search for: helpexit

...SIP/teliax-00000000> Playing 'vm-no.ulaw' (language 'en') -- <SIP/teliax-00000000> Playing 'vm-messages.ulaw' (language 'en') -- <SIP/teliax-00000000> Playing 'vm-opts.ulaw' (language 'en') -- <SIP/teliax-00000000> Playing 'vm-helpexit.ulaw' (language 'en') In this case, I'd hear "gin" "essages". The 'password', 'youhave', and 'no' prompts are actually so short you don't hear them at all. http://help.teliax.com/discussions/support/1924-asterisk-1620-rc6 --- I...

Dear sir, after 3 month with asterisk i start SIP A-Z provider for test my solution. Please take a look and test Or send live traffic he he http://msarn.com Dome C. ----------------------------------------------------------------

...'vm-no' (language 'en') -- Playing 'vm-messages' (language 'en') -- Playing 'vm-opts' (language 'en') -- Playing 'vm-starmain' (language 'en') -- Playing 'vm-opts' (language 'en') -- Playing 'vm-helpexit' (language 'en') -- Playing 'vm-options' (language 'en') -- Recording the message -- Playing 'vm-rec-name' (language 'en') -- Playing 'beep' (language 'en') -- x=0, open writing: voicemail/default/100/greet format: wav...

...1-01d646b0> Playing 'vm-no.gsm' (language 'en') -- <SIP/line1-01d646b0> Playing 'vm-messages.gsm' (language 'en') -- <SIP/line1-01d646b0> Playing 'vm-opts.gsm' (language 'en') -- <SIP/line1-01d646b0> Playing 'vm-helpexit.gsm' (language 'en') DTMF begin '3' received on SIP/line1-01d646b0 DTMF begin ignored '3' on SIP/line1-01d646b0 DTMF end '3' received on SIP/line1-01d646b0, duration 130 ms DTMF end passthrough '3' on SIP/line1-01d646b0 -- <SIP/line1-01d646b0>...

Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C