search for: rizwan

...of the dialed no. i need to set it to DNID because our every user has 5 dids and i want to show the caller at the end of the month which numbers he dialed for every call, along with other cdr info. Our rating depends on the dialed number also. here is my extensions.conf exten=> 1212,1,Dial(SIP/rizwan) ;Primary did for user rizwan exten=> 1212,2,Hangup exten=> 1714,1,Goto(,1212,1) ;Secondary did for user rizwan here if the caller dials 1714 the call is jumped over to 1212, and the CDR(dst) field after hangup is set to 1212. i need to set it to 1714 even if the caller is on ex...

...== Everyone is busy/congested at this time (1:0/0/1) here is my configuration SERVER A sip.conf ~~~~~~~~~~ [general] register=>fromtrixbox:mysecret@192.168.0.7 [receive] ;this should receive call from server B type=friend ;host=192.168.0.7 host = dynamic secret=mysecret context=rizwan ;trunk=yes insecure = very [backtorazi] ;this is used to throw call to server B (works fine) type=friend host=dynamic secret=mysecret context=rizwan SERVER A extensions ~~~~~~~~~~~~~~~~~~~ [rizwan] exten=> 5123,1,Dial(SIP/backtorazi/1234) ;this dials to server B exten=> 5123,2,Hangu...

...ser is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R and r options in Dial application but they dont work. -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070530/ac78db36/attachment.htm

.... Im not sure if this can be done. I am going to try doing it somehow, meanwhile i want your suggestions about how i can share a mysql connection with different calls in a dialplan. I am using asterisk1.4.2 and asterisk addon1.4.0 package for mysql connectivity. Thanx in advance -- Best Regards Rizwan Hisham -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080806/08562916/attachment.htm

Is it possible to select a portion of string? For example, in many websites, they give the first paragraph in the main page and link for the article on the other page. I remember this is available in PHP. Any help is appreciated. Thanks Rizwan -- Posted via http://www.ruby-forum.com/.

...e incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070817/9706efcf/attachment.htm

...eadline. 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>

...i grant 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>

...a fully developed web based remote provisioning system. I cant find anything reliable on the internet. Have already checked ataconfig.com and voxilla-ays.com. have tried to contact them but got no response. So if anybody knows a good provisioning system then plz tell me about it. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071024/0a228d78/attachment.htm

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>

...s my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from different IPs for the same username. Can anybody help? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070911/62694010/attachment.htm

...ur settup. we cant just ask the customers to change their ports for registration (many of them dont know what a port is) Is it possible to make single asterisk server listen on two different ports? if there is any other option we can use, plz inform me about it. Thanx in advance. -- Best Regards Rizwan Hisham -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081117/8a33aba7/attachment.htm

...e, on the 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>

...e seen 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

...annel to execute h extension, but is it possible to execute the h exten even then. i want the h extension to excute everytime. how can i do this. i have used the g flag in dial which tell asterisk to execute remaining extensions even after hangup but its not doing in the above described case. -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070509/e6643e06/attachment.htm

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080125/4f833de8/attachment.htm

...ing. What 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

...Now we need 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 yo...

...ssible? For example, I have two controllers named ''upload'' with an action list_uploads and a controller named notes with list_notes action. I want to bring both of these action under a single view which could mean a table with list_notes and list_uploads. I thank in advance! Rizwan Reza -- Posted via http://www.ruby-forum.com/.

...unsuccessful gexp_1$x NULL #or gexp_1$x[1] NULL #but this works gexp_1$geneA I am sure that this is a fundamental error on my part since I am new to R, I would appreciate any advice. I am trying to construct a for loop to analyze data for sets of genes at a time but stuck at this step. Regards Rizwan