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...s to SIP/YYY-000002f7)" I don't want that asterisk receive the call in local because I can't read headers in local... anyone have a solution to accept a call with the same CALL-ID in SIP channel ? thank you for all, regards, -- Alexandre Rendour Acropolis Telecom <http://www.acropolistelecom.net> Direct: +33 (0) 181813201 Support: +33 (0) 811 851 851 rendour at acropolistelecom.net <mailto:rendour at acropolistelecom.net> Adresse : 161-163 avenue Gallieni Paris - Porte de Bagnolet 93170 Bagnolet

Hello, I have conferences in my database. I need at some moments, to access the database without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I just got some interesting information from Kirk Telecom (www.kirktelecom.com). This company has been in the business of providing DECT solutions (IP gateway, base stations, repeaters and handsets) either to be used with Cisco CallManager (SCCP protocol) or with the Innovaphone IP PBX system (H.323). Two important elements: 1. It seems they foresee a SIP version of their product in Q1

Does anyone know if the 7912G (which the wiki says can do either sccp or sip) uses the 7940/60 sip firmware? I ask this because the only firmware I can seem to find on TAC for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use 7940/60 firmware, can someone point me to the right location for it? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone:

...isk-addons-1.6.2.0-rc1.tar.gz #cd asterisk-addons-1.6.2.0-rc1 #./configure && make menuselect && make && make install && make samples thank you for help Cordialement, BERGANZ Fran?ois cid:image001.gif at 01C8F7CD.6BC1D2C0 <http://www.acropolistelecom.net/> http://www.acropolistelecom.net P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090824/93866e62/attachment-0001.htm...

Hello, I need to send a tiff via fax with my asterisk 1.6.1.0. I tried in the dialplan [default] exten => _X.,1,SendFax(/root/test.tiff) but I have: salledeconf1*CLI> console dial 111 at default [Dec 23 16:24:22] WARNING[31739]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory -- Executing [111 at default:1]

Hello, I am looking for a good prepaid solution. What is the best ? Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081212/5188f96b/attachment.htm

Is it possible to allow reinvites to/from specific devices? For example; exten 2001 and 2002 can reinvite to each other, but not 2003 and 2004 exten 2003 and 2004 can reinvite to each other, but not 2001 and 2002 Can that be done? Devices 2001 & 2002 are behind one firewall, and 2003 & 2004 are behind another. Tim

Hello all, Here my architecture : Proxy1-asterisk1-proxy2-phone1 If a call arrived from proxy1 to phone1 AND phone1 always forward to proxy, asterisk1 say: -- Now forwarding SIP/phone1-0000001d to 'Local/969990349 at proxy2' (thanks to SIP/proxy2-0000001e) Why it use Local ? I just need to use as a normal call, not a local Thank you Francois -------------- next

Hello Folks, I'm trying to decide here between a few cards for connecting an Asterisk box to a single E1 channel (either PRI or R2 signaling): - Digium E100P: has been replaced by the TE110P below, but can still be had at places like digitnetworks.com for $475, and I guess there's always a place for good-olde-obsolete cards in the world as long as they work :-) - Digium TE110P:

hello, I will do a server to do a lots of conferences (MeetMe). I want to know that if I dont use a digum card, the limit of simultaneous calls is harder without a card than with a card ?if, yes, how harder is the limit? thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part --------------

...r) 14. Re: Faxing success rate on PRI (Marco Signorini) 15. Re: How to install spandsp from source in lenny ? (James Sneeringer) ---------------------------------------------------------------------- Message: 1 Date: Mon, 9 Mar 2009 15:29:04 +0100 From: BERGANZ Fran?ois <francois at acropolistelecom.net> Subject: Re: [asterisk-users] problem with an agi in PHP To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users at lists.digium.com> Message-ID: <006601c9a0c3$66331360$32993a20$@net> Content-Type: text/plain; charset="iso-8859-1&q...

Hello, I need help for that error message: ?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to? My network is: Client1-- -----------asterisk1------asterisk2 Client2-- ? With client1, I do a call ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Asterisk1 forward the call to

Hello, My network is: Client_SS7_1-- -----------asterisk1------asterisk2 Client_SS7_2-- ? I receive a fax from Client_SS7_1 ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Then, asterisk2 forward the fax to Client_SS7_2 I want that the SIP signaling go to asterisk2, But, I need that the RTP don?t go

Hello, I work on voicemail.conf and I need that ${VM_DATE} is in french! How can I do it? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090319/7c1de958/attachment.htm

Hello, I have a var like ?blabla? with the ? I need to suppr the ? Is it possible with the ${var:x:y} ? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am trying to do : Exten =>_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:10000)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:10000)' Why? I forgot something ? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que

Hello all, I have a problem when I try to install FastAGI. I try to do #perl Makefile.PEL And it return : Can't locate inc/Module/Install.pm in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at Makefile.PL line 1. BEGIN failed--compilation aborted

Hello, In my dialplan, I do s,n,DIAL( ) If my called phone response and after hangup, asterisk execute the h,1, But, if I the caller hangup at ringing (cancel), it don?t execute the h,1, Know you why? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML

Hello, I need help to use my sangoma card a108d. I need that another server give me an E1 with a clock. The server with the sangoma reseive the E1 clock on port1 and is MASTER E1 on port2. But, I cant receive the clock (I am connected). Anyone can help me? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire.