similar to: Concurrent calls including mysql taking lot of time for execution

Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk:

Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh -------------- next part

Hi All, We have built an asterisk server (asterisk - 1.4.26.2) where there would be around 322 concurrent calls going on, but I can see that full log grows rapidly, in one day it reaches to around 10-15 GB if I turn on the sip debug and its tedious even by using any commands to get the required call from the log if there is any problem. Is there any way of splitting the full log into parts

Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter the extension or access code, it some time takes it and some times does'nt recognize the digits dialled. We also tried auto and info for dtmf but could not get the dtmf to work reliably, can

Hi, We are experiencing this issue of redial dtmf tones generated randomly in the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one FXS is used for Fax and rest are empy) connected to the netgear switch and all the phones are connected to this switch and there are no non sip devices in the

Hi,all I did some test about MySQL''s Insert performance on ZFS, and met a big performance problem,*i''m not sure what''s the point*. Environment 2 Intel X5560 (8 core), 12GB RAM, 7 slc SSD(Intel). A Java client run 8 threads concurrency insert into one Innodb table: *~600 qps when sync_binlog=1 & innodb_flush_log_at_trx_commit=1 ~600 qps when sync_binlog=10

Hi All, We are using group paging and our asterisk version: 1.4.22.1, but when ever any one page to the whole group (28 extensions), the calls which are on hold on those extensions will be dropped, is there any other way to have this feature or to go with Overhead paging. Currently this has become a serious problem, can anyone through some light on this group paging senario? Thank you very much

Hi all, When i do ''xm dmesg'' the last statement says "*** Serial input -> DOM0 (type ''CTRL-a'' three times to switch input to Xen)" (i have no clue what''s that supposed to mean??) But when i press ctrl-a three times at the serial console, nothing happens. Iam using minicom to connect to the serial port of xen machine. Once xen

Hi, I am trying to release domU pages from page_list and xenpage_list after domU shutdown while retaining the rest of the domain information. To achieve this in __domain_finalise_shutdown i call domain_relinquish_resources. This is failing to release pages from page_list for type PGT_l2_page_tables and crashing dom0. To be specific, while testing on mini-os i saw that when

device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote: Hi, We have few systems with asterisk 1.4.22.1 and we use sip

Hi All, Today I faced a problem with channels getting stuck. We use asterisk 1.4.18.1, and there were 2 extensions (channels) that got stuck. When I try to do "soft hangup <channel>", it says "Requested for soft hangup" for that channel, but if we go and check once again those channels are still stuck. Also even after asterisk restart it did'nt go, finally we had to

Hi We are facing a problem for orphaned parked calls, we have the following config: asterisk -1.4.22.1 dahdi-linux-complete-2.2.0.2+2.2.0 and when we get an incoming call and after it gets parked, after some set time (here its 2 min), it goes back to the operator, but the problem is that randomly it tries to call SIP/5060 instead of SIP/2200 (where 2200 is the extension number of the operator)

I am working through Tom Minka's lectures on Data Mining and am now on Day 32. The following is the link: http://alumni.media.mit.edu/~tpminka/courses/36-350.2001/lectures/day32/ In order to use the functions cited I followed the instructions as follows: Installed tree package from CRAN mirror (Ca-1) Downloaded and sourced the file "tree.r" Downloaded the function

Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director

Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. Thanks, Regards, Arstan Jusupov -------------- next part -------------- An HTML attachment was

Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc

Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News -

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26