search for: jasonsolves

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

I am getting this error [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 8f68421-22821e1e at localhost for seqno 102 (Critical Response) [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging up call 8f68421-22821e1e at localhost - no reply to our critical packet. any Ideas? Jason

I used to have fonality and I could press * when I got to someones voice mail to go back to the menu. I assume I add that to the dialplan but how? Thanks BTW I went back to 1.2.12 and transfer works and DTMF works and it seems to be much better for now. Thanks for you help Jason

I put my voicemail groups into different contexts so that I can use Dial by name and escape. I had set ext 500 as exten => 500,1,VoiceMailMain(${CALLERID(number)}@default|s) but now that the contexts are different. this does not work #1 how do I have everyone use an ext to get the voicemail regardless of context. #2 can I get the mail buttons to work on my polycom 501s and swissphones #3

I have 1.2.12.1 Voicepulse using IAX I get about 30-40% issues with not having the DTMF tones work. I have 3 questions #1. Voicepulse says they are sending them, Is there some setting I can adjust to make sure my end is working? #2. I have set the Dialplan to play a sound "Operator" then go to a SIP phone. I would like it to write Caller ID Time .... to a file I can read and find

I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2 digits into the dialplan. error -- Invalid extension '81' in context 'impact' on SIP/207.174.111.34-b77167f8 I pressed 8107 and ideas my dial plan is (part of it) [impact] exten=>s,1,Answer() exten=>s,n,Set(CALLERID(name)=Impact - ${CALLERID(number)}) exten=>s,n,Background(IMPACT)

I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct DTMF tones 25% of the time. I have to call several times to enter an extension. I have a router and a packet shaper and some other stuff. Anyone have any other ideas why this might happen. I do not have any Zap channels but I am running CentOS4. I also do not have any cards installed. Thanks

I am having a bunch of issues with 1.4 and want to go back to 1.2 any ideas on the best way I saw someone say "apt-get remove" will this work for asterisk or do I need to do it for each libpri, addons, zaptel and asterisk? Thanks Jason

I have users in my dialplan that go from SIP to Cell When they are at their desk and they hit reject call, it goes to the next thing in the dialplan, thus transferring to their cell. Not what they want. Is it possible to change the reject button to make it go to voice mail or a new ext? Thanks Jason

From what I know this log show everything working Perfect. Then it goes to the Welcome screen then after a long time of processing, it errors out with a 0x10000 error Any Ideas? 1005195711|so |4|00|---------- Initial log entry ---------- 1005195711|so |4|00|+++ Note that bootrom log times are in GMT +++ 1005195711|hw |4|00|Initial log entry. 1005195711|wdog |4|00|Initial log entry