search for: kkm

...sterisk to send a REFER back to the trunk, and essentially stay out of the loop. As set up, Asterisk initiates a new call using INVITE to the trunk and then bridges the original incoming and the new outgoing calls. Is that doable at all? What should be configured on the trunk peer? Thanks, -kkm

https://bugzilla.mindrot.org/show_bug.cgi?id=2246 Bug ID: 2246 Summary: PAM enhancements for OpenSSH server Product: Portable OpenSSH Version: 6.6p1 Hardware: Sparc OS: Solaris Status: NEW Severity: enhancement Priority: P5 Component: PAM support Assignee: unassigned-bugs at

...all and NAT. We are terminating multiple DID calls, originating and transferring. A provider offers both SIP and IAX trunking. Cateris paribus, what is the preferred solution to choose? What points to consider? I can name the provider if this is not against this list policy--is it? Thanks, -kkm

...: * Fix issue where MixMonitor() recordings would be shorter than total duration. (Closes issue #17078. Reported,tested by geoff2010. Patched by dhubbard) * When StopMonitor() is called, ensure it will not be restarted by a channel event. (Closes issue #16590. Reported, patched by kkm) * Allow hidecalleridname feature to work. (Closes issue #17143. Reported, patched by djensen99) * Resolve deadlocks in chan_local. (Closes issue #17185. Reported, tested by schmoozecom, GameGamer43) * Ensure channel state is not incorrectly set in the case of a very early answ...

...umentation from the TeX files by running 'make asterisk.txt'. (Closes issue #17220. Reported by lmadsen. Tested, patched by pabelanger) * When StopMonitor() is called, ensure that it will not be restarted by a channel event. (Closes issue #16590. Reported, patched by kkm) * Small error in the T.140 RTP port verbose log. (Closes issue #16998. Reported, patched by frawd. Tested by russell) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8 Thank you for your c...

...avkumar. Patched by navkumar, bencer. Tested by suretec) * Prevent loss of Caller ID information set on local channel after masquerade. (Closes issue #17138. Reported by kobaz, patched by jpeeler) * Fix SIP peers memory leak. (Closes issue #17774. Reported, patched by kkm) * Add Danish support to say.conf.sample (Closes issue #17836. Reported, patched by RoadKill) * Ensure SSRC is changed when media source is changed to resolve audio delay. (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler) * Only do magic pickup whe...

What unit is dtmftimeout measured in? The sample configuration is provided below. Does it mean to say that the sample configuration file's dtmftimeout=3000 equates 1/8000th of a second? ; The amount of time a DTMF digit with no 'end' marker should be ; allowed to continue (in 'samples', 1/8000 of a second) ; ;dtmftimeout=3000 -- Brian Camp IT Freedom direct

...: * Fix issue where MixMonitor() recordings would be shorter than total duration. (Closes issue #17078. Reported,tested by geoff2010. Patched by dhubbard) * When StopMonitor() is called, ensure it will not be restarted by a channel event. (Closes issue #16590. Reported, patched by kkm) * Allow hidecalleridname feature to work. (Closes issue #17143. Reported, patched by djensen99) * Resolve deadlocks in chan_local. (Closes issue #17185. Reported, tested by schmoozecom, GameGamer43) * Ensure channel state is not incorrectly set in the case of a very early answ...

...umentation from the TeX files by running 'make asterisk.txt'. (Closes issue #17220. Reported by lmadsen. Tested, patched by pabelanger) * When StopMonitor() is called, ensure that it will not be restarted by a channel event. (Closes issue #16590. Reported, patched by kkm) * Small error in the T.140 RTP port verbose log. (Closes issue #16998. Reported, patched by frawd. Tested by russell) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8 Thank you for your c...

...developers: * Make the mixmonitor thread process audio frames faster. (Closes issue #17078. Reported, tested by: geoff2010. Patched by dhubbard) * When StopMonitor is called, ensure that it will not be restarted by a channel event. (Closes issue #16590. Reported, patched by: kkm) * Fix up hidecallerid feature in chan_dahdi. (Closes issue #17143, #7321. Reported, patched by djenson99) * Resolve deadlocks in chan_local. (Closes issue #17185. Reported, tested by schmoozecom, GameGamer43) * Ensure channel state is not incorrectly set in the case of a very...

...avkumar. Patched by navkumar, bencer. Tested by suretec) * Prevent loss of Caller ID information set on local channel after masquerade. (Closes issue #17138. Reported by kobaz, patched by jpeeler) * Fix SIP peers memory leak. (Closes issue #17774. Reported, patched by kkm) * Add Danish support to say.conf.sample (Closes issue #17836. Reported, patched by RoadKill) * Ensure SSRC is changed when media source is changed to resolve audio delay. (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler) * Only do magic pickup whe...

Cannot find asterisk-mib.txt and digium-mib.txt anywhere. Were they dropped? -kkm

Hi All, We are running asterisk 1.8.3.2 and if in future i need to upgrade with new release then how should i upgrade it. The reason i am asking is i did lots of customization in make menuselect so do i need to keep all option remember or is there anyway i just copy paste existing configuration file in new tarball and just run make make install something like that... How you guys doing this

We are hiring a VoIP developer. Would it be within the list guidelines to post a position description? Not sure if that's in the scope of this list, but does not match asterisk-biz either. -kkm

Good day all! I have an issue which has plagued me for quite sometime now...and as I close in on its cause, I have reached a point where additional info would be greatly helpful! When a SIP device dials another SIP device...Asterisk connects the calls and displays the channel information. If one of those SIP devices hangs up, Asterisk receives the hangup notice and disconnects the call/channel.

Hello, I thought to post this here before my manager starts his own coding project to give us a workaround. My situation I'm running into is as follows: 1. A call comes into our Asterisk system, it's trunked from one office to another via IAX. 2. Call enters a queue and is picked up by one of the agents. 3. That agent has to transfer the call, could be for a number of reasons the client

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Today, i lost connection with the mail server. I think that there are some problem with the index file and it's log file. Following is the log for losing connection. Sep 10 11:18:01 biosoft dovecot: POP3(hjnam): Transaction log file /home/hjnam/mail/.imap/INBOX/dovecot.index.log: marked corrupted Sep 10 11:18:01 biosoft dovecot: POP3(hjnam): Fixed index file

Dear List, I'm writing to gauge interest in new syntax for positional-only function parameters to be added to R. The pattern of functions accepting other functions as inputs and passing additional ... arguments to them is prevalent throughout the R ecosystem. Currently, however, all such functions must one way or another tackle the problem of inadvertently passing arguments meant to go to