Re: [asterisk-dev] MIXMONITOR - periodic beep is not present in RECORDING

Thanks Jonathan, Will try this . On Wed, Jan 12, 2022 at 1:46 PM Jonathan Rose < jonathan.r...@motorolasolutions.com> wrote: > > On Wed, Jan 12, 2022 at 12:37 PM bala murugan > wrote: > >> Thanks jonathan >> >> I will flip them and give it a try >>

Re: [asterisk-dev] MIXMONITOR - periodic beep is not present in RECORDING

Thanks jonathan I will flip them and give it a try Can you also share the patch for 1.6.2 ? thanks, Bala On Wed, Jan 12, 2022 at 1:12 PM Jonathan Rose < jonathan.r...@motorolasolutions.com> wrote: > > > On Wed, Jan 12, 2022 at 12:08 PM bala murugan > wrote: > >> T

Re: [asterisk-dev] MIXMONITOR - periodic beep is not present in RECORDING

or the changes you mentioned? On Wed, Jan 12, 2022 at 12:36 PM Jonathan Rose < jonathan.r...@motorolasolutions.com> wrote: > On Wed, Jan 12, 2022 at 10:41 AM bala murugan > wrote: > >> Hi , >> >>I am trying to use the MIXMONITOR Option B( interval ) - Play a

Re: [asterisk-dev] MIXMONITOR - periodic beep is not present in RECORDING

t with a dial option L > > On Wed, Jan 12, 2022 at 6:51 PM Joshua C. Colp wrote: > >> On Wed, Jan 12, 2022 at 12:41 PM bala murugan >> wrote: >> >>> Hi , >>> >>>I am trying to use the MIXMONITOR Option B( interval ) - Play a >>> per

[asterisk-dev] MIXMONITOR - periodic beep is not present in RECORDING

Hi , I am trying to use the MIXMONITOR Option B( interval ) - Play a periodic beep while this call is being recorded . I can see the Beep tone is played , but the same Beep is not present in the RECORDING . Any ideas ? Any Suggestions? Is this a known issue? Thanks, Bala --

Re: [asterisk-dev] [BOUNTY] X-Header on SIP BYE

Hi Ernst , I can develop this for you in chan_sip driver on the version you are using 15.7.2 . If you could provide more details like - How are you planning to pass the value for the X-Header ? - Is this needed for all BYE initiated by asterisk ? - Is this required for Bridged

Re: [asterisk-dev] Unable to compile asterisk with MALLOC_DEBUG

i tried with asterisk 13 and still the same behaviour On Sun, May 26, 2019 at 2:31 PM bala murugan wrote: > Understood , I hope this basic should still work on 12 . > > Upgrade is not an Option for me just because of this . > > On Sun, May 26, 2019 at 3:09 AM Jared Smith >

Re: [asterisk-dev] Unable to compile asterisk12 with MALLOC_DEBUG

right? You should try with > the latest 13.x or 16.x release. > > -Jared > > On Sat, May 25, 2019 at 3:34 PM bala murugan > wrote: > >> Hi , >> >> I am trying to compile asterisk12 with MALLOC_DEBUG , it always fails >> with below >> >> ael.fl

[asterisk-dev] Unable to compile asterisk12 with MALLOC_DEBUG

Hi , I am trying to compile asterisk12 with MALLOC_DEBUG , it always fails with below ael.flex: In function 'ael_yyfree': ael.flex:667: error: lvalue required as left operand of assignment ast_expr2.fl: In function 'ast_yyfree': ast_expr2.fl:255: error: lvalue required as left operand of

Re: [asterisk-dev] ASTERISK - DTMF Inband Handling

Thanks Matt , see response inline On Fri, Jul 14, 2017 at 4:50 PM, Matt Fredrickson <cres...@digium.com> wrote: > On Wed, Jul 12, 2017 at 10:47 AM, bala murugan <fightwit...@gmail.com> > wrote: > > Hi , > > > > can anyone provide answers or comm

[asterisk-dev] ASTERISK - DTMF Inband Handling

Hi , can anyone provide answers or comments on the below 1. What is the expected minimum and maximum silence duration between the dtmf tones. 2. mindtmfduration = 80msec in /etc/asterisk/asterisk.conf what is the significance of this parameter ? . Is there any maxdtmfduration or

Re: [asterisk-dev] Load on SIP MESSAGE how many per sec asterisk can Handle

nd > confirmation by 200 OK or 202 before sending next MESSAGE. > > If you send to many URI:s, then expected throughput might be higher. > > Regards > > Gunnar > > Den 2017-06-12 kl. 18:23, skrev bala murugan: > > Thanks Gunnar , > > This is load test to un

