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Hi i am using pap2 phone adaptors as clients to connect to asterisk server i am able to make calls but i cannot access voice mail using phone or start recording while call is in progress and when i place a call to local sip extension there is a long pause ( 15 sec ) before the call gets dialled i assume that the problem would be due to the dial plan in PAP2 if so please help me changing it

HI I have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ? i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

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Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help,

Hi all. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten => 123,2,Dial(SIP/3000,,wW) ; wW allow one-touch recording During the call, I press *1 but it records nothing. David Morrow

On 2023-07-20 04:36, Itamar Reis Peixoto wrote: > > my predict is that they will continue as a #rebuilder / #freeloader, > writing software is a hard work. > #offensive terms to the community :-), hide hat wrote it. No, they didn't. That term was bandied about on social media by people who were speculating about the reasoning behind discontinuing the practice of debranding and publishing packages from RHEL minor releases. Mike McGrath responded to the use of that term by social media personalities to explain that the only group that Red Hat (fo...

I''m wanting to write a spec that a model is applying an :order option to a find call, but I don''t want to completely specify all of the find parameters. So I want to write something like this, say in a controller spec User.should_receive(:find).with(:all, hash_with_at_least(:order => ''user.name ASC'')) get ''index'', :sort =>

...07-20 04:36, Itamar Reis Peixoto wrote: > > > > my predict is that they will continue as a #rebuilder / #freeloader, > > writing software is a hard work. > > #offensive terms to the community :-), hide hat wrote it. > > > No, they didn't. > > That term was bandied about on social media by people who were > speculating about the reasoning behind discontinuing the practice of > debranding and publishing packages from RHEL minor releases. > > Mike McGrath responded to the use of that term by social media > personalities to explain that the only g...

<table cellspacing='0' cellpadding='0' border='0' ><tr><td style="font: inherit;"><font size="2">Dear All,</font><br> &nbsp;<br> <font size="2">&nbsp;&nbsp;&nbsp;&nbsp; This is my first post on the wine organisation mailing lists, i would like to inform all of you i am using

Hi Guys, Has somebody encountered (significantly) different read/write speeds when using gmirror? I have 2xWD WD30EFRX RED drives which are configured as follows: $ gmirror status Name Status Components mirror/root COMPLETE ada0p2 (ACTIVE) ada1p2 (ACTIVE) mirror/data COMPLETE ada0p4 (ACTIVE) ada1p4 (ACTIVE) mirror/root is mounted

> I can't predict the future but my feeling is that AlmaLinux has a good > chance to become the second Gold standard. I disagree with you. Both Rocky Linux and AlmaLinux are in a very comfortable position, and they will likely stay that way. my predict is that they will continue as a #rebuilder / #freeloader, writing software is a hard work. SuSe hardfork will probably be only an

Hi I am trying to install Dovecot on HP-UX .I did ./configure but when I did make i am getting following error, please advice me what I have to do . uasng074 at root# make No suffix list. make all-recursive No suffix list. Making all in src No suffix list. Making all in lib source='ioloop-epoll.c' object='ioloop-epoll.o' libtool=no \ DEPDIR=.deps

Hi (and Merry Christmas), I am trying to compile R on an AIX IBM machine using the native C and FORTRAN compilers... and everything goes smoothly (congratulations for the configure and make files, this is very nice)... until the pacakge 'cluster' (see dump below).... Any hint ? Laurent 1501-510 Compilation successful for file pam.f. cc -I/data1/laurent/R-1.6.1/include

Hi, I'm speech recognition researcher and would like to do some research on recognition robustness in echo distortion of speech signal. Since VOIP is becoming wide spread, I'd like to simulate (one or more) common echo distortions that mostly appear in voip communications ? Any example, FIR or IIR filter or acoustical system response ? Any other distortion worth researching ? Thanks

Hi i have installed AAH 2.6 and configured some extensions the calls are working fine. but if i dont answer a call then it says " the person at extension " and hangs up . it doesnt spell out the extesion number nor it goes to voice mail box. *************************** Asterisk CLI log **************************** dialparties.agi: Extension 200 is available...skipping checks --

Hi i am looking for a good ivr system for my company. these are my question are there any good ivr's that can be easily integrated with asterisk ? and are there any large scale deployment of asterisk to date ? thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Got 1 issue I can't seem to knock out of this particular box. The IVR works fine on the zap channels and the incoming SIP routes. But coming in via the IAX2 route leaves me with a silent phone. The prompts all work still letting me navigate the menu. But just can't hear anything. This is with A@H 2.8 (Asterisk 1.2.7.1, with FreePBX 2.1.0 also installed) Any thoughts on where to

Hi List, Is it possible to define how 'in' works for an object of a specific class (to achieve a similar result to implementing the iterator protocol in a class Python)? Cheers, Simon Knapp [[alternative HTML version deleted]]

I saw in one of Dave C.''s comments to a ticket that "our current plan is to deprecate the mocking framework." I hadn''t heard anything about that, but then again I haven''t paid super close attention to the list. Are we planning on dumping the mock framework in favor of using Mocha (or any other framework one might want to plug in?). Pat