[asterisk-users] Any free video (or audio) softphone VOIP client under Linux with touchscreen friendly interface ?
Hi, I'm curious if anyone knows of any possibility to use video VOIP client (like Ekiga or Linphone or...) under Linux that could be operated by touchscreen friendly GUI (bigger buttons, large keypad, etc...) ? I like Ekiga, but GUI is small and cannot be operated via touchscreen... But maybe there are some skins for existing clients that are more touchscreen friendly ? Thanks in advance, regards, Rob. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I use X-Lite from local and external ip (when I'm not at home) ?
Hi, X-Lite demo version has only one SIP account possible. I'd like to set it up in such manner that I could register with Asterisk being at home (local LAN, local ip) and at work (external ip). Is this possible since X-Lite allows only one sip account settings ? Thanks in advance, regards, Rob. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 423 Interval Too Brief and expiry settings insip.conf
- Original Message - From: Robert Rozman [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, March 20, 2008 7:40 PM Subject: [asterisk-users] 423 Interval Too Brief and expiry settings insip.conf Hi, I'm getting this error when registering with SIP server using Asterisk 1.4.10 and Freepbx... I'm getting this error no matter what I try to setup in sip.conf : - I'm getting confused whether options are maxexpirey=36000 or maxexpiry=36000 ? - Can I solve this with some settings in sip.conf or is this problem harder ? - I've read something about Asterisk's bug on this error, but am not sure it really patching is necessary or can be avoided with different settings ? Thanks in advance, regards, Rob. --- (10 headers 0 lines) --- -- Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER [Mar 20 18:34:49] VERBOSE[7840] logger.c: --- SIP read from xxx.xxx.xxx.xxx:5060 --- SIP/2.0 423 Interval Too Brief Call-ID: [EMAIL PROTECTED] CSeq: 174 REGISTER From: sip:@ xxx.xxx.xxx.xxx;tag=as200dbc2c Min-Expires: 600 Server: Cirpack/v4.41f (gw_sip) To: sip:59972778@ xxx.xxx.xxx.xxx;tag=00-08013-1313fd60-3a4260273 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received= xxx.xxx.xxx.xxx;rport=5060;branch=z9hG4bK259bcd00 WWW-Authenticate: Digest realm=XXX.XXX,nonce=1313fbd315b1cefb52c870440e6f5455,opaque=1311d5ce56cc060,stale=false,algorithm=MD5 Content-Length: 0 - [Mar 20 18:34:49] VERBOSE[7840] logger.c: --- (10 headers 0 lines) --- [Mar 20 18:34:49] VERBOSE[7840] logger.c: -- Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx [Mar 20 18:34:49] VERBOSE[7840] logger.c: Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER Hi, I explored further and I find something weird in my registration sip packet. It says expires : 120 in any case. Does this field mean expiry for SIP registration or just some expiry for sip packet ? If it is for SIP registration, why is always 120 regardles of settings in sip.conf ? Thanks in advance, regards, Rob. REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 10.253.1.31:5060: REGISTER sip:10.253.1.31 SIP/2.0 Via: SIP/2.0/UDP 10.135.125.59:5060;branch=z9hG4bK2b4e2906;rport From: sip:[EMAIL PROTECTED];tag=as38898bbb To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 107 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=59972778, realm=voip.siol, algorithm=MD5, uri=sip:10.253.1.31, nonce=178ab9c669db2748795e5a3442b8657b, response=7035dac9ab19de49f63d45f8c162558c, opaque=178aaca821979d1 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Howto connect to Cirpack softswitch with Asterisk ?
Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Thanks in advance, regards, Rob. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Howto connect to Cirpack softswitch withAsterisk ?
- Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, April 02, 2008 10:51 AM Subject: Re: [asterisk-users] Howto connect to Cirpack softswitch withAsterisk ? On 10:11, Wed 02 Apr 08, Robert Rozman wrote: Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Yes, it works fine. Where do you get stuck ? It's basically a normal sip connection setup. Hi, thanks for response I have it registered and receiveing incoming calls, but outgoing calls don't work. I'm attaching sip log below, the basic problem is that some sort of authentication is desired on outgoing calls... Cirpack says: SIP/2.0 407 authentication required and then Cirpack says: SIP/2.0 403 Wrong login or password I'm attaching full log below.. I'd kindly ask if someone can shed some light, where to specify outgoing authentication (I use freepbx also) ? Can incoming calls be proceeded to ring local extensions without actually taking call (so ISP won't charge for just ringing) ? Thanks in advance, regards, Rob. SIP full log : Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER -- Executing [EMAIL PROTECTED]:1] Macro(SIP/202-b654e668, dialout-trunk|2|041461620||) in new stack -- Executing [EMAIL PROTECTED]:1] Set(SIP/202-b654e668, DIAL_TRUNK=2) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/202-b654e668, DIAL_NUMBER=041461620) in new stack -- Executing [EMAIL PROTECTED]:3] Set(SIP/202-b654e668, ROUTE_PASSWD=) in new stack -- Executing [EMAIL PROTECTED]:4] GotoIf(SIP/202-b654e668, 1?noauth) in new stack -- Goto (macro-dialout-trunk,s,6) -- Executing [EMAIL PROTECTED]:6] GotoIf(SIP/202-b654e668, 0?disabletrunk|1) in new stack -- Executing [EMAIL PROTECTED]:7] Set(SIP/202-b654e668, _NODEST=) in new stack -- Executing [EMAIL PROTECTED]:8] Set(SIP/202-b654e668, DIAL_TRUNK_OPTIONS=tr) in new stack -- Executing [EMAIL PROTECTED]:9] Set(SIP/202-b654e668, GROUP()=OUT_2) in new stack -- Executing [EMAIL PROTECTED]:10] Macro(SIP/202-b654e668, user-callerid|SKIPTTL) in new stack -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/202-b654e668, user-callerid: device 202) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/202-b654e668, AMPUSER=202) in new stack -- Executing [EMAIL PROTECTED]:3] GotoIf(SIP/202-b654e668, 0?report) in new stack -- Executing [EMAIL PROTECTED]:4] GotoIf(SIP/202-b654e668, 0?start) in new stack -- Executing [EMAIL PROTECTED]:5] Set(SIP/202-b654e668, REALCALLERIDNUM=202) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/202-b654e668, REALCALLERIDNUM is 202) in new stack -- Executing [EMAIL PROTECTED]:7] Set(SIP/202-b654e668, AMPUSER=202) in new stack -- Executing [EMAIL PROTECTED]:8] Set(SIP/202-b654e668, AMPUSERCIDNAME=pl_51) in new stack -- Executing [EMAIL PROTECTED]:9] GotoIf(SIP/202-b654e668, 0?report) in new stack -- Executing [EMAIL PROTECTED]:10] Set(SIP/202-b654e668, AMPUSERCID=202) in new stack -- Executing [EMAIL PROTECTED]:11] Set(SIP/202-b654e668, CALLERID(all)=pl_51 202) in new stack -- Executing [EMAIL PROTECTED]:12] Set(SIP/202-b654e668, REALCALLERIDNUM=202) in new stack -- Executing [EMAIL PROTECTED]:13] NoOp(SIP/202-b654e668, TTL: ARG1: SKIPTTL) in new stack -- Executing [EMAIL PROTECTED]:14] GotoIf(SIP/202-b654e668, 1?continue) in new stack -- Goto (macro-user-callerid,s,23) -- Executing [EMAIL PROTECTED]:23] NoOp(SIP/202-b654e668, Using CallerID pl_51 202) in new stack -- Executing [EMAIL PROTECTED]:11] Macro(SIP/202-b654e668, record-enable|202|OUT) in new stack -- Executing [EMAIL PROTECTED]:1] GotoIf(SIP/202-b654e668, 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing [EMAIL PROTECTED]:4] AGI(SIP/202-b654e668, recordingcheck|20080402-143454|1207139694.24) in new stack -- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck recordingcheck|20080402-143454|1207139694.24: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/202-b654e668, No recording needed) in new stack -- Executing [EMAIL PROTECTED]:12] GotoIf(SIP/202-b654e668, 0?skipoutcid) in new stack -- Executing [EMAIL PROTECTED]:13] Set(SIP/202-b654e668, DIAL_TRUNK_OPTIONS=) in new stack -- Executing [EMAIL PROTECTED]:14] Macro(SIP/202-b654e668, outbound-callerid|2) in new stack -- Executing [EMAIL PROTECTED]:1] GotoIf(SIP/202-b654e668, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing [EMAIL PROTECTED]:3] NoOp(SIP/202-b654e668, REALCALLERIDNUM is 202) in new stack -- Executing [EMAIL PROTECTED]:4] GotoIf(SIP/202-b654e668, 1?normcid) in new stack -- Goto (macro-outbound-callerid,s,9) -- Executing [EMAIL PROTECTED]:9] Set(SIP/202