Re: [asterisk-dev] Load on SIP MESSAGE how many per sec asterisk can Handle

On Mon, Jun 12, 2017 at 12:23 PM, bala murugan <fightwit...@gmail.com> wrote: > Thanks Gunnar , > > This is load test to understand how many message it can handle and where > the bottleneck is . > > 14 MESSAGE / sec - takes longer time for the Message to be processed , not

Re: [asterisk-dev] Load on SIP MESSAGE how many per sec asterisk can Handle

taskprocessor works or implemented . thanks, bala On Wed, May 24, 2017 at 3:18 AM, Gunnar Hellström < gunnar.hellst...@omnitor.se> wrote: > Den 2017-05-23 kl. 23:58, skrev bala murugan: > > Hi , > > Is anyone aware of how many SIP MESSAGE per sec asterisk can handle , i

Re: [asterisk-dev] Issue with Parsing Contact Header without Brackets and with additional HeaderParameters seperated with semicolon

Here you go george , i have raised a issue in Jira see below 1. Asterisk 2. ASTERISK-27045 Issue with Parsing Contact Header without Brackets and with additional

Re: [asterisk-dev] Issue with Parsing Contact Header without Brackets and with additional HeaderParameters seperated with semicolon

find it mentioned required always please advise . btw i am using chan_sip On Wed, Jun 7, 2017 at 5:18 PM, George Joseph <gjos...@digium.com> wrote: > On Tue, Jun 6, 2017 at 9:44 AM, bala murugan <fightwit...@gmail.com> > wrote: > > Hi , > > > > Can anyone t

Re: [asterisk-dev] Issue with Parsing Contact Header without Brackets and with additional HeaderParameters seperated with semicolon

Thanks George , I am using chan_sip On Jun 7, 2017 5:19 PM, "George Joseph" <gjos...@digium.com> wrote: > On Tue, Jun 6, 2017 at 9:44 AM, bala murugan <fightwit...@gmail.com> > wrote: > > Hi , > > > > Can anyone tell me if there is a know bug

[asterisk-dev] Issue with Parsing Contact Header without Brackets and with additional HeaderParameters seperated with semicolon

Hi , Can anyone tell me if there is a know bug raised and fixed when we handle or Parse Contact Header , if it is presented without brackets I get a INVITE with Contact:sip:p65549t000m112562c59100@10.196.0.111:5089 ;+g.3gpp.accesstype="cellular";+sip.instance="" currently this is

Re: [asterisk-dev] BOUNTY - ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code()

Hi Ross , I can work on providing a fix for this , if you could let me know how to recreate this problem and also the asterisk version you are experiencing this issue . thanks, Bala On Wed, May 24, 2017 at 3:20 PM, Ross Beer wrote: > HI All, > > > I would like to

[asterisk-dev] Load on SIP MESSAGE how many per sec asterisk can Handle

Hi , Is anyone aware of how many SIP MESSAGE per sec asterisk can handle , is there a benchmark has this been load tested and results available some where , if there is can you some one share it please . The reason is we ran 16 per sec and we see the ast_msg_queue is backing up with lot of

Re: [asterisk-dev] Asterisk Memory Debugger (MALLOC_DEBUG) and DONT_OPTIMIZE

there are multiple CLI commands - with no detailed info on what it is and how to use to spot where the leak is Is there any doc as such with example . thanks, Bala On Fri, May 5, 2017 at 2:32 PM, Richard Mudgett <rmudg...@digium.com> wrote: > > > On Fri, May 5, 2017 at 11:55 AM, bala mu

[asterisk-dev] Asterisk Memory Debugger (MALLOC_DEBUG) and DONT_OPTIMIZE

Hi , Can someone please help me understand what to look for in the /var/log/asterisk/mmlog to check where the leak is , since on Exit it throws me millions of line under Exiting with the following memory not freed. need some knowledge on reading this file so that i can pinpoint where the leak is

Re: [asterisk-dev] [BOUNTY] 26414 - ExternalIVR attended transfer - no audio

Hi Chris , i can work on this and see if i can provide a fix for this . i have updated the Jira requesting for the PCAP . please provide . thanks, On Wed, Jan 18, 2017 at 4:53 AM, Chris Maciejewski wrote: > Hello, > > I would like to offer $800 bounty to fix bug below: >

Re: [asterisk-dev] ARI Bridge Behavior

Any idea how you disabled the channel_varset from stasis . can you provide the config or steps ? thanks, On Tue, Jan 10, 2017 at 8:47 AM, Michael Petruzzello < michael.petruzze...@civi.com> wrote: > The following occurs when channels are in a mixing bridge with dtmf_events > enabled created

