similar to: Performance of realtime for millions of SIP user

Hi, I have a simple dialplan. [test] exten => _X.,1,Dial(SIP/1000&SIP/1002) I have registered user whose context is test. Now I am dialing any number, so it will enter into test context. It will dial 1000 & 1002 both. Both keeps ringing. Now the problem is, when any of them answer, another one keeps ringing for 10 to 15 sec. Please help me , what's wrong here. Thanks, Krunal

Hi, I have a query on regression output generated by R. > result=lm( Y~X , data=trail) > summary(result) After running this 2 statements the following output is generated. Call: lm(formula = Y ~ X, data = trail) Residuals: Min 1Q Median 3Q Max -245.30 -90.77 -30.30 54.99 532.78 Coefficients: Estimate Std. Error t value Pr(>|t|)

Hi, I am trying to run many different regressions using a FOR Loop. The input data that is read into R has the following variables · Volume · Price1 · Price2 · Price3 · Price4 · Price5 · Trend · Seasonality I want to run 5 regressions, with the Volume as an dependent variable and Price, Trend & Seasonality as

Hi, During tests with a IAX2/PSTN gateway I've been getting strange results for processor idle time and load. I (re)search(ed) this issue for a while, but I didn't get any good explainations. Can somebody help me? I have several sites that rely on a central server for connection to the PSTN. Calls to the PSTN are routed over the Internet to this PSTN gateway using IAX2 in trunk mode. To

Anybody sucessfully got stable 1000Hz clock without Digium harware and kernel 2.6? We need to consult some peoples how to clock asterisk stable with exactly 1000 Hz without much kernel/drives patching/tweaking. Some test results we made so far: 2.6 with digium card - stable 1000 Hz. 2.6 with ztdummy - uses RTC and the clock is 1024, not 1000. 2.6 with some Realtime kernel patch - provides stable

Hi, Here is the scenario: Originating local channel using AMI. On answering the channel, it will goes to a context. Which start to playback a file. & after hangup at h extension I am caliing an agi script which insert CDR into DB. Now the problem is when I script hangsup during payback CDR(billsec) returns currect result. But when it hangsup after playback cdr(billsec) returns 0 . Please

hi all, I can claim that our hotfoon service has run millions of mobile voip minutes using speex on mobile and desktops. it works! earlier you needed more powerful cpus (until the integer port), however now the cpus are much faster and the code complexity is much lower. my only regret is of not being able to write some demo apps for speex on win mob. I will surely deliver it in june to settle this

Dear ALL I want to transfer call from one phone 2 another phone so this is asterisk feature or SIP Phone feature or endpoint feature how can i transfer phone call from to another phone Rgd Satish patel --------------------------------- Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. -------------- next part -------------- An HTML

Dear all, i am currently evaluating which codec to be used for a "very narrowband" voice communication system (using either VoIP either GSM CSD) on mobile devices running Windows Mobile and Symbian OS. I had experience in using AMR-NB 4.75 with VAD on Symbian OS and all worked very well. However, AMR it's not royalty free and if i need to use it on Windows Mobile i will need

I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter "localnet" defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have

Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S -------------- next part -------------- An HTML attachment was scrubbed...

Hi All, Just wondering is it safe to use asterisk 1.8 latest branch on production ? http://svn.asterisk.org/svn/asterisk/branches/1.8/ Revision 317100 -S -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110505/373bd6fc/attachment.htm>

Hi All, I am new to this community and just subscribed. We have Asterisk running in production but I could not find out in documentation as well as web that how many maximum number of registrations an Asterisk Server can support. We have it on a 1.4 GHz Processor, 2 GB RAM and 40 GB HDD IBM Server. Please suggest urgently. Thanks. Best Regards, ------------------------------------- Abid

I am using the current HEAD of asterisk and for asterisk-addons. I have been trying to setup realtime mysql voicemail but no sucess. I keep getting this error below. The necessary modules are loaded, res_config_mysql.so ... Any pointers will help. Thanks mail*CLI> realtime mysql status Aug 24 00:56:50 ERROR[963]: res_config_mysql.c:596 mysql_reconnect: MySQL RealTime: Failed to connect

This no going to work. AS I told you I need something under guestfish command list not from guestfs commands. Thanks, Chintan ________________________________ From: Nikolay Ivanets <stenavin@gmail.com> Sent: Friday, June 28, 2019 10:58 AM To: Chintan Patel Cc: Chintan Patel; libguestfs@redhat.com Subject: Re: [Libguestfs] Guestfish command - "copy-out" not working for symbolic

Hi, The asterisk wiki page says: "Sorcery.conf allows you to try to configure other PJSIP objects such as transport using realtime and it currently won't stop you from doing so. However, some of these object types should not be used with realtime and this can lead to errant behavior." Which objects and is this still true in 1.13.1 ? Thanks, Antonio. PS:

guestfish --ro -a /path/to/disk run : mount /dev/your-blkdev / : download /etc/resolv.conf /path/on/host/dst.file -- +380979184774 Mykola Ivanets пт, 28 черв. 2019 о 21:29 Nikolay Ivanets <stenavin@gmail.com> пише: > > All API is exposed as gustfish commands > > -- > Mykola Ivanets > > пт, 28 черв. 2019, 21:28 користувач Chintan Patel

Hello, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. Thanks! Matt Hoskins | NPG Corp | Systems Architect

HI, Does asterisk will ignore the setting in files if realtime is applied, say asterisk will ignore all the setting in sip.conf if realtime table sip_buddies is applied? ango

I have a pjsip install that is not pulling it's realtime registrations from an ODBC database. When I have them directly pull from a MySQL database everything seems to work. I am having trouble finding a requirements document for the pjsip realtime registrations. Is there some kind of special config that has to be used to trigger the connection for realtime registrations over ODBC?