[asterisk-users] 423 Interval Too Brief and expiry settings in sip.conf
Hi, I'm getting this error when registering with SIP server using Asterisk 1.4.10 and Freepbx... I'm getting this error no matter what I try to setup in sip.conf : - I'm getting confused whether options are maxexpirey=36000 or maxexpiry=36000 ? - Can I solve this with some settings in sip.conf or is this problem harder ? - I've read something about Asterisk's bug on this error, but am not sure it really patching is necessary or can be avoided with different settings ? Thanks in advance, regards, Rob. --- (10 headers 0 lines) --- -- Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER [Mar 20 18:34:49] VERBOSE[7840] logger.c: --- SIP read from xxx.xxx.xxx.xxx:5060 --- SIP/2.0 423 Interval Too Brief Call-ID: [EMAIL PROTECTED] CSeq: 174 REGISTER From: sip:@ xxx.xxx.xxx.xxx;tag=as200dbc2c Min-Expires: 600 Server: Cirpack/v4.41f (gw_sip) To: sip:59972778@ xxx.xxx.xxx.xxx;tag=00-08013-1313fd60-3a4260273 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received= xxx.xxx.xxx.xxx;rport=5060;branch=z9hG4bK259bcd00 WWW-Authenticate: Digest realm=XXX.XXX,nonce=1313fbd315b1cefb52c870440e6f5455,opaque=1311d5ce56cc060,stale=false,algorithm=MD5 Content-Length: 0 - [Mar 20 18:34:49] VERBOSE[7840] logger.c: --- (10 headers 0 lines) --- [Mar 20 18:34:49] VERBOSE[7840] logger.c: -- Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx [Mar 20 18:34:49] VERBOSE[7840] logger.c: Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using dedicated eth2 card for SIP trunk line to ISP provider - how to setup ?
Hi, I'm about to test VOIP connection (from my ISP provider) directly through dedicated network card instead of going through ADSL gateway with analog phone port - SPA 3000 - Asterisk. I need to have eth2 set on dhcp (to retrieve IP automatically) and then work with it under Asterisk as dedicated VOIP trunk. Anyone with more insight how to setup such situation ? Any more info anywhere ? Thanks in advance, regards, Bulek. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?
Hi, I have an older phone with touch screen from Philips. It have it connected to Sipura 3000 FXS port and majority of features work ok. But phone also has touchscreen and web browser that I'd love to use for accessing my local web pages. But the phone only allows me to setup ISP phone number (username and password) and it wants to call it to get to Internet. Since it is connected to Sipura3000, call can come to Asterisk and I'd love to somehow fool that device and connect it to local web pages ? I guess I could somehow mimic ISP internet calling feature on local Asterisk server, but have no clue even where to start searching ... Any advice ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I connect device on FXS of Sipura 3000 to internet virtually ? - it can only call ISPs numbers on POTS line
Hi, I have an older phone with touch screen from Philips. It have ti connected to Sipura 3000 FXS port and majority of features work ok. But phone also has touchscreen and web browser that I'd love to use for accessing my local web pages. But the phone only allows me to setup ISP phone number and it wants to call it to get to Internet. Since it is connected to Sipura3000, call can come to Asterisk and I'd love to somehow fool that device and connect it to local web pages ? I guess I could somehow mimic ISP internet calling feature on local Asterisk server, but have no clue even where to start searching ... Any advice ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bristuff vs. vISDN vs. mISDN for hfc card ?