Re: [asterisk-dev] Asterisk Load Performance

How to disabled the channel_varset from stasis in asterisk 12 , can you please provide the steps or configuration . thanks, bala On Fri, Jun 17, 2016 at 5:31 PM, Matthew Jordan wrote: > On Fri, Jun 17, 2016 at 1:37 PM, Richard Mudgett > wrote: > > > >

[asterisk-dev] G722 and Inband DTMF Detection Support

Hi , I tried to use inband using G722 , but asterisk complained we cant use inband and use rfc2833 on G722 , So wondering why there is no support for inband on g722 as some of the Telecom Switches are sending Inband DTMF when using G722 Codec . Is it hard to support this in asterisk and Will

Re: [asterisk-dev] G722 and Inband DTMF Detection Support

Any Input ? On Wed, Sep 23, 2015 at 10:45 AM, bala murugan <fightwit...@gmail.com> wrote: > Hi , > > I tried to use inband using G722 , but asterisk complained we cant use > inband and use rfc2833 on G722 , So wondering why there is no support for > inband on g722 as some of

Re: [asterisk-dev] Reject incoming call

Who is responsible to answer ?? On Feb 12, 2015 6:05 AM, Raj Roy Ghandhi roy.gan...@gmail.com wrote: Hi Friends, I am trying to implement a simple dial plan with asterisk. 1. Ring the inbound call 2. wait for 2 seconds 3. call agi script with cli 4. hangup But when it gets hangup I see

[asterisk-dev] Asterisk created mmlog

Hi , I am trying to understand the content written to mmlog file looks like it is difficult , with this how I can find where is memory leak happening is there any document on this what to look for and all the cli commands and how to use to find where is memory leak . Thanka --

[asterisk-dev] Confbridge Performance using G722 @ max 3 Party in 1 conference.

hi , Has anyone tried measuring the asterisk (10 or 11 or12) performance against the Confbrdige Application using G722 HD codec with atleast 3 party in 1 conference . If yes can you please provide me what tool was used to measure this performance , it is hard to get a tool that supports g722

[asterisk-dev] Regarding CLI Command Result

Hi , I installed asterisk 12 and when I run command connecting to CLI core show taskprocessors I see below results , not sure the processor name with Hex char what it means has anyone got the same result and if possible can explain how to interpret this results . core show taskprocessors

Re: [asterisk-dev] Regarding CLI Command Result

thank you mark On Tue, Aug 5, 2014 at 11:56 AM, Mark Michelson mmichel...@digium.com wrote: On 08/05/2014 09:56 AM, bala murugan wrote: Hi , I installed asterisk 12 and when I run command connecting to CLI core show taskprocessors I see below results , not sure the processor name

[asterisk-dev] Peer callingpres - not used when we make SIP Outdial

Hi , I tried to set the parameter value callingpres and tried making outdial to the same peer , but the value we set to this parameter is never getting used when make an outdial and looked at the code and it is never getting used in the outdial portion , not sure if this is a BUG . I tried

Re: [asterisk-dev] Asterisk Leaks FileDescriptor in handle_recordfile - if Call Disconnect happens while playing beep

, Matthew Jordan mjor...@digium.com wrote: On Wed, May 21, 2014 at 7:33 PM, bala murugan fightwit...@gmail.com wrote: On Wed, May 21, 2014 at 8:23 PM, bala murugan fightwit...@gmail.com wrote: HI , has any one noticed in res_agi.c handle_recordfile , if there is a call disconnect

[asterisk-dev] Asterisk Leaks FileDescriptor in handle_recordfile - if Call Disconnect happens while playing beep

HI , has any one noticed in res_agi.c handle_recordfile , if there is a call disconnect while beep is played back to caller the beep filestream is not getting closed and it leads to FileDescriptor Leak. This is still there in asterisk 12 . I have a Fix and Let me know I can submit the same .

Re: [asterisk-dev] Asterisk Leaks FileDescriptor in handle_recordfile - if Call Disconnect happens while playing beep

On Wed, May 21, 2014 at 8:23 PM, bala murugan fightwit...@gmail.com wrote: HI , has any one noticed in res_agi.c handle_recordfile , if there is a call disconnect while beep is played back to caller the beep filestream is not getting closed and it leads to FileDescriptor Leak

Re: [asterisk-dev] [BOUNTY] ASTERISK-19454: outbound proxy not being cleared

The CR says this is Fixed On Tue, Dec 10, 2013 at 2:57 PM, Henry Fernandes he...@usinternet.comwrote: We'd like to have bug ASTERISK-19454 ( https://issues.asterisk.org/jira/browse/ASTERISK-19454) fixed and are offering a $500 bounty for this. Please email me if you are interested. -H

[asterisk-dev] Asterisk 12 production release ...

Hi , I see asterisk 12 is in beta release , any idea when is the production release officially expected ... thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To