similar to: Commercial Asterisk Support

Hi, We've got a customer site with Cisco 7940G Phones connecting to our Asterisk Server via SIP. However, upon running a "sip show peers" it states that the phone is : user/user (Unspecified) D 255.255.255.255 0 Unmonitored All phones can call out via our Asterisk Server fine regardless. The problem is when they try and place a call to one another, where it

Hi there, Anybody on this list knows where I can obtain Hong Kong DID's from ? Cheers, Sahil

Hi, We are terminating around 60 channels on one of our Asterisk boxes, which the client sends in H323 mode. Client (MERA) --> H323 --> Asterisk --> IAX --> Asterisk The problem we face is that at random intervals the H323 process (as part of Asterisk) dies and can no longer accept new calls whilst Asterisk is still running happily. We have to then kill asterisk and start it

Hi, We recently got a PRI installed at one of our local centres however, we appear to be having interesting issues. Everything seems to be installed in the correct manner. asterisk*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203

Hi, We seem to be having an interesting issue with Asterisk whereby, it keeps routing calls coming in to the 'default' context.... regardless of what changes occur to h323.conf. <SNIP> [POP-A] type=user host=1.2.3.4 context=international </SNIP> == Starting H323/ip$1.2.3.4:12914/16313 at default,12126599878,1 failed so falling back to exten 's' == Starting

I wish to second this approach to learning R. I tried for several years to translate other stat programs or provide parallel analyses with R. This dabbling-in-R approach did not work . When a transferred to a research unit that could ill afford commercial software, I devoted my entire time to doing everything in R. This was a difficult learning process, but I eventually became proficient

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

Hi sahil: I think the reason why you cannot get page shared is due to the gref you got. Gref is responsible for a page allocated from domU, in my understanding it should not be 0, that is a gref 0 can not be shared, that''s why I skip gref 0 to be nominated. The gref is nominated to Xen and later used to find a corrspond MFN, so it shall not always be the same.

Hi all I have a following problem. I want to use sjphone to connect to asterisk sip server and then I want asterisk to do a conversion to h323 and send this to h323 gateway. sjphone---sip----ASTERISK----h323-----GATEWAY Example: if someone from plane PSTN line dials 123456 the gateway will forward this to asterisk and asterisk will forward this to sjphone and the other way around. Could

Hi, I'm looking for someone to do time-to-time mantainence on some of our machines going up in New York. The person *MUST* be stationed in New York. Areas of expertise required: - Proficiency in Linux: Slackware, Fedora - Proficiency on Cisco Routers If anybody is interested, please contact me off-list. Regards, Sahil Gupta VoiceValley

Thanks For support. One more doubt is that , As Centos 7 major release is launched , now support for centos 6.x where x is latest mirror point release is supported or not ? ? and further 6.y where y>x will be released or not . You are requested to calrify . On Fri, Aug 1, 2014 at 5:30 PM, <centos-docs-request at centos.org> wrote: > Send CentOS-docs mailing list submissions to >

Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk 1.4.x , but in 1.4.x the oh323 channel don't have more,

Hello all, My name is Sahil Yerawar, Btech 3rd Year Undergraduate student at IIT-Hyderabad. I am currently working on improving vectorization diagnostic remarks in LoopVectorize.cpp. I have some familiarity with LLVM as we studied it in Compiler Engineering Course at IITH. Here is our first effort to improve one of the remarks. https://reviews.llvm.org/D44067 You might want to have a look at

On 9/29/2017 3:37 PM, Rolf Turner wrote: > On 30/09/17 07:45, JLucke at ria.buffalo.edu wrote: > > <SNIP> > >> >> The conceptual paradigm for R is only marginally commensurate with >> that of >> standard statistical software. >> You must immerse yourself in R to become proficient. > > Fortune nomination. For newer list members wondering what

Senior UNIX Systems Administrator Position Summary: A Senior UNIX administrator who can design, implement and maintain a solid, reliable UNIX infrastructure in a heterogeneous FreeBSD/Linux environment. Responsibilities: ? Maintaining FreeBSD/Linux server infrastructure ? Maintaining all firewalls, including Cisco and FreeBSD IPFW rules ? Maintaining all FreeBSD/Linux applications ? postfix,

We had been using rsync and Commercial SSH to sync up two SUN boxes running Solari2.6 and Commercial SSH Version 2.0.12. It has never been a problem syncing between over 80,000 files and 4GB of content We are upgrading to Solaris 8 and OpenSSH_3.6.1p1 However now when I go between the older and the newer environment even with just a few hundred files, the rsync hangs or freezes when it hits a

I am running version 1.1.1 with mail_location: maildir:~/Maildir. This is working great as all our users have UNIX accounts with nologin shells. New domains (and their users) are about to come online and we would like to migrate to a setup with virtual mailboxes/users. From the wiki and comments within dovecot.conf, I see it is possible to do this piecemeal so both local and virtual users

Hello, all How can I write sound file with external G723.1 codec ( actually I have CISCO that can make H323 call to Asterisk box with G723.1 or G729 codec ) I am trying to start Record application by specifying in extensions.conf [writesound] exten => s,1, Answer exten => s,2,Record(soundexample:g723sf) or ...... ( soundexample:g729) I'am using oh323 channel driver, in oh323.conf

Hi List, does somebody know a SIP Provider which offers reseller possibilities? Moritz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040828/bddb5898/attachment.htm

Note I'm not asking for any legal advice here obviously, simply what the intention of the R foundation is with regard to allowing commercial connection to R. I've looked at various threads on the r-devel archive and it looks like this may have been discussed before, but as far as could tell, not to any great resolution, and not, it seems, specifically covering this angle. In the