similar to: Nokia cell connectel to asterisk

Hi List; If I configured one SIP account or one IAX account [sipuser1] or [iaxuser1] then how many calls can be originate/terminate using the same account [sipuser1] or [iaxuser1]? In other words, can 10 IP Phones (users) do a calls via Asterisk using the same account (SIP or IAX2)? If yes, how can I control the number of calls per account? Regards Bilal

Hi, Can somebody please shed some light on how to use the QUEUE_VARIABLES() function? The built-in help says ---cut--- Return Queue information in variables [Description] Makes the following queue variables available. QUEUEMAX maxmimum number of calls allowed QUEUESTRATEGY the strategy of the queue QUEUECALLS number of calls currently in the queue QUEUEHOLDTIME current average hold time

Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password

Hello All We have developed an application for Home Delivery Restaurants using Asterisk, Java (JSP/ JSF) and MySQL. Here is listing of its features. If someone is interested then we can provide him more details. - POP up window with caller data containing his/her name, address and transactions history. - In case of new customer, Pop up window with blank form to add customer data and

I am running r79979 of Asterisk Trunk, and I am having problems trying to use app_queue.so. I want to use the extension 510 to be a line where users can call technical support. Extensions 511 and 512 are used by the operators to dynamically make themselves a Queue Member or not. So, operators call 511, and they should get added to the Queue as a Queue member. When users call 510 then, it

Hello list, I have defined a new MoH-class in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; *[106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes* In sip.conf I have this commented out : ;mohinterpret=default ;mohsuggest=default Asterisk sees these moh-classes and files : vps2301*CLI> moh show classes Class: default Mode: files

Hi Mohammad, A lot of time is being spent in addressing metadata calls as expected. Can you consider testing out with 3.11 with md-cache [1] and readdirp [2] improvements? Adding Poornima and Raghavendra who worked on these enhancements to help out further. Thanks, Vijay [1] https://gluster.readthedocs.io/en/latest/release-notes/3.9.0/ [2] https://github.com/gluster/glusterfs/issues/166 On

Hi Kashif, Thank you for your feedback! Do you have some data on the nature of performance improvement observed with 3.11 in the new setup? Adding Raghavendra and Poornima for validation of configuration and help with identifying why certain files disappeared from the mount point after enabling readdir-optimize. Regards, Vijay On 07/11/2017 11:06 AM, mohammad kashif wrote: > Hi Vijay and

Hi Vijay Did you manage to look into the gluster profile logs ? Thanks Kashif On Mon, Jun 12, 2017 at 11:40 AM, mohammad kashif <kashif.alig at gmail.com> wrote: > Hi Vijay > > I have enabled client profiling and used this script > https://github.com/bengland2/gluster-profile-analysis/blob/ > master/gvp-client.sh to extract data. I am attaching output files. I >

I am talking about the SIP. Now the new mobiles (Nokia, Erecson, Panasonic, iPot, ... etc) all of them support SIP capability. They are able to register to any SIP server (by giving the IP address, username and password). Fring is one of the software that can be installed on the mobile devices and can register on the SIP servers. BUT, the new mobiles currently come with built in SIP (no need to

Thank you for the acknowledgement. On Thu, Aug 31, 2017 at 8:30 PM, mohammad kashif <kashif.alig at gmail.com> wrote: > Hi Atin > > Thanks, I was not running any script or gluster command. But now gluster > status command started working. CPU usage also came down and looking at the > ganglia graph, cpu usage is strongly correlated with network activity. > > It may be

Hello All We need to connect our client's offices located in Dubai and Pakistan. Suggest us some economical solution. -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kashif at haditelecom.com MSN: kashif__naeem at hotmail.com Gmail: meet.kashif at gmail.com Skype: kashif.naeem 302 Y Commercial Area,

I am having a problem with the Call Queue Feature. If I let a user into the Queue prior to an agent being available for them to take the call, they experience the following: 1) they hear that they are the first in line 2) when the agent finally logs in (the caller on hold in the queue is sent to the users phone) 3) the AGENT is still in the login phase hearing that they are "successfully

Hello All We have started a voice portal for Parents and Students. They can listen Grades, Attendance status and other relevant information over *phone*. Please read features below and to listen Demo IVR call at *00 92 42 8315427. *Initially we are deploying it for a School and planning to spread it for Colleges and Universities. *Your comments and feedback are valuable to us.* *TeleKaam

I rather like agedu It probably does what you want. But as Mohammad says you do have to traverse your filesystem. https://www.chiark.greenend.org.uk/~sgtatham/agedu/ agedu: track down wasted disk space - chiark home page<https://www.chiark.greenend.org.uk/~sgtatham/agedu/> www.chiark.greenend.org.uk agedu. a Unix utility for tracking down wasted disk space Introduction. Suppose

Hello All One of our client Bank has 900 employees working in different locations. They need to record all internal and external calls. Can any body suggest Call Recording Solution for this requirement. We need to know the Hardware / Bandwidth and all requirements and costing. Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006

Hi Alex, John Thanks for confirming my suspicion that there is no getting away from POSIX tree traversal . I was aware of age-du but not robinhood. Cheers Kashif On Wed, May 2, 2018 at 8:57 AM, JOHE (John Hearns) <JOHE at novozymes.com> wrote: > I rather like agedu It probably does what you want. > > But as Mohammad says you do have to traverse your filesystem. > >

Would it be possible for you to turn on client profiling and then run du? Instructions for turning on client profiling can be found at [1]. Providing the client profile information can help us figure out where the latency could be stemming from. Regards, Vijay [1] https://gluster.readthedocs.io/en/latest/Administrator%20Guide/Performance%20Testing/#client-side-profiling On Fri, Jun 9, 2017 at

Dear All, We are trying to install CDR Stats module. We are able to open its web pages but unable to retrieve data from CDRs. Can anybody suggest that how to connect this module with Asterisk CDRs database ? Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kashif at haditelecom.com MSN:

Hello, We are going to start development for a product based over Asterisk. According to you, which is the preferred language for AGIs / IVRs development in Asterisk. I got opinions that Perl is going to be replaced by PHP for all future developments. -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kashif