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...rry 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 for your help. ---Harry Khamis -- Harry Khamis Statistical Consulting Center Wright State University Dayton, OH 45435 USA Phone: (937) 775-2433 Fax: (937) 775-2081 Homepage: www.math.wright.edu/People/Harry_Khamis/index.html

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, 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 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, 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.

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

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

...your opensips is doing Record-Routing, but the > BYE does not contain the corresponding Route hdr, so SIP routing is > impossible. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 04/09/2013 08:05 PM, Nick Khamis wrote: > > Hello Everyone, > > I saw an earlier post about this issue: > http://www.mail-archive.com/users at lists.opensips.org/msg23052.html > > And was wondering if there was anything we can do on our end to fix this > problem? It seems that providers are not obligated...

...(Justin Killen) 2. Re: analog phone digit delay (jg) 3. Re: analog phone digit delay (Justin Killen) 4. Re: analog phone digit delay (jg) 5. Re: analog phone digit delay (Steve Edwards) 6. Re: PCI Passthrough of T1 cards (Mauricio Tavares) 7. Re: PCI Passthrough of T1 cards (Nick Khamis) 8. Fwd: AQuA Meter ? waveform analysis to get continous MOS scores for your network (Sevana Oy) ---------------------------------------------------------------------- Message: 1 Date: Mon, 8 Jul 2013 10:14:31 -0700 From: Justin Killen <jkillen at allamericanasphalt.com> Subject:...

...(Justin Killen) 2. Re: analog phone digit delay (jg) 3. Re: analog phone digit delay (Justin Killen) 4. Re: analog phone digit delay (jg) 5. Re: analog phone digit delay (Steve Edwards) 6. Re: PCI Passthrough of T1 cards (Mauricio Tavares) 7. Re: PCI Passthrough of T1 cards (Nick Khamis) 8. Fwd: AQuA Meter ? waveform analysis to get continous MOS scores for your network (Sevana Oy) ---------------------------------------------------------------------- Message: 1 Date: Mon, 8 Jul 2013 10:14:31 -0700 From: Justin Killen <jkillen at allamericanasphalt.com> Subject:...

Hello Everyone, The documentation suggests using unixodbc for asterisk realtime. Is there any way we can just use native database clients such as libmysqlclient from MySQL? The native clients tend to be more up-to-date. Thanks in Advance, Nick.

Hello Everyone, I just ran fsck.ocfs2 on /dev/drbd0 which is a one gig partition on a vm with limited resource (100meg of ram). I am worried that the process crashed because it has not responded in the past hour or so? fsck.ocfs2 /dev/drbd0 fsck.ocfs2 1.6.4 [RECOVER_CLUSTER_INFO] The running cluster is using the cman stack with the cluster name ASTCluster, but the filesystem is configured for

...a2billing and it's LCR functionality. I just wanted to know what other solutions you may be using. Maybe a tool that is a self contained module (ie a2billinig -> Asterisk -> LCR Tool -> Trunk). Is there such a tool? It should be open source as is all good software. Kind Regards, Nick Khamis

Anyone here using Level 3 or AT&T wholesale sip terminations services? I would like to know on any minimums they would require? Also, an idea of how competitive the rates are. I am not asking to disclose your custom rate deck, just a "what to expect". Finally, if you guys can PM me contact info to someone from the wholesale department, I would really appreciate it. Kind Regards,

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.

Hi, OCFS2 document is well documented that it is best practice to isolate the network heartbeat from other traffic. Question: Is it also best practice to isolate the cluster/ocfs2 heartbeat? Please also can you explain why or why not. Regards, Vijay -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Everyone, I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP model. We are looking to interconnect with the PSTN world, and our supplier has given us a few options. We can either do this over traditional PRIs, A-Links or the SS7IP new. I am really interested in SIGTRAN, and was wondering how some of you have integrated it into your architecture. Can Asterisk handle

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.