similar to: Seeking Beta testers for enterprise mystery service

You could use MTR command. Its a trace route improved. Marlon Araujo > On Jan 20, 2015, at 08:59, asterisk-users-request at lists.digium.com wrote: > > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or,

I have V2.0.0 about read to roll. There's quite a few new features, clean-ups and bug fixes. I hope this will be the last major update for awhile, and as such I'd like to squash any issues before releasing it. New- ** Dynamic day/date listboxes to prevent invalid date entries ** Textbox focus and tabbing order ** CDR-like view at past conferences to show participants ** Vastly

Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Since I passed from version 1.0 to the 1.2.3. I have Pb with the callerid. If somebody call with presentation of the number all is well. If somebody make call in masked number, i couldn't send a callerid to the phone. It is in a call center and i use the callerid to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the

To build in current unstable (and, I suspect, squeeze) xen-3 seems to need the following patch. (I'm not really sure what I'm doing and just cribbed from r14469 in the debian-kernel tree which is where the pertinent change to the linux-support package came from. It seems to work....) The errors from only editing rules.defs are attached. Index: debian/bin/gencontrol.py

typing his number, though there is a 15 seconds timeout, and even if I type the number very fast it still may happen to me. *same => n,Read(mobileNumber,app/input-mobile,10,,2,15)* In the logs: When it fails: - - <SIP/ipbx-iwred-000002e> Playing 'app/input-mobile.slin' (language 'fr') - - User disconnected When it succeeds: - - <SIP/ipbx-iwred-000002e> Playing

Hi, I am looking for a hosted / virtual IPBX *PLATFORM* for service provider.Such hosted IPBX platform is aimed to be as a service, so that final customers don't have to install, maintain, or upgrade any PBX hardware or software. It should have a control panel for end users to create / edit extension, conference rooms , IVR menus, etc Could you please indicate companies that offer such

Hi, I'm sending this mail here as I think it is of interest for at least europeans amongst you. The event presented below is not commercially-oriented, but really of interest for professional Asterisk users. in two weeks (9&10 october), the event "Asterisk and VoIP in enterprise" will take place in Brussels, Belgium. It is organised by Profoss ( http://www.profoss.eu ) to

I would like to ask, if someone observe also problem with peer qualify problems, my asterisk log is full with UNREACHABLE/REACHABLE messages, even when two asterisks are in LAN environment, please take a look into this debug, I can't find any problem with packet loss, all qualify requests are replied and acknowledged, I will submit bug report, if you will also not find any problems here...

Hi Is there a user client that a group, like customer service can use? We have today an avaya IP-office with phonemanager pro and I want something equal to phonemanager pro, where you can logon to ques and see how many calls is in that queue and so on. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir:

Has anyone had any luck getting the Avaya IP SoftPhone to work in wine? I get these errors on Ubuntu 8 Code: preloader: Warning: failed to reserve range 00000000-60000000 wine: creating configuration directory '/home/microchp/.wine'... preloader: Warning: failed to reserve range 00000000-60000000 preloader: Warning: failed to reserve range 00000000-60000000 err:dosmem:setup_dos_mem

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

Code-named R+Team, our commercially supported R group is looking for beta testers for R+ on Solaris and R+ on Linux. We'd love to get your feedback for our R+Professional version that has additional funcationalities for enterprise support. The sooner we get your feedback and inputs, the faster we'll make changes for the final release! To participate in our beta test program, please

Blogsaic.com is great new blog advertising site that uses images instead of a text description to advertise the purpose of the blog. http://www.blogsaic.com/ this site is in early development and needs beta testers to help develop the sites content. Content is user generated as opposed to using search bots, to ensure that only useful and entertaining blogs are listed. cheers for all and any

Note that it's "commercially supported", not just "commercial". Also, "commercial" does not mean "closed source". Open source (and even GPL) and "commercial" need not be mutually exclusive. Remember that the "free" in "free software" is as in "freedom" rather than as in beer (as far as GPL is concern, at least).

Hello, I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have done this before between Asterisk 1.6 and Avaya but had some issues placing external calls from the Asterisk to the Public network which is connected to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is outdated and has no support. On the Asterisk side I have Aastra 6731i SIP phones

Sort of a warning before I start: This post may be considered to describe a rather amateurish approach to distributing software which may annoy some people, but I sincerely hope it doesn't. I've been working for some years with David Clayton on a project which started life as an S package but has now turned into an R library. It is (now) called GLMMGibbs and estimates the parameters of

Hi! I've been on this list for over 3 years and this is my first post. We have a reporting application for Asterisk that is soon to be in beta. It's a windows application that generates reports from log files (CDR and queue). It has a drag and drop approach to report creating. There is a pivot grid and you just put the data that you want in a report, where you want it. We're

Dear UseRs, I have an update ready for the xlsReadWrite package. It runs my unit tests just fine but there are quite a lot of changes and it would be good if some other people could try it out before it will be submited to CRAN. I won't have more time this week, but I hope to publish xlsReadWrite at the end of next week and I will incorporate eventual bugreports and/or (minor) suggestions

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --