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...m on a deadline. 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.h...

...So when 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>

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>

...welcome any idea about transfert application as it is a main issue for me: AGI application, Use of Transfert built in Proper use of extension.conf file, Patch to the source code of asterisk (I am able to do such a patch but I don't know where to look... chan_sip? apps directory, other?) Best ragards, Daniel -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com

...in future, 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>

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

...FROM voicemessages 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.h...

...tal.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- 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/dc0831dd/attachment.html>

...the direct sip 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.ht...

hello i am a new user for this group. i am working at a ISP. here i want to made a bridge firewall i am using fedora core 3. i want to block a serirs of ip address 192.16.18.0/255.255.255.0 and want to give the accesss only 172.16.18.0/255.255.255.0. but iptables not be able to block ip;s its passes all the ip series. i made my machine as bridge. i think my bridge passes all the

...com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > Best Ragards > > Rizwan H Qureshi > > V: +971 (0) 528272154 > > linkedin.com/in/rhqureshi <http://linkedin.com/in/rhqureshi> > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20...

sorry for this, but I want know: flac commandline syntax. support ordinary win. com.line ? other words: if I tape at encoding options --cuesheet=*_cdimage.cue, flacenc look it as '*_cdimage.cue' or as ''any'_cdimage.cue'? best ragards... "Opossum" <darckopossum@yandex.ru>

The config file indicates that if mail_location is empty, dovecot will try to find the mailboxes automatically (and by implication determine if the user is using mbox or Maildir format). But if the mbox inbox is non-standard (I have procmail delivering to ~/.mail) then one needs to specify mail_location = mbox:~/mail:INBOX=~/.mail and presumably dovecot will no longer try to determine what

Hi, it seems that there is a bug in asterisk's dtmf mode autodetection. Assume following sip.conf: [sipprovider] disallow=all allow=g726 dtmfmode=auto DTMF does not work. It seems rfc2833 mode is chosen despite it being obvious that this cannot work! The following configuration is necessary to get DTMF to work: dtmfmode=info In my opinion, this behaviour is counter-intuitive. I am using

...wish to start it, however I got such result. IBM*CLI> help jabber No such command 'jabber'. IBM*CLI> help jabberstatus No such command 'jabberstatus'. Any one can help me on this, or may be I miss out somethings that cause jabber applications did'nt install. Best ragards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080219/7091a627/attachment.htm

...wish to start it, however I got such result. IBM*CLI> help jabber No such command 'jabber'. IBM*CLI> help jabberstatus No such command 'jabberstatus'. Any one can help me on this, or may be I miss out somethings that cause jabber applications did'nt install. Best ragards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080222/adafcd9a/attachment.htm

...alled party Extension: +420phonenumber Call manager commands used: Action: login Username: call_manager Secret: call_password Events: off Action: originate Channel: SIP/CiscoPhone Context: sip Priority: 1 Timeout: 30000 CallerID: Martin Edlman <38> Exten: +420phonenumber - -- Ragards, Martin Edlman Fortech, spol. s r.o, Ropkova 51, 57001 Litomy?l Public GPG key: http://edas.visaci.cz/#gpgkeys -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Fedora - http://enigmail.mozdev.org iD8DBQFH8KRoqmMakYm+VJ8RAh/gAKCsObn2hmsvuMqkrsnp9RJoYRBKNQC...

...getting this call only at night time. Asterisk server has no record of this call in log file or cdr. I have also blocked all incoming calls coming from unknown ip addresses etc. Still last night there was a call to a customer. Plz help me figure out the solution for this problem. Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. 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/20110224/76d8ff6d/attachment.htm>

I am using asterisk 1.8.3. I am getting this error: [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because extension not found in context 'smvoice-mediaport'. "dialplan show" gives me that the context is present: [ Context 'smvoice-mediaport' created by

...ipt for the weekly updation of the PDB archive that takes place every wednesday 1.00 PM pacific time.I have at present the GETPDBUPDATE.PL script with me.but I want to know if rsync has got something better to provide. Got something to tell about this, mail me at peeps_anka@yahoo.com Thankyou ragards , Preetha Thomas. __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/