similar to: Tier 1 Service Providers (AT&T, Level 3)

Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording

Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can

Hello Everyone, I have disallow=all and allow=g729 set in sip.conf however, it seems that asterisk still thinks it support other codecs: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw). How can I disable gsm,ulaw,alaw..... Thanks in Advance, Nick.

Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g729, and gsm would also be good....

Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not sure if we are using the sipp correctly etc.. but nevertheless, is there any documentation that describes how we can get the most our of our Asterisk box. For example when we hit the "too many file" error, and fixing it using ulimit..... Also, is there any way we can allocate sufficient

Hello Everyone, I have bumped into the thralling penguin page on linux vs solaris for asterisk. Does the benchmark still hold with the newer versions of kernels? Curious to know of your thoughts. Also, they mentioned running it on Sun Fire x2100, but no benchmarks were given for that. Can increased performance be accomplished simply by changing to Solaris or OpenSolaris? Kind Regards, Nick.

Anyone try this? I saw a post here: http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/94041-setup-of-sangoma-a101-in-my-elastix.html But not sure if it's possible. What I am asking is if there are any T1 cards with virtual functions implemented in their drivers to allow pci-passthrough? Kind Regards, Nick.

Hello Osama, and Hisham, At 1330GMT there was some malicious activity coming from your network IP 37.75.210.90. Please act accordingly. Things that may be of use "972599779558" N.

Hello Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an OpenSIPS/Asterisk integration. Most approaches have OpenSIPS relay the UA's REGISTER request to Asterisk which has "host=dynamic" set for the

Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent "issues" that

Hello Everyone, We are looking to interconnect with a local ILEC over an OC-n transport layer. They basically gave us two options in terms of mapping the SONET to the DS3: * VT1.5s mapping * DS1s mapping The second option is quite clear. We would MUX the connection, and plug the lines into qaud t1 cads etc... The tech mentioned that with the second option we would also need a DACS to convert

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

Hi, Inspired by a recent rant about one particular provider, I am getting very curious about something I've never mastered. I'd like someone to explain this here or at least post a link or two that can educate me and probably countless others who have no knowledge in this area. I'm sure there are several of you reading this that know all about the subject. What are the various

Hello, I'm planning to start using R. Before getting into it, I'd like to ask a couple of questions. Does R carry out loglinear model analysis? That is, will it provide the chi-squared goodness of fit test statistic for a given hierarchical loglinear model? Maybe even do a model selection procedure (like Brown's two-step procedure, or forward/backward selection)? Thanks

Hello Everyone, I am required to write a java program that will get our asterisk to: * Query the database for phone numbers * Loop through numbers and dial * Play message * Get dial pressed response - If 1 = Yes - If 2 = No - If 3 = Connect to Agent * AMD Capable * Disposition I am proficient with Java and found the Asterisk-Java API. My questions are: * What is the

Hello Everyone, The timezone is set correctly on the OS America/Toronto: mv /etc/localtime /etc/localtime.bak cp /usr/share/zoneinfo/America/Toronto /etc/localtime I even tried adding the timezone setting to sip.conf: timezone=America/Toronto However. Asterisk wants to be in Bucharest? Thinking about it, I want to be in Bucharest! Cheers, Nick.

Hello There, I would like to play a busy tone (ie BEEP BEEP BEEP) when the underline carrier sends back 486 Busy Here. Looking at Dial parameters ( http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial), it mentioned something about the r parameter as not being very professional or something like that... Then there was: U(x): Executes, via gosub, routine x on the called channel. This is similar

Hello Everyone, We are looking for a simple open source auto dialer with "polling" capabilities. What we would like is a program that we can upload leads to, and have asterisk: i) Dial numbers ii) Play pre-recorded iii) If user presses one, forward the call to an agent There are so many solutions out there it's hard to make a decision on what works, what has just a limited free

I have a Polycom Soundpoint IP335. There are no inbound routes set to the phones yet. However, the phones are getting phantom rings. What is the legitimacy of these calls? Is there something I need to block to stop it? I believe its people trying to hack the phones/phone system but I cannot find where I read that before. Thanks, --E -------------- next part

I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In