[asterisk-users] Problems with bridging data calls over Wildcard TE405P
Hi, I recently installed an Asterisk server at a customer's here in Germany, using a Wildcard TE405P - Ports 1 and 2 (group 1) are connected to the phone system (Deutsche Telekom), whereas ports 3 and 4 (group 2) connect to the existing HiPath 3500 PABX. Currently, the dialplan is doing nothing but routing calls through (bridging) for the time being: -- [inbound] exten = _4048.,1,Dial(Zap/g2/${EXTEN}) [outbound] exten = _0.,1,SetCIDNum(4048) exten = _0.,n,Dial(Zap/g1/${EXTEN}) -- This works great for all voice calls inbound and outbound, also faxes and modem calls are routed just fine. Data calls from ISDN devices in both directions, however (like home banking) are only connected, yet no data flows and after a couple seconds the answering side hangs up. At the console this looks like -- -- Executing SetCIDNum(Zap/119-1, 4048) in new stack -- Executing Dial(Zap/119-1, Zap/g1/0531590||tT) in new stack -- Requested transfer capability: 0x08 - DIGITAL -- Called g1/0531590 -- Zap/4-1 is ringing -- Zap/4-1 answered Zap/119-1 -- Hungup 'Zap/4-1' -- I tried to read up on everything regarding data calls, but only found so much different, sometimes conflicting, info that I'm hoping somebody here has some experience with that. Thanks Marc ___ --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] Putting a call recording into a mailbox
Hi, I was wondering, after recording a call (through either the monitor()-application or automon), is there a way to put the recorded file into a user's mailbox? So far, we just send out the file as an email attachment, but having it in my mailbox would just be so much more convenient... (^_^) Marc Rohlfing ___ --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
AW: [Asterisk-Users] Putting a call recording into a mailbox
Hi, I don't understand what you are asking: what's the difference between sending out the email as an attachment so it ends up in a user's mailbox versus having it in the user's mailbox. Aren't they the same? Oops, my bad: I'm talking about the user's *voicemail* box here - should have been more precise there... Marc Rohlfing ___ --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
AW: [Asterisk-Users] syntax error
Hi, Does anyone know why this row: exten = s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} != ${RGPREFIX}]?4:3) took me some squinting, but the parantheses seem correct - so I presume the Asterisk parser can't cope with that convoluted an expression (using a function within a variable, basically). Try putting LEN(${RGPREFIX}) into a separate variable first, then refer to it in your GotoIf in a second statement. Marc ___ --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
AW: [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06
Good morning, Why still use mpg123? Start using format_mp3 from asterisk-addons and your * will play mp3 by itself... good point - did that, and everything's working again. Thanks! Marc Rohlfing ___ --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] Eicon Diva Server with v3.0 drivers
Hi, I'm trying to get an Eicon Diva Server4BRI card running under Ubuntu 6.06 - by downloading the v3 driver package from Melware and compiling everything. Yet, after activating the necessary modules (divas and divadidd) and interactively configuring the card (/usr/lib/divas/Config), starting up the adapter fails! The error /usr/lib/divas/divactrl load -c 1 -Debug produces complains about the missing protocol image A: can't open protocol image: (/usr/lib/divas/te_etsi.2q0) I know from version 2.0 that you had to download these images from either isdn4linux.org or melware.de, yet none of them still have the files available. /usr/lib/divas/ does contain some *etsi* files - do they help somehow? Any hint appreciated. Marc Rohlfing ___ --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
AW: [Asterisk-Users] Eicon Diva Server with v3.0 drivers
Hi, The error /usr/lib/divas/divactrl load -c 1 -Debug produces complains about the missing protocol image A: can't open protocol image: (/usr/lib/divas/te_etsi.2q0) It is not necessary any more to download any firmware files (they are incompatible anyway). All needed files are part of the v3 package. It is odd, that the file te_etsi.* is searched. This is needed only if the DMLT code (te_dmlt.*) is not available. Can you please provide the list of files which are installed in /usr/lib/divas and possible logs /var/log/diva* ? As usual, the second I sent my request, I tried something else and it worked (^_^) Seriously: If I run the autogenerated startup script (/usr/lib/divas/divas_cfg.rc), the card is activated just fine. capiinfo shows all 8 B-channels, so I guess I'm good to go. Maybe this should be stated more clearly in the INSTALL and README files - it's especially confusing for veterans who try to do things the 2.1 way... In any case: Thanks for the quick reply. Marc Rohlfing ___ --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] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06
Hi, I made the mistake of upgrading both my Linux box (to Ubuntu 6.06) and Asterisk (to 1.2.9.1) at the same time. Now, when trying to compile mpg123 - using the tried and true make mpg123 -, the build fails with an error make[3]: Entering directory `/usr/src/asterisk-1.2.9.1/mpg123-0.59r' make[3]: *** No rule to make target `\ ', needed by `mpg123'. Stop. Maybe there's someone out there more versed in Linux who has an idea what might have gone wrong. Thanks! Marc ___ --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
AW: [Asterisk-Users] Free/Open pci telco card
Hi, While I was surfing the net last week, I found a link for open source pci telco cards. I'm not sure if it were isdn or analog related. The layout an all the stuff was free downloadable, so that you can build your own cards. Does anybody have the link? you're probably talking about the Zapata Telephony Project and their Tormenta-Cards: http://www.zapatatelephony.org/ Marc -- Marc Rohlfing Training Manager OASY AG Heinz-Nixdorf-Ring 1 Gebäude C 33106 Paderborn Telefon: +49 5251 / 68893 -13 Fax: +49 5251 / 68893 -09 E-Mail: [EMAIL PROTECTED] Internet: www.oasy.de ___ --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] Asterisk evaluating CLIP, then getting out of the way
Hi, I've just come upon an interesting question regarding the use of Asterisk as an Application Server connected behind a conventional ISDN PBX: The user wants to forward all incoming calls through the PBX to Asterisk over S0-Lines, have Asterisk do some processing (which includes looking up the final internal extension for the incoming call) and then returning the call to the PBX (a Siemens HiPath 3000) and have that ring the phone. The catch: In order not to tie up 2 B-channels for every call on the leg between PBX and * (thus needing a big, expensive ISDN card in the Asterisk server), we would like to have Asterisk 'drop' the call completely once it has been returned to the HiPath PBX. I've tried to dig up information on the web, but wasn't able to find out whether that is possible at all - any feedback is appreciated! Marc Rohlfing -- Marc Rohlfing Training Manager OASY AG Heinz-Nixdorf-Ring 1 Gebäude C 33106 Paderborn Telefon: +49 5251 / 68893 -13 Fax: +49 5251 / 68893 -09 E-Mail: [EMAIL PROTECTED] Internet: www.oasy.de ___ --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