similar to: Integrating H.323 gateways with Asterisk?

If I have an object `obj` that is a SpecTask, and subsequently invoke it, is there a way to programmatically determine the number of tests that were successful, failed, and pending as a result of running that SpecTask? -- John Feminella Principal Consultant, Distilled Brilliance

Are there any generic install guidelines for compiling the Zaptel drivers on FC5? This is my first install of Asterisk (and my first FC5 system) and I'm having a great deal of trouble getting it to cooperate. make clean and make are definitely not playing nice, telling me that "You don't appear to have the kernel sources installed" when I'm pretty sure that I do. Any

Does anyone have any recent news on the Virbiage FT201 IAX Hardphone? I'd *really really* like to deploy these phones instead of SIP hardphones, and I can't help but wonder if I'm going to shoot myself in the foot (or another sensitive area) by deploying a ton of SIP phones just to find the IAX Hardphones were released a week later... Thanks, Brian D'Arcy

In our experience, the total cost of softphones(money, reduced sound quality and lower reliability) in a large call center environment is actually greater over time than the cost of a channelbank and cheap analog headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2 kinds of SIP analog adapters and we've tried channelbanks over the last 3 years. Right now we are half done

Hi, I've been asked sometimes to tailor call history features embeded in SIP hardphones. For example, a cutomer wanted internal call to be taken out. Another wanted calls to sorted according specific criteria. 1. Have you identified a phone offering the possibility to display as Call History, an XML list produced on a distant web server ? With this feature, you would simply have to tell the

I am running into a problem with allocating these files to a list as they are read in through a for loop. I know I am probably doing something wrong, but I can't figure out what. I know this is not reproducible. I am ending up with a data frame of the very last file to be read in. I know it is in the indexing, but I can't wrap my head around it. thanks for all of the help, Stephen

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to

I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with

06.10.2015 1:22, Joshua Colp ?????: > On 15-10-05 05:58 PM, Dmitriy Serov wrote: >> 05.10.2015 23:24, Joshua Colp ?????: >>> On 15-10-05 05:22 PM, Dmitriy Serov wrote: >>>> Hello. Do I understand correctly that the current implementation >>>> res_pjsip does not support ZRTP? >>>>

Hello Since SIP/RTP is a pain to use with road warriors who need to connect from any location over the Internet, I'd like to get them some IAX phones instead. For those of you using this protocol instead of SIP, what would you recommend as IAX hardphones and Windows (and ideally Mac) softphones? Thank you.

Hi, Is anyone aware of a wired sip hardphone supporting 802.1x authentication ? I've been told some Avaya and Alcatel ip phones supported 802.1x. As 802.1x is widely used with wireless hardphones, I'm wondering whether or not, 802.1x could also be valuable for wired environments. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Friends, We have implemented "Asterisk" in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost? Thank you. Regards, Chandra. --------------------------------- Ring'em or ping'em. Make PC-to-phone

hi all anyone that've heard of any working IAX hardphones yet? roy

Here is the code in question: describe UsersController do render_views … … describe "POST to ''create'' action" do describe "failure" do before(:each) do @attr = { :name => '''', :email => '''', :password => '''', :password_confirmation =>

What hardphones support xml/html/xhmtl/microbrowser? I need an inexpensive SIP hardphone that can run simple applications (queue status, etc). The phones I know of: Aastra 480i, 9112i, 9133i (though limited by 3 LCD lines on the 91xx seems kind of silly) Cisco 79xx Mitel 5235 Polycom IP601 Any others? -Dan

Hi! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

Hi, I've searched the archives for a discussion on the use of R for developing "small world" or "scale-free" network models but have been unable to dig anything up. Is anyone working on these types of models using R, and if so, have you found this language amenable to their development? thanks, matt farnsworth

I'm debating between a cloud PBX or, perhaps, rasberry pi. For a SOHO, maybe three hardphones, rasberry pi would suffice? I would be amazed, but, if so, great. thanks, Thufir -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160706/84a0ac7c/attachment.html>

Hi, Is anyone aware of SIP Diversion header ? It seems currently supported by Comverse (formely NetCentrex) softswitch and some hardphones (Thomson ST2030). An old draft (draft-levy-sip-diversion-08.txt) mentions this header. ha I'm wondering if this could be used -------------- next part -------------- An HTML attachment was scrubbed... URL:

Guys, this is puzzling. Seems I cant use any of the feautes (call transfer, record call, etc) defined in features.conf when a call comes in thru zap and I answer it on hardphones... Although I CAN use them when Im the one that originates the call, when received I just cant. My dialplan includes wtWT on all Dial cmds just to be sure but it doesn't seem to be working. Any pointers?