Re: [asterisk-users] Addons
Hi If you use debian install the libmysqlclient-dev package David a écrit : Hello Asterisk-Users, I'm trying to install addons 1.2.6 on Asterisk 1.2.16 (is that OK?), but my MySQL server is installed on a different sever, so the MAKE of the addons fails with the following (truncated) error message: app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory. Is there any way to bypass/ignore the fact that MySQL is installed separately and enable the installation of the addons? Thanks, David Get the Yahoo! toolbar and be alerted to new email http://us.rd.yahoo.com/evt=48225/*http://new.toolbar.yahoo.com/toolbar/features/mail/index.phpwherever you're surfing. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple host= in sip.conf
Not supported jsut use host=dynamic with username and secret. Alex Yusuf a écrit : Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw canreinvite=no [vz3] context=outbound type=friend host=.z.z.z.z disallow=all allow=alaw canreinvite=no However, is there anyway I can have just one account for him, with mult host= statements, so I can authenticate him based on his IP in just one place? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SCCP + hint
Thanks for your reply I use asterisk 1.4.4. Thanks in advance. Cheers, Alex. Michiel van Baak a écrit : On 12:19, Wed 23 May 07, Alexandre VERNIOL wrote: Hi all, Does someone know if it's possible to use hint function with skinny ? Can anyone send me an example ? Thanks in advance, Alex. What version of asterisk are you using? hints on chan_skinny work in -trunk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SCCP + hint
Hi all, Does someone know if it's possible to use hint function with skinny ? Can anyone send me an example ? Thanks in advance, Alex. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Echo
Hi, Could be bandwith or/and latency ... Many causes... Alex Asterisk a écrit : Hello all, One of our clients reported that they are experiencing echo in SIP calls (even on internal ones). What do you think could be causing echo in internal SIP calls? We're using Polycom telephones, do you think they could be causing it? Thanks, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing - AstJax click2call Firefox greasemonkey script - click and dial phone numbers in any webpage
Really Great!!! Works for me in France I have just change the pattern and that's ok reallygood job! Cheers, Alex Richard Hamnett a écrit : Hi there, Just to announce that I've improved upon a greasemonkey script which allows users to dial any number (in the given regex format) by turning it into a clickable hyperlink. The script uses greasemonkey's ajax callback to a simple php controller script, so that the click does not navigate away from the current page. It requires an Asterisk Manager connection. See http://yaptele.com/asterisk-firefox-click-to-dial-ajax-script for more details. Kind Regards, Richard Hamnett ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help installing on OpenSuSE 10.2
make[1]: g++: Command not found You have just to install cpp Alex, Malcom Kemp a écrit : Thanks to all that have helped me so far. I have made a lot of progress. I am able to make prilib and zaptel. Now to Asterisk… After installing the kernel source, I have: # cd /usr/src/linux # make cloneconfig # make prepare-all Then I have run ./configure in the asterisk-1.4.4 directory. I have: # make clean # make Which goes through a number of compiles and then ends up with this: asterisk2:/usr/src/asterisk-1.4.4 # make menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [LD] stereorize.o frame.o - stereorize make[1]: g++: Command not found make[1]: *** [stereorize] Error 127 make: *** [utils] Error 2 Any suggestions would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM410P
Great !!! Thanks a lot !! Nitesh Divecha a écrit : Hello, Here is my config: - /etc/zaptel.conf # T1 Configuration span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 span=3,1,0,esf,b8zs bchan=49-71 dchan=72 span=4,1,0,esf,b8zs bchan=73-95 dchan=96 /etc/asterisk/zapata-channels.conf You need to #include zapata-channels.conf in your zapata.conf ; signalling = pri_cpe is USER ; signalling = pri_net is NETWORK group = 1 switchtype = national signalling = pri_net context = from-zaptel channel = 1-23 group = 2 switchtype = national signalling = pri_net context = from-zaptel channel = 25-47 group = 3 switchtype = national signalling = pri_net context = from-zaptel channel = 49-71 group = 4 switchtype = national signalling = pri_cpe context = from-zaptel channel = 73-95 I use FreePBX as my front-end to route calls... so I just assign the trunk groups which I want to use... Regards, Nitesh Alexandre VERNIOL wrote: HI all, Does some one can give me his configuration (zapta.conf, zaptel.conf, sample for extensions.conf) for TDM410P card or similar (1/2/3/4 BRI card) Thanks in advance. Cheers, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AT530 Telephone
Hi, What sort of DTMF do you use in the AT530 ? It seems that just a problem of DTMF otherwise it don't work with your softphone. Cheers, Josu Lazkano Lete a écrit : Hello everybody. I have two AT530 telephones and one X-Lite extension conected to my Asterisk. This is part of my extensions.con. exten = 105,1,Answer exten = 105,2,Background(/home/user/suport) exten = 1,1,Dial(SIP/101,30,Ttm) exten = 2,1,Dial(SIP/102,30,Ttm) When I call to 105 extension from the AT530 telephones and I select option 1 it doesn't redirect to 101 extension. Otherwise with the X-Lite extension I select 1 or 2 options and it works perfectly. Anyone has the same problem? I must push another button to redirect well? Thanks to all. Bye! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AT530 Telephone
Use this one DTMF_RFC2833 Be sure to have in your peers definition this line (sip.conf): [peer] dtmfmode=rfc2833 Cheers, Josu Lazkano Lete a écrit : I have DTMF_RELAY which do you recomend? the options are. DTMF_RELAY DTMF_RFC2833 DTMF_SIP_INFO thanks - Original Message - From: Alexandre VERNIOL [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 10, 2007 10:15 AM Subject: Re: [asterisk-users] AT530 Telephone Hi, What sort of DTMF do you use in the AT530 ? It seems that just a problem of DTMF otherwise it don't work with your softphone. Cheers, Josu Lazkano Lete a écrit : Hello everybody. I have two AT530 telephones and one X-Lite extension conected to my Asterisk. This is part of my extensions.con. exten = 105,1,Answer exten = 105,2,Background(/home/user/suport) exten = 1,1,Dial(SIP/101,30,Ttm) exten = 2,1,Dial(SIP/102,30,Ttm) When I call to 105 extension from the AT530 telephones and I select option 1 it doesn't redirect to 101 extension. Otherwise with the X-Lite extension I select 1 or 2 options and it works perfectly. Anyone has the same problem? I must push another button to redirect well? Thanks to all. Bye! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM410P
HI all, Does some one can give me his configuration (zapta.conf, zaptel.conf, sample for extensions.conf) for TDM410P card or similar (1/2/3/4 BRI card) Thanks in advance. Cheers, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] usereqphone=yes seems to don't work
Hi all, I'm looking at a function to add user=phone into sip's trame. So I include usereqphone=yes into the [general] of my sip.conf. But it seems to don't work; so is there an other way to add this user=phone through * ? Cheers, -- Alexandre VERNIOL Technicien VoIP Revendeur Directcentrex Hotline : 0892 46 05 12 Ticket : http://ticket.directcentrex.com www.directcentrex.com www.frontier.fr www.directnom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
