loa

Re: [asterisk-users] Asterisk V/s FreeSwitch

We have used in production Asterisk 1.4 to do 3,000 concurrent calls at about 80 CPS without media going through the system. This is on a vmware ESXi server. The server is a Dell R610 with 2 X5670 (6 cores each at 2.93 GHz so 12 physical, 24 logical cores). Each vm gets 2 cores and 2 GB of RAM. We

Re: [asterisk-users] Early Media configuration doesn't seem to be working

Hi, on a similar setup I set in sip.conf: prematuremedia=no progressinband=never in the peers configuration. With this config you tell Asterisk not to handle inband information at all. But: Maybe you won't get any inband "error messages" also. Greetings from Wuppertal Max Grobecker Am 07.02.

Re: [asterisk-users] checking if a phone number is UP

Markus No we do checks ahead of line count checks in the dialplan code. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Markus" Sent: Thursday, February 09, 2012 12:08 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-

Re: [asterisk-users] Automatic Number Identification and anonymous calls

Hello, I know about the german phone system that the sense of an anonymous call is, that the called party has no way to get the caller's number in any way. The last switch honours the "anonymous" bit and removes the phone numbers before sending the call to the called party. In EURO-ISDN you have

Re: [asterisk-users] checking if a phone number is UP

Thanks for the detailed followup. Inbound reliability improvement is on our 2012 goals. When you place your outbound test call, are you mindful of the carrier it goes on, do you vary them, etc? In other words, do you do anything to be sure that while carrier X can place a call to carrier Y, you

Re: [asterisk-users] checking if a phone number is UP

From: "Carlos Alvarez" Sent: Thursday, February 09, 2012 11:25 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] checking if a phone number is UP A very interesting solution. Is there any code you'd share

Re: [asterisk-users] Garbled voicemail

Hi Dan, my wild speculation: It's some kind of timing/synchronisation problem. Do you use jitter buffer an/or echo cancelation? Best regards, Ruben -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Dan R

Re: [asterisk-users] checking if a phone number is UP

Reply to self, missed the line count part. Nevermind then :) Am 09.02.2012 18:10, schrieb Markus: But wouldn't that mean that every customer line is busy every 30 minutes for a few milliseconds for real callers? Unless there is more than 1 concurrent call enabled on the customers line. :) Am

Re: [asterisk-users] checking if a phone number is UP

But wouldn't that mean that every customer line is busy every 30 minutes for a few milliseconds for real callers? Unless there is more than 1 concurrent call enabled on the customers line. :) Am 09.02.2012 15:59, schrieb Bryant Zimmerman: We designed our solution the following way. We have

[asterisk-users] Garbled voicemail

Our Asterisk system (1.8.8.1-1digium1~squeeze) has been very stable and generally doing a good job -- except that one day, voicemail recordings started being garbled.

Re: [asterisk-users] checking if a phone number is UP

Brilliant! `log the call and busy out` is the thing I was missing. thank you so much On Thu, Feb 9, 2012 at 4:59 PM, Bryant Zimmerman wrote: > We designed our solution the following way. > > We have several land line numbers hooked to an asterisk testing server. > The testing server places on

Re: [asterisk-users] checking if a phone number is UP

A very interesting solution. Is there any code you'd share for this? We don't have inbound issues all that often (as far as we know), so I'm curious whether you had a lot of reliability issues before this, or possibly we have more problems than we believe. On Thu, Feb 9, 2012 at 7:59 AM, Bryant

[asterisk-users] Answering machine dectection (AMD)

Hi, I'll try to havebeen help for asterisk AMD module. Sorry for my bad english but I'll try to speak the better I'll able to do. So, here's my project: I did IVR. If you're pressing 1 key, asterisk's calling a mobile phone line. During the ringing, asterisk is lunching musiconhold for waitting

Re: [asterisk-users] TE410P (1st) without cables always green

Hello Shaun, 1) dahdi-linux-complete-2.2.0.2+2.2.0 , is generatiing 1K int/s after patches.. 2) Looking in 2.2.1 code, i see to wct4xxp.c the alarmdebounce set in 2500 and in 2.2.0 the alarmdebounce seted to 0, I load the module with alarmdebouce=0 paramter => the interrupts are up to 1000

Re: [asterisk-users] Asterisk V/s FreeSwitch

Am 09.02.12 16:45, schrieb Patrick Lists: > Iirc a long time ago there was a discussion about load testing by > playing MoH was not a realistic test. Something about all MoH music > getting streamed synchronized so basically Asterisk only has to stream > one file and sorta multiplex that single out

Re: [asterisk-users] Asterisk V/s FreeSwitch

From: "Patrick Lists" Sent: Thursday, February 09, 2012 10:42 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk V/s FreeSwitch On 09-02-12 14:52, Stefan Schmidt wrote: > Am 09.02.12 14:19, schrieb Bryant Zimmerman: >> Stefan

Re: [asterisk-users] Stuck DAHDI Lines

2012/2/9 Antonio Modesto > ** > Hi, > > Sometimes some of my dahdi channels become stuck, It is very strange, > here is the output of the "core show channels" command: > > pabx*CLI> core show channels > Channel Location State > Application(Data) > Local/104@ramais-cc0

Re: [asterisk-users] Asterisk V/s FreeSwitch

On 09-02-12 14:52, Stefan Schmidt wrote: Am 09.02.12 14:19, schrieb Bryant Zimmerman: Stefan This is on target with my configuration I am working on. What kind of dialplan were you using when running the tests. Were you doing database lookups or just answering the calls and playing hold music

[asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?

Hi everyone, I have tons of CDR from an Asterisk with a PRI connection. I want to know som extra details about the calls like the maximum number of calls in peak hours, etc...so I am looking for a php or other type of script that would show this to me in a GUI graphica format. Ideally, it would am

Re: [asterisk-users] Help with Codes and Polycom Phones

On 02/09/2012 04:08 AM, David Klaverstyn wrote: I can get Asterisk to work with G722 and the sound is superior compared to uLAW. I tried to get it working with Siren7 and Siren14 but I cannot. It always says incompatible codec and what is this Siren14 and Siren22 with Polycom. Is this different

Re: [asterisk-users] checking if a phone number is UP

We designed our solution the following way. We have several land line numbers hooked to an asterisk testing server. The testing server places one call every X seconds per line to a number we want to test . We cycle through each number in our testing pool. Each number on average is tested once e

Re: [asterisk-users] centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?

Hi, I'm also interested in rpm packages including chan_gtalk and res_jabber because I do not want to have a build environment on my productive server. Does anybody knows the reason why this is not available via rpm? best regards Christoph Am 28.11.2011 05:30, schrieb Vladimir Mikhelson: I j

Re: [asterisk-users] checking if a phone number is UP

I would probably set the "calling" server with a specific callerID, match on that caller ID on the recipient server and then log to a database or trigger a script to import my data. By doing that I can confirm the carrier side of things are up and that my server is actually processing calls. With

[asterisk-users] checking if a phone number is UP

hi, We have a phone number from third party provider which is used for inbound calls. How could I monitor if this *phone number* is reachable? the initial idea doesn't sound elegant: - on my SIP server I set couple seconds of ringing before Answer(). - the monitoring server calls to that phone nu

Re: [asterisk-users] Stuck DAHDI Lines

If you do a "core show channel conicse" and use the id from the LOCAL side of the call that is stuck that should shut down the channel without clearing all your DAHDI channels that are up. Does it only happen when you are calling throught a LOCAL context? Thanks Bryant --

[asterisk-users] Stuck DAHDI Lines

Hi, Sometimes some of my dahdi channels become stuck, It is very strange, here is the output of the "core show channels" command: pabx*CLI> core show channels Channel Location State Application(Data) Local/104@ramais-cc0 104@ramais:1 Up Tran

Re: [asterisk-users] Asterisk V/s FreeSwitch

Am 09.02.12 14:19, schrieb Bryant Zimmerman: > Stefan > > This is on target with my configuration I am working on. What kind of > dialplan were you using when running the tests. > Were you doing database lookups or just answering the calls and playing > hold music. Any example would be appre

Re: [asterisk-users] Asterisk V/s FreeSwitch

From: "Stefan Schmidt" Sent: Thursday, February 09, 2012 8:24 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk V/s FreeSwitch just done the test again. 13500 concurrent calls at 1750 cps with open rtp ports but without much

Re: [asterisk-users] Asterisk V/s FreeSwitch

just done the test again. 13500 concurrent calls at 1750 cps with open rtp ports but without much media transportet, only signaling. see attached screenshot. 1 concurrent calls with media playing musiconhold but i only have a 100mbit connection on this server so i cant do more here. the vers

Re: [asterisk-users] Asterisk V/s FreeSwitch

From: "Stefan Schmidt" Sent: Thursday, February 09, 2012 6:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk V/s FreeSwitch Am 07.02.12 12:38, schrieb virendra bhati: > Hi List, > > Why FreeSwitch can handle more then 1,0

