similar to: SIP phone failover using DNS SRV?

Hi all, is it possible that asterisk uses two proxies with SRV? The enddevices are registered on one of the two Proxies (Kamailio). The two proxies communicate with each other. And asterisk can choose one of this proxies with SRV. asterisk | \ | \ Proxy1 Proxy2 I have tries to solve this problem with two trunks for this proxies and Dial(... at proxytrunk) but on this way the

Hi, I am looking to use DNS SRV records for load balancing and failover across multiple Asterisk servers. The Asterisk servers share the exact same configuration via mySQL replication. I would like to know which particular SIP phones support DNS SRV and would like to hear of any success stories. Many SIP phones claim to support DNS SRV, yet there is usually very little documentation on how to

With rtcachefriends = yes in sip.conf, my SIP phone registered to Asterisk Server A cannot dial another SIP phone registered to Asterisk Server B. The error message is: "Cannot create channel of type SIP (Cause 3 - no route to destination)". The two phones _can_ call each other if I set rtcachefriends = no. The common extensions.conf simply uses Dial(SIP/extension) to dial extensions.

I have 8 aastra phones that are loosing registration. On the phone gui it says 408 as the registation error after a minute or say they register. In the cli it eill say the phone is now unreachable then it will show it registering then available. At first they did it every hour all the phones. After messing with the experation it does it every 15 nin. Any ideas on how to troubleshoot this? I tried

I was able to successfully make the master branch of xapian, but I can't do the same for svn/clustering branch. The bootstrap fails with this log: http://pastebin.com/D1hbLp7k Can someone who has successfully installed the clustering branch tell me what am I doing wrong here? Thanks Satwant Rana -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everyone, I was working on the Paice-Husk Stemmer, which is a Bite Size Project for Xapian, and I have created a C++ as well as Snowball version of it. I read the algorithm, and picked the rules from here: http://www.comp.lancs.ac.uk/computing/research/stemming/paice/descript.htm The C++ code takes rules as input from a file and generates the stem of given word, whereas the Snowball version

Hi All. Someone is attacking on my SIP server. There are lot of requests coming in and I am not able to stop it because I am unable to detect the IP address. I used wireshark to capture the packets. Although I am using very strong password for my SIP users but still is there any way to drop these packets and stop this attack. I tried dropping packet after matching some string (most of the

I tried the suggested changes, but still haven't been able to compile the branch. Here's the log: http://pastebin.com/HR17USXR Thanks, Satwant Rana On Sun, Mar 30, 2014 at 2:37 PM, Gaurav Arora <gauravarora.daiict at gmail.com>wrote: > Hello Satwant, > > This seems to be problem with doxygen installation in the bootstrap > script. Source seems to be broken and not

Hi, I created a dummy dialplan where I ask the user to enter the age. [macro-age] exten => s,1,Background(my/age) ;;Play recorded message to enter age exten => s,n,WaitExten(10) exten => _XX,1,Set(AGE=${EXTEN}) ;; this line is not executing, instead dialplan is terminating with error given below. exten => s,n,NoOp(${AGE}) exten => s,n,GotoIf($[${LEN(${AGE})} >

Hi All, I am trying to record the call using MixMonitor. exten=>_XXXX,n,MixMonitor(${EXTEN}.wav,b) What i want to do is- when first time a call is made to some number say 1100, a new file (1100.wav) is created. When call is made 2nd or 3rd time, no new file is created instead call recording is appended to file created in above step. Now I know that 'a' option is used to append the

hi, is there a way of calculating of measuring dependence between two categorical variables. i tried using the chi square test to test for independence but i got error saying that the lengths of the two vectors don't match. Suppose X and Y are two factors. X has 5 levels and Y has 7 levels. This is what i tried doing >temp<-chisq.test(x,y) but got error "the lengths of the two

Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in

Hi All, In asterisk, default directory to store the call-recording files is /var/spool/asterisk/monitor. Can we change this directory? How? -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, There is an option of starting the recording of call after the call is bridged. [ b option]. Is there any way of running an AGI script only if call is bridged otherwise not. Thanks -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part

Hello, all, I have a Linksys 3102 from a VoIP provider. It use SRV record to register to the provider's SIP server. When I configure this line into my Asterisk, the register doesn't work if I use their domain name. So it like this: If I use register => user:pwd at proxy.provider.com then I got: [2010-05-17 11:47:19] WARNING[2366] chan_sip.c: No such host: proxy.provider.com

I am trying to upload the constants to my shared server but built in FTP in Ubuntu is not working -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFF+G/Ez6ZCGd0@public.gmane.org To unsubscribe from this group, send email to

We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values

Hi friends, I haven't done such a simulation before and any help would be greatly appreciated. I need your guidance. I need to simulate end to end data for Reliability/survival analysis of a Pump ,with correlation in place, that is at 'Transactional level' or at the granularity of time-minutes, where each observation is a reading captured via Pump's sensors each minute. Once

Hi all, we have two servers A, B as a failover MGS/MDT pair, with IPs A=10.12.112.28 and B=10.12.115.120 over tcp. When server B crashes, MGS and MDT are mounted on A. Recovery times out with only one out of 445 clients recovered. Afterwards, the MDT lists all its OSTs as UP and in the logs of the OSTs I see: Lustre: MGC10.12.112.28 at tcp: Connection restored to service MGS using nid

Hello LLVM Developers, We are working on extending currently available register rematerialization to include cases where sequence of multiple instructions is required to rematerialize a value. We had a discussion on this in community mailing list and link is here: http://lists.llvm.org/pipermail/llvm-dev/2016-September/subject.html#104777 >From the above discussion and studying the code we