similar to: Asterisk bounty PRI 2B channel transfer for NI2 PRI line

Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5

Hello, I'm french, so excuse my poor English. I'm face to a terrible thing, with has stole a lot of my time. On the .184 machine, I've the following iax.conf : [general] rtcachefriends=yes bandwidth=high tos=reliability jitterbuffer=no autokill=yes #include "iax.voip1.conf" #include "iax.renoir.conf" The iax.voip1.conf file contains : [VOIP1] type=friend

I am terminating a PRI, NI2 signaling into a PBX (My company's PBX not the PSTN) from an Asterisk server. Caller ID number appears to be transmitted caller id name is not being transmitted is their a compile time flag on libpri that I need to un-comment to enable this feature or is this is unsupported for now? Thanks Chris. -------------- next part -------------- An HTML attachment

I did some searching to find out what I need to do to get CID&NAME inbound on one of our PRI circuits. Right now, we're only getting number (well, asterisk shows the number in both the name and number portions of CID). Is there anything special I need to do with Asterisk to have it accept the name portion of CID along with the number on a PRI that is provisioned NI2? Thanks, John

Okay, now I am really confused. I have two PRI's coming in from two different Carriers (QWEST and ELI), both of them are supposed to be setup to pass name and number on incoming calls. Problem that I am having is that I am not receiving inbound caller id name on either PRI, the only thing that both carriers have in common is that I am terminating into a DMS switch at the carrier.

We need an expert in least cost routing (LCR) for an Asterisk project. Please provide references and a resume of your experience. Contact us at voip3@nibble.net.

I've got the VICI predictive dialer runnning over IAXs to another asterisk server. It dials fine. I can make phone calls manually with no problem. When VICI dials a new number it rings the other end once and I get the error: Jan 26 13:53:10 NOTICE[10206]: Dropping incompatible voice frame on IAX2/VOIP3/5 of format slin since our native fo rmat has changed to gsm I have set ALLOW=ALL in

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

Couple of patches to fix an overflow, optimize a bit and support bigger resolutions using Windows 8.

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 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

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:

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

Hello all, Just a quick note to let you all know that I have placed a bounty on Lauchpad to get bitrate peeling added to Vorbis. It is a feature that I think we would all like to have, and would probably pay something to get, but it hasn't been done. My request to you is to add to the bounty. I have seeded it with US$20, which is not enough to motivate a developer to get it done, but I am a

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 >

http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I

http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20Meet%20 Me%20video%20conferencing I posted this bounty for $US2,000 some months ago. Basically I needed the ability for 4 or 5 of us to conference on a weekly basis which is why I was happy to offer this bounty, however I have only had 2 people make brief inquiries and no one has really offered any substantial indication they

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

(forgive the brief interruption to -users with a mostly -dev issue, just wanted to publicize this on behalf of the larger community) If there are any ambitious coders out there (not too many shekels yet, but I expect some folks may pony-up) please see: www.voip-info.org/?page=Asterisk+Bounty+Email+TTS We are at $150 & counting. Maybe lobby your exec's for $50 to contribute to this,

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