search for: findmefollow

...together to make it a single 10 digit number but there is something wrong in my syntax probably. $npaa = "('$_POST[anpa]')"; $nxxa = "('$_POST[anxx]')"; $blocka = "('$_POST[ablock]')"; *$grplist = $npaa.$nxxa.$blocka;* $sql="INSERT INTO findmefollow(grpnum, strategy, grptime, grppre, grplist, annmsg_id, postdest, dring, needsconf, remotealert_id, toolate_id, ringing, pre_ring) VALUES ('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]',$grplist,'0','$_POST[postdest]','','...

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

Hellos I am using freepbx and asterisk. I am writing an AGI script to edit the values in findmefollow table. The script will enable users to delete and add follow me numbers from their phones. I want it to enable users enable/disable follow me. I can't seem to find a value in the database that deals with enabling/disabling followme. Please help -- Best Regards, James Mutuku Ndeti Agile Syste...

Hi Everyone, I have done some php coding to come up with my own FollowME module for FreePBX. The need for this has some security considerations behind it. This is what my code does at core: $sql="REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre, grplist, annmsg_id,postdest, dring, needsconf, remotealert_id, toolate_id, ringing, pre_ring) VALUES ('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]','$grplist','0','$postdest','','&...

Hello, I configured both asterisk and grandstream 2000 accourding to howtos on the web.. And everything seems working fin. But if i reload asterisk grandstream stops working with BLF. I need to restart the phone to enable BLF again. Any clues??

...#39;re using a Wildcard TE410P for these incoming calls. I've included the configuration for the ivr and a scrubbed segment from the log. If any additional information is needed, please let me know. Any help is appreciated in advance! Jon [ivr-3] include => ivr-3-custom include => ext-findmefollow include => ext-local include => app-directory exten => h,1,Hangup exten => s,1,Set(LOOPCOUNT=0) exten => s,n,Set(__DIR-CONTEXT=default) exten => s,n,Answer exten => s,n,Wait(1) exten => s,n(begin),Set(TIMEOUT(digit)=3) exten => s,n,Set(TIMEOUT(response)=10) exten => s,...

...on will go into a loop saying 'I'm sorry, that is an invalid response, please try again.' over and over. This does not happen with a commercial digital phone such as a cellphone or Avaya handset. The dialplan follows: [ivr-2] include => ivr-2-custom include => ext-findmefollow include => ext-local include => app-directory exten => h,1,Hangup exten => s,1,Set(LOOPCOUNT=0) exten => s,n,Set(__DIR-CONTEXT=default) exten => s,n,Answer exten => s,n,Wait(1) exten => s,n(begin),Set(TIMEOUT(digit)=3) exten => s,n,Set(TIMEOUT(response)=4) exten...

Hi all, I have a connect between a siemens hipath & Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting "The number you have dialed is not in service" In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk)