similar to: Sipura 1001 Adapter with two lines using one RG11 jack

Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM "Customer Relations Management" with an SIP dialers built in.the question is the following.. is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will

Hi All, I have a number of SPAX00X units (spa1001, 2002, etc) and about 30 odd PAP2-NA units all hooked up to Asterisk. As you can imagine, setting them up took a while, and changing settings on them also takes a while. In order to prepare for future deployments, I'd like to use XML provisioning (or any kind of remote provisioning). I figured since Sipura/Cisco won't release the utility

Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have been

I'm having lots of problems with sipura spa1001's and spa2002's. Asterisk claims they are busy when they aren't. Other times, it claims to be ringing them, but they aren't really ringing. I have done the following to try to resolve the problem: 1) I upgraded all my spa1001's and 2002's to their latest firmware (3.1.5). This lessened, but did not resolve

I have a phone (a panasonic globalrange phone) which always sends a fully qualified phone number. That is, for a local Canadian number, even if I key in 6135551212 it actually sends to asterisk 01116135551212. This means of course, along with "normal" phones I end up having twice as many extensions for outdialed numbers. Is there any way I could canonicalize this down to the more

Hi i have the running i both sides, client and server sides the puppet 0.25.4 Get this error on server side: puppetmasterd[5363]: Report puppet_dashboard failed: wrong Content-Length format And receive this error on my client side: warning: Value of ''preferred_serialization_format'' (pson) is invalid for report, user default (b64_zlib_yaml) I am getting any reports on my

On Thu, Feb 13, 2014 at 5:52 PM, Renato Golin <renato.golin at linaro.org> wrote: > On 13 February 2014 13:47, Evgeniy Stepanov <eugenis at google.com> wrote: >> Hm, I see that -funwind-tables on arm-linux-androideabi target >> replaces this "cantunwind" with a proper unwind table. >> Hence http://llvm-reviews.chandlerc.com/D2762. > > If Android is

I'm sure this might not be the correct place to ask and I have done a Google but I can't seem to find anything that says there is a VSP that will work with * in the Ukraine. I have a friend that lives in Kiev and basically want a phone number there to be able to talk to him and have him call me. If anyone has any information on it and they are willing to share please advise.

Hi all, What are the best and free tools for remotely adding, removing users on Asterisk server and also for monitoring the status of the Asterisk server, like how many users are logged on etc. I need tools for which I don't have to pay. Thanks, Zeeshan A Zakaria www.acabling.com <http://www.acabling.com/> -------------- next part -------------- An HTML attachment was

I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider -> IAX2 over the Internet -> 20Mb fiber connection -> router -> Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

Due to an issue I'm having with 7.x, and trying to track it down, I spent tonight getting my server setup to allow my to break into the debugger when it hangs, and hopefully dump core ... But, although I *think* I've got it all, I'm obviously missing something, as it isn't breaking ... First ... I'm running a proliant server, and when I connect via SSH to ILO on that

Hi, I am trying to do my first puppet configuration, already installed the puppetserver and client, in this link show my configuration and my puppet structure: http://paste.pocoo.org/show/212227/ But when i run the client side daemon i get this message: info: /Class[main]/Node[basenode]/Class[inittab]/File[inittab]/source: No specified sources exist err:

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

I am trying to build cross compiler for custom processor (say XYZ) but on compilation it is giving error llvm-build: error: invalid native target: XYZ (not in project) I have tried configuring like these 1. ./configure --target=XYZ 2. ./configure --target=XYZ --enable-targets=XYZ 3. ./configure --enable-targets=XYZ But every time it is not recognising the XYZ processor. What could be the

Hi all, I've just been told from an ex workmate that my VSP (who I used to work for) has put an anti flooding limit of 80 SIP messages per IP per minute in place. I run the phone system for a facility that has a lot of extensions, but would rarely have more than 4 or 5 simultaneous external calls. Am I in danger of tripping over this limit? It sounds dangerously low to me.

Hello ! I'm already using "fBasics" to generate alpha-stable variables or compute their density or distribution function but do you know where I could find .R tools for computing the correlation and fit a regression between two alpha-stable variables ? Thanks in advance ! Kind regards, Pascal Grosbuis (France) [[alternative HTML version deleted]]

Hi All, I seem to be having a problem with all my VSPs. When I am registering with them I don't seem to be passing my port number. This problem causes other users the inability to call my VoIP number with the VSP. My VSP showed me what they are seeing. I have changed my useragent to be: Linksys/SPA941-4.1.15 Linksys/SPA941-4.1.15 Contact sip:1234321234@aa.bb.cc.dd with no

All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,

Dear All, I like to simulate a physiologically based pharmacokinetics model using R but am having a problem with the daspk routine. The same problem has been implemented in Berkeley madonna and Winbugs so that I know that it is working. However, with daspk it is not, and the numbers are everywhere! Please see the following and let me know if I am missing something... Thanks a lot in advance,