similar to: Re: Square key KTS app on *

Yes - the problem was a missing signalling line in zapata.conf. Now in and out work. Also, it was news that "reload" from the console doesn't reflect changes made in zaptel and zapata.conf entries. Any other config files that it doesn't reload? >From: "Asterisk@Home" <asteriskathome@yahoo.com> > >Did you go into AMP and configure some place for

Is there a configuration difference for clone X100P cards versus "compatible"? I have a similar problem to what David Shaw posted earlier today. 0.5 installed OK, but mine just with one X100P clone. Default config files, edited zapata.conf per the FAQs so it includes the line channel => 1 without the semicolon. Any outgoing call attempt returns "all circuits are busy"

We are using Bellsouth 8867 phones on our TDM400B FXS lines (asterisk-1.0.3). It has a "Voicemail" light, which appears to be MWI (according to the manual it works with voicemail from the telco that sends a FSK signal). The dialtone stutters when a line has voicemail, so I know that I have the mailbox setting right in zapata.conf, but the light doesn't go on. I am also getting

Hi, I've been having a little problem with my asterisk servers, I have 4 identical asterisk servers setup (same hardware, same OS, same config). Once in a while (once or twice a day) one of the server crashes on the cron job reload. But I realized this only happens on 3 of the 4 servers. Tried to spot the difference between that one server that wasn't crashing. The difference I found was

Being bored to death by these long weekends with nothing to do? **** Why not go bounty-hunting? **** There are some feature requests in the bug tracker with monetary bounties attached. * Windows manager * FreeBSD Zaptel drivers http://bugs.digium.com/bug_view_page.php?bug_id=0000847 * IAX incoming/outgoing limit * 2B channel transfer on PRI * MGCP media gateway support All of these have

I have the following helper method in application_helper.rb: def format_date(date) day = to_s(date.day) month = to_s(date.month) time = to_s(date.time) date = day + "/" + month + " - " + time return date end I am trying to call this method in a view like this: <%= format_date(bounty.created_on) %> create_on is a timestamp in mysql. I am

I bought an Aastra PT480 from digium, but I wanted to see if I could get some more help with this before Monday. Any help would be appreciated. I have the phone connected to the TDM400P card, and I also have the T100P and the X100P in the same box. My problem is, it appears as if the phone and asterisk can't understand each other. The port the phone is connected to always remains

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

I am the one that placed the bounty. After it being there for 2 months and getting no takers (and very few if any people asking about it), we are almost finished writing it in house. I'll keep the bounty up untill we do finish our product so if anyone beats us to getting it working they'll get paid... W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com >

I saw some of the heat about the $20 bounty earlier. So I don't want to put a low bounty out. Quote me a bounty, and I'll see if I can get it approved by management. :-) I'm in need of getting this bug fixed. Bug has all of the details, but basically 1.4.22 broke it all. I've waited as long as I can - hoping the bug would 'resolve itself' - but now I'm putting a

The bounty stands at $5,500. I'm seriously considering taking a shot at it if I can find a decent T.38 provider to test with (I'm still hoping for reliable PAYG T.38). It looks like a lot of very smart people have done a lot of very hard work (t38modem, spandsp) that would go towards getting this working. At this point it appears to be mostly a matter of integration

bounty offer prolonged to 31.4.2015 (end of april) Dne 3.3.2015 v 16:22 Marek Cervenka napsal(a): > hi, > > i'm offering bounty[1] $500 (five hundred) US dollars for resolving > https://issues.asterisk.org/jira/browse/ASTERISK-22708 > > fix must be available for asterisk 11.x and asterisk 13.x and accepted > to upstream > As part of this fix we should see seamless

Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

I sent this yesterday from my wrong email account. yeah, Thank you. I screwed up by only looking at adsi.conf, and not asterisk.adsi. Once I saw that, things started falling into place. Are you able to pass arguments to the phone so that for every call there is unique data on the screen. I see that works for caller id with $Call1p and $Call1s, but what If I obtain an OrderID from my

Hi, I've just initiated a new bounty for the above; http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL Any takers or any contributors please respond to me privately. I do not know exactly how the bounty process works, but I can coordinate on this ? SW -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sirs, I can't compile the source spandsp-0.0.2pre10; when i try to do the "make" sentence the following errors appear: # make Making all in src make[1]: Entering directory `/export/usr/src/spandsp-0.0.2/src' make all-am make[2]: Entering directory `/export/usr/src/spandsp-0.0.2/src' if /bin/sh ../libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT

Has anyone got Polycom Shared Call Appearance working with Asterisk ? If Asterisk doesn't support this, I am willing to put up a bounty of 1000 to get it to work. John Bittner Simlab.net Shared Call Appearance Signaling A shared line is an address of record managed by a server. The server allows multiple endpoints to register locations against the address of record. SoundPointR IP

Folks, I am in the process of fighting my way through my first foray into ADSI phones. I have a very nice Aastra 480 from Sayson, purchased through Cylogistics. Very nice phone. However, my experience in configuring ADSI has been pure pain so far, and it's not all working yet... 1) The locking mechanism for these ADSI phones is... baroque? Arcane? Perhaps there is a reason why you would

Hi Nicolas, There doesn't seem to be any interest in using asterisk and video. I posted a $1,000 bounty to get video meet me working without a single reply. I have now just bumped this to $2000 http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+vid eo+conferencing This is a legitimate commercially binding bounty, I hope this might inspire some people to develop at least