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Where can I learn the syntax for DBput and DBget? is it working with MySQL? do I need to set up tables? URiel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030508/a2598dc8/attachment.htm

After rebooting my asteriks server, e-mail notifications are no longer being sent after a voice-mail is left. I can see the messages in /var/spool/asterisk/vm. has anybody had the same experience? how was it resolved? Uri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040410/fc494bb4/attachment.htm

Every time I send an e-mail to the * list, I receive this "AntiSpam UOL" E-mail. is anybody else experiencing the same? How can I get rid of it? Uriel -----Original Message----- From: AntiSpam UOL [mailto:andersoncbr.sspam@uol.com.br] Sent: Wednesday, September 24, 2003 11:51 PM To: uriel@adelphia.net Subject: RE:RE: [Asterisk-Users] SIP / GrandStream Configuration Ol?,

I have two loop-start phone lines connected to an Adtran TA 750 channel bank. When a call comes into asterisk thru the first phone line and the call is redirect (in transit) back thru the second phone line, the channel bank will leave one or both lines opened (ie hangup with spawn a connection again). does any body have a suggestion: 1) I checked the cables into the channel bank from both phone

...groups, then > >>; you can answer it by picking up and dialing *8#. For simple offices, just > >>; make these both the same > >>; > >>callgroup=1 > >>pickupgroup=1 > >> > >> > >>Jeremy McNamara > >> > >>Uriel Carrasquilla wrote: > >> > >> > >> > >>>Does anybod have a suggestion on how I can pick up a call from my > >>>extension for another extension that is ringing. I have a channel > >>>bank with multiple extensions and when a call comes in for s...

...r it by picking up and dialing *8#. For simple >offices, just >> >>; make these both the same >> >>; >> >>callgroup=1 >> >>pickupgroup=1 >> >> >> >> >> >>Jeremy McNamara >> >> >> >>Uriel Carrasquilla wrote: >> >> >> >> >> >> >> >>>Does anybod have a suggestion on how I can pick up a call from >my >> >>>extension for another extension that is ringing. I have a >channel >> >>>bank with multiple extensi...

I have one of my asterisks off an xDSL modem with a dynamic IP address. The IP address keeps on changing so I am curious as to what other people are doing to work around this issue. I am contemplating to register for a dynamic DNS service but then my communications become dependent on the availability of this service. The "host=dynamic" for SIP seems interesting. Any suggestions?

On Wed, 30 Apr 2003 00:24:19 -0400, Uriel Carrasquilla wrote: > >Gary: >I just copied the content from chan->exten to chan->dnis. I am calling from How are you doing this coying ? >one extension to another. >Have you got DynExtenDB to work? nope, haven't got over the first problem yet. Gary .

I was looking at the shell in /usr/sbin and the following statement does not seem right: while :; (read colon + semi colon). In C that would be 2xsemi-colon. is this correct in a bash shell? I am having problems with safe_asterisk. when it dies, it is not being brought up again. It works very well for stop/start/startus but does not bring up asterisk when it fails. -------------- next part

vmail.cgi rocks (if I can borrow the expression for Mark Street). As Mark pointed out, the /vm/INBOX messages are created with 0700 security and vmail.cgi is not happy. Apache/cgi/vmail.cgi cannot play them unless I fool around with the Apache wrapper or chmod 755 *.* thefiles myself. (Tedious, that is why I like computers). Obviously this is not acceptable. I took a trip to the

I have been fooling around with DynExtenDB and run into two glitches. 1) The code is looking for (chan->dnis) and in my case I find (null). I forced (chan-dnis) to be the same as (chan->exten). So far so good. Now I can connect and talk. This lead me to the second glitch. 2) As soon as the call ends by hanging up, the code issues a (ast_spawn_extension). This causes asterisk to drop

Can anyone give me presise instructions on how to compile cdr_mysql.so? When I initially installed asterisk on the system, I didn't have mysql installed. Since then I have installed mysql, created the database and table structure for cdr_mysql and placed the appropriate settings in the cdr_mysql.conf file. However when I do a show modules at the CLI I cannot find cdr_mysql.so.

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Hi everyone (Mark, Jim). I am new to the list but thanks to both Mark and Jim, I have being using "asterisk" since summer 2001. I am just updating my version that was a year old. Yes, I know, I got busy with other things like paying bills so I don't have to sleep with the dog anymore. Anyway, one of the frustrations I have been dealing with (keep in mind that my version of

...No such device (Anon) > 3. newbie - Asterix and modem cards (Alex G > Robertson) > 4. Re: newbie - Asterix and modem cards (Anon) > 5. RE: Booting error - Unable to specify channel > 2: No such device (Todd Lieberman) > 6. RE: vm e-mail notification stopped (Uriel > Carrasquilla) > 7. Re: Booting error - Unable to specify channel > 2: No such device (Anon) > 8. Re: Adding two FXO cards - not working > (Hermann Wecke) > 9. editing errors/typos in rev 2 of The Asterisk > Handbook (current > version on digium's site) (Andrew D Kirch) &...

All, is anyone using Bayonne in conjunction with Asterisk? I'm currently using only Bayonne, but I'm investigating the possibilities of switching the telephony frontend over to Asterisk, and have Asterisk route the IVR tasks to Bayonne through H323. Anyone care to share his views on this approach? Any pointers or do's and don'ts? All info is greatly appreciated! Regards,

I have seen a couple of messages on the Xten and the work done by William Walsh (Kudos). It is not clear in my mind the advantages of SIP phones versus using GnoPhones (once we complete the work for the Windows version). Since I lack the experience with IP SIP phones, can someone, high level, tell me when it makes sense to use them. Is it complicated to set up on the Asterisk side? Thank you.

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

Below you will find, what I believe to be a typical setup with a T100P card. My question is - 1. Is this correct? 2. What kind of phones would be needed here... (Would you have to use Digital phones) And if so what would you recommend. PRI/T1----- | | | ---------------- | | | Channel Bank | | |

I have modified the content of some of the vm-*.gsm files in /var/lib/asterisk/sounds. The only problem is that everytime I recompile apps_voicemail.c the original vm-*.gsm files overwrite the content of /var/lib/asterisk/sounds. My CVS is 04/07/03. I used to just be able to make modifications to the running system, then copy from /var/lib/asterisk/sounds/vm-*.gm to /usr/src/asterisk/sounds/ and