Re: [Asterisk-Users] Voice Mail App
On Wed, 2003-10-01 at 21:08, PBX wrote: > I have a general question. Can the voicemail app be rewritten? If so, > is it written in C / C++ now? Cudos for starting a new thread. Bummer you used HTML. Yes voicemail could be rewritten, this is the beauty of open source software. It is written in C like the rest of asterisk. Expect that anything other than C will be unlikely to be re introduced into asterisk core. Also if you plan on doing any patching, you need to look over the disclaimers from Digium. They need you to sign on and send it back before they accept patches. If you need development help further, please ask around on the asterisk-dev list. You will find the topics are more development based. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice Mail App
You mentioned that Asterisk is programmed using C. Can apps be written in C++ also or only C. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Posted At: Wednesday, October 01, 2003 10:23 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Voice Mail App Subject: Re: [Asterisk-Users] Voice Mail App On Wed, 2003-10-01 at 21:08, PBX wrote: > I have a general question. Can the voicemail app be rewritten? If so, > is it written in C / C++ now? Cudos for starting a new thread. Bummer you used HTML. Yes voicemail could be rewritten, this is the beauty of open source software. It is written in C like the rest of asterisk. Expect that anything other than C will be unlikely to be re introduced into asterisk core. Also if you plan on doing any patching, you need to look over the disclaimers from Digium. They need you to sign on and send it back before they accept patches. If you need development help further, please ask around on the asterisk-dev list. You will find the topics are more development based. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice-mail App
At 13:39 25-2-2003 -0600, you wrote: In case you have never setup a 7960 before. The easies way is to setup dhcp and have the code on a tftp server. option tftp-server-name "your.tftp.edu"; If you need a copy of the xml configs or dial plans for the phone let me know, and I send some your way. There is a nasty bug with REFER in the SIP code for that phone before version P0S3-04-3-00.bin I'd love to see one of these XML files too. Apparently the ATA186 can also use such a format, so it would be neat to get one of those samples too :-)) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice-mail App
A few note about each file. OS79XX.TXT: Should always have this old version of the code for the phone. Ringlist.xml: Lets you have custom ring tones (not a good to have Bart saying eat my shorts as your ringer) SIP-MAC-ADDRESS.cnf : Phone setup goes here. Set You telnet password if you want to log into the phone and do sip debugging. SIPDefault.cnf: Default values, note this is were you tell your phone what version of code to download. dialplan.xml: The dialplan for the phonemostly to set timeouts before dialing. John Todd wrote: I'm interested in the example xml configs, if you have some to send. I'm getting some 7960's shortly, and would appreciate the hints. JT In case you have never setup a 7960 before. The easies way is to setup dhcp and have the code on a tftp server. option tftp-server-name "your.tftp.edu"; If you need a copy of the xml configs or dial plans for the phone let me know, and I send some your way. There is a nasty bug with REFER in the SIP code for that phone before version P0S3-04-3-00.bin Mark Spencer wrote: I have played with the timeouts :int timeout, int ftimeout for ast_readstring(chan, password, sizeof(password) - 1, 2000, 1, "#") To no effect, Could you give me a pointer to where I can start looking next to track down this strangeness. Maybe something in the SIP driver? Getting in and out of band DTMF? etc... Best place to look is probably the RTP code, where the digits are generated. Specifically look at the rfc2833 routines, assuming that's how they're being sent. I just got a 7960 on loan, so I'm going to set it up so that I can try to duplicate any problems you're having. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users xml-7960.tar.gz Description: GNU Zip compressed data
Re: [Asterisk-Users] Voice-mail App
Indeed rtp.c: static int dtmftimeout = 400; /* was 300 samples */ Did seem to fix the problemThat was almost too easy. All hell will probable brake out tomorrow when real users get on the system. Mark Spencer wrote: I have played with the timeouts :int timeout, int ftimeout for ast_readstring(chan, password, sizeof(password) - 1, 2000, 1, "#") To no effect, Could you give me a pointer to where I can start looking next to track down this strangeness. Maybe something in the SIP driver? Getting in and out of band DTMF? etc... Best place to look is probably the RTP code, where the digits are generated. Specifically look at the rfc2833 routines, assuming that's how they're being sent. I just got a 7960 on loan, so I'm going to set it up so that I can try to duplicate any problems you're having. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