Re: [asterisk-users] TE410P (1st) without cables always green

Hello Shaun, I take this approach : - zaptel 1.2.XX , 1.4.XX I can see interrupts working , I made some patches to compile in 2.6.32 kernel tree. - dahdi 2.0.0 to dahdi-2.2.0, I can see interrupts working, I made some patches to compile in 2.6.32 kernel tree - This "itop" from 2.2.0 tree

Re: [asterisk-users] Asterisk V/s FreeSwitch

Wow, I bet even asterisk developers wouldn't believe so. What have they done !. No, actually can you tell if server was processing media along with the calls as well !? I once tested without media and really I had some 1000+ CCs on asterisk server on a regular dev machine with choppy audio on an a

Re: [asterisk-users] Asterisk V/s FreeSwitch

Am 07.02.12 12:38, schrieb virendra bhati: > Hi List, > > Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What > technology FreeSwitch is used and asterisk don't. I don't know it's the > right or wrong but this question come to my mind... > I had done some load tests with ast

[asterisk-users] AGI with wrong ${AGISTATUS} Value

Hi everybody We're facing a strange problem with Asterisk (1.8.2.3) executing an AGI. The script (python) is updated and after a invocation by the dialplan, the new code is executed but the ${AGISTATUS} variable shows the wrong value. The right value only appears if a 'core reload' is executed

Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

On Thu, 9 Feb 2012 11:13:38 +, Steven Howes wrote: >Why not just use the latest version?.. Because converting Asterisk to run on that non-x86 platform is quite some work, so I need to know what I'm missing by staying with a 1.4.x release. -- _

Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

On 9 Feb 2012, at 11:08, Gilles wrote: > Does someone of a good site/blog that keeps track of new releases of > Asterisk, and explains what the major changes/features when they do > occur? Why not just use the latest version?.. S -- ___

Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

On Wed, 08 Feb 2012 20:23:54 -0600 (CST), Richard Mudgett wrote: >The CHANGES file is not just a dump. It is a manually created file that >documents each feature addition. There is a ChangeLog file that is a dump >of every single commit made to the source file. Sorry about that. Indeed, the CHA

Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

On Wed, 8 Feb 2012 18:17:46 -0800, Chad Wallace wrote: >Maybe the release announcements are what you're looking for. e.g., >for 1.8: > >http://www.asterisk.org/node/51444 > >And you can probably find the same for 1.4, 1.6.x, and 10 without too >much trouble. Thanks. It's closer to what I was loo

Re: [asterisk-users] Help with Codes and Polycom Phones

Only the higher end Polycoms support Siren7 and Siren14. I believe only the VVX and SoundStation IP phone support those codes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn Sent: Thursda

Re: [asterisk-users] Automatic Number Identification and anonymous calls

2012/2/8, Kevin P. Fleming : > On 02/08/2012 12:40 PM, Olivier wrote: >> 2012/2/8, Kevin P. Fleming: >>> On 02/08/2012 10:06 AM, Carlos Alvarez wrote: On Wed, Feb 8, 2012 at 2:35 AM, Olivier>>> > wrote: I always thought that ANI (Automatic Number

[asterisk-users] Help with Codes and Polycom Phones

Hi All, This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these: G722 Siren14.24kbps Siren22.32kbps Siren14.32kbps Siren22

Re: [asterisk-users] SIP hardware phones

2012/2/8, Carlos Alvarez : > If the customer is so cheap that they won't properly build out the network, > why would they have gigabit switches to the desktop which have a limited > set of applications that actually benefit from it? > > Then there's PoE, which is expensive to start and very expensi

Re: [asterisk-users] (last call for comments) Proposed changes to Asterisk release and support cycles

Le 09/02/2012 09:49, Administrator TOOTAI a écrit : Le 08/02/2012 23:28, Kevin P. Fleming a écrit : On 02/08/2012 04:02 PM, Danny Nicholas wrote: Not a complaint, per se, just a question. Why are the LTS versions "odd" (11, 13, 15, etc) and the non-LTS (10, 12, etc) even? As I read the char

Re: [asterisk-users] (last call for comments) Proposed changes to Asterisk release and support cycles

Le 08/02/2012 23:28, Kevin P. Fleming a écrit : On 02/08/2012 04:02 PM, Danny Nicholas wrote: Not a complaint, per se, just a question. Why are the LTS versions "odd" (11, 13, 15, etc) and the non-LTS (10, 12, etc) even? As I read the chart, Digium/Asterisk is committing to a new LTS versio