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Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150318/0f71f151/attachment.html>

Guru''s, Is anyone shar with me the steps to configure hardware RAID in T2000 server (LSI drivers) and use rootdisk hardware mirror -- Thanks a lot! Malahat Qureshi Ph.D. (MIS) Email: malahatq at hotmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://mail.opensolaris.org/pipermail/zfs-discuss/attachments/20060730/b0bd7aa5/attachment.html>

...eed to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough. On Wed, Mar 18, 2015 at 3:43 PM, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote: > > Hi All, > > I have to forward incoming call on PRI back out to PRI but I need the > > original Callerid to passthrough. Is it possible with DAHDI PRI cards > > without involving the service provider? > > > > Thanks > > It depends who your service...

Hi Everyone, I am following the ''Agile Web Dev. with Rails'' book, and I used instant rails to do the install on a Windows machine. Everything seemed to work fine until I got to: Booting WEBrick... After that, nothing happens. I tried creating a new rails application, but I get the same problem. I also tried reinstalling to no avail. Any ideas? Thanks for the help.

...wer the incoming call, play a message, do some IVR > doodling, or chat with an agent then initiate the transfer. > > Have fun > > --Don > > *From:*asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *Rizwan > H Qureshi > *Sent:* Wednesday, March 18, 2015 7:16 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] PRI Callerid Passthrough > > Thanks AJ and David, > > We were actually using GSM gateways by setting busy forward number on > the SI...

HI Every one Could some one provide me definitions of following bivariate distributions gamma, exponencial, Weibull, half-normal , Rayleigh, Erlang,chi-square thanks A.S. Qureshi

.... I am a wee bit of a programmer also, so if your idea needs changes in the code please dont hesitate to share, otherwise you WILL get a call from me with a special background noise crafted just for you :) Meanwhile i'll try my best to come up with a solution. Cheers -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 3333 6767 26 E: rizwanhasham at gmail.com W: www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110506/460a82e8/attachment.htm>

...ded? transfer would generally answer the incoming call, play a message, do some IVR doodling, or chat with an agent then initiate the transfer. Have fun --Don From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rizwan H Qureshi Sent: Wednesday, March 18, 2015 7:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Callerid Passthrough Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every in...

...at I need to know is: 1. can i register end user with multiple asterisk servers at a time? 2. If not, Can I re-route registeration requests to different servers using 1 asterisk server as a gateway and multiple clustered asterisk servers behind it? cheers Thanks in advance -- Best Regards Rizwan Qureshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101014/0132244b/attachment-0001.htm

...right permissions nothing happens. Is there any workaround to this problem? I need to continue the execution of call files on secondary server if primary server fails. The call files are suppose to retry for 45 mins if the call does not get connected. Thanks in advance. -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131121/ad5df417/attachment.html>

...e, do some IVR >> doodling, or chat with an agent then initiate the transfer. >> >> Have fun >> >> --Don >> >> *From:*asterisk-users-bounces at lists.digium.com >> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of >> *Rizwan H Qureshi >> *Sent:* Wednesday, March 18, 2015 7:16 AM >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* Re: [asterisk-users] PRI Callerid Passthrough >> >> Thanks AJ and David, >> >> We were actually using GSM gateways by setting busy fo...

...these errors before and did manage to get rid of them but I cant remember how i did it and even dont remember the reason for these errors. Looks like a header file for mysql addon is missing which is actually missing (i have checked). How am I suppose to find it? Plz help. -- Best Regards Rizwan Qureshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101006/619c6467/attachment.htm

Hi List, Can anyone please tell me how to use the phoneprov.conf to provision my client's atas. I read the file but dont know how to actually use it. -- Best Regards Rizwan Qureshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101027/7f9de26c/attachment.htm

Hi Everyone, How can I create a TLS based sip trunk between two asterisk servers. I have been trying to do it but i dont know how to defined the client certificate on the asterisk server. Has anyone tried this? -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140910/5e1b8a04/attachment.html>

...pass-through through the outgoing SIM. I have tried every possible configuration on Asterisk that I know of but does not work. At the moment the callerid is always of the SIM we use to send the call out from. Has anyone tried this before? I need this very badly. Thanks -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140917/4f649951/attachment.html>

...1st server and it dies and, we activate the second server but we lose the call files. I could have a cronjob on both servers and create callfiles reading execution time from database, but this involves some other complications. Any crazy ideas would be helpful. Thanks -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131031/c4e5e096/attachment.html>

Hello list, i have one question related to call forwarding. i have 2 number for the inbound and i want to configure asterisk like that. When the customer call the first number 0522XXXXXX the call will be forwarding automatically to anther number 0520xxxxxx Does anybody have a solution to this problem. Thanks and Regards. -------------- next part -------------- An HTML attachment was

...ages WHERE dir = '/var/spool/asterisk/voicemail/default/1757XXXXXXX/INBOX'] I know that the error is caused due to stale odbc connection with mysql. But i want to find out if there is a cure for it. Why the connection went stale in the first place also. Any ideas? -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 3333 6767 26 E: rizwanhasham at gmail.com W: www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110428/b584d5b3/attachment.htm>

...from /usr/lib/ruby/gems/1.8/gems/rails-2.2.2/lib/commands/server.rb:49 from /usr/local/lib/site_ruby/1.8/rubygems/custom_require.rb:31:in `gem_original_require'' from /usr/local/lib/site_ruby/1.8/rubygems/custom_require.rb:31:in `require'' from script/server:3 Thanks Akeel Qureshi -- Posted via http://www.ruby-forum.com/. -- 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, s...

...uri addresses of our customers. My guess is that someone has been sniffing my server's sip traffic. In that case what should i do to get rid of the sniffers? If you think there is another reason for that then please tell me even if you dont have the solution. Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 3333 6767 26 E: rizwanhasham at gmail.com W: www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110228/5075d095/attachment.htm>