Hi, some time ago we used bristuffed Asterisk for our hfc cards cause it offered more features (echo cancellation most important) and was quite stable... I'm seeing now (I'm putting together Asterisk after a long time with hfc card) that there are now 3 choices for hfc chipsets : vISDN, mISDN and bristuff. What are pros and cons of each of them and what do you put in your Asterisks ? How they differ in functionality ? Thanks in advance, regards, Rob. ___ --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] How to stream audio to external app for speech recognition and recognize dtmf in parallel ?
Hi, we're writting interface module for our speech recognition system. We would like to export stream of audio samples to external app, but to preserve dtmf recognition and dialplan progress. I wonder if recording application would be a good start for that (recording application obviously streams audio and makes recording out of it in parallel) We're also interested in best way to report speech recognition results back Best way would be to be able to call extension in dialplan, for instance : 1, DTMF 1 2, DTMF 2 support, spoken word support sales, spoken word sales -- but also putting results in variable would be probably fine Any advice how to develope such scenario ? What is the best module code to start with ? Any similar solutions ? Thanks in advance, regards, Rob. ___ --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] Bristuffed asterisk 1.2.10 on Suse 10 - problemwith module versionmagic
- Original Message - From: Paul Hewlett [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, September 17, 2006 3:34 PM Subject: Re: [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problemwith module versionmagic On Saturday 16 September 2006 20:35, Tzafrir Cohen wrote: On Fri, Sep 15, 2006 at 02:58:02PM +0200, Robert Rozman wrote: I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64 x86_64 x86_64 GNU/Linux and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I get this : laps1:~/Voipy/Bristuff/bristuff-0.3.0-PRE-1s/zaptel # modprobe zaphfc FATAL: Error inserting zaphfc (/lib/modules/2.6.13-15.11-smp/misc/zaphfc.ko): Invalid module format and this in dmesg : zaphfc: version magic '2.6.13-15.11-smp gcc-4.0' should be '2.6.13-15.11-smp SMP gcc-4.0' This means you built zaptel with the wrong kernel headers. Is there a SUSE power user in the crowd? From memory as I do not use SUSE anymore.. Suse already has the zaptel modules in its kernel under the (IIRC) the extra directory (instead of misc) . you end up with 2 sets of zaptel modules in the linux module tree and modprobe then gets confused as to which to load. Look for a directory /lib/modules/2.6.13-15.11-smp/extra You must (again from memory) delete the old modules abd rerun depmod or rebuild your asterisk Thanks for the hint. I already did that. To me it seems that something is messed up with settings, so zaphfc compiles with slightly different version magic - note that the problem lies only in 'SMP' : '2.6.13-15.11-smp gcc-4.0' vs '2.6.13-15.11-smp SMP gcc-4.0' That's weird for me Any further help ? Thanks in advance, regards, Rob. ___ --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] Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic
I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel : Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64 x86_64 x86_64 GNU/Linux and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I get this : laps1:~/Voipy/Bristuff/bristuff-0.3.0-PRE-1s/zaptel # modprobe zaphfc FATAL: Error inserting zaphfc (/lib/modules/2.6.13-15.11-smp/misc/zaphfc.ko): Invalid module format and this in dmesg : zaphfc: version magic '2.6.13-15.11-smp gcc-4.0' should be '2.6.13-15.11-smp SMP gcc-4.0' What am I doing wrong? Anyone sucessfully using latest Bristuff under Suse ? Thanks in advance, regards, Rob. ___ --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] I am looking for a webphone on MY SITE
- Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 26, 2006 2:41 PM Subject: [Asterisk-Users] I am looking for a webphone on MY SITE I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? Aware of this one ? http://www.hem.za.org/jiaxclient/ HTH, Regards, ___ --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] Jingle support - can we test the feature ?
- Original Message - From: Tim Panton [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 21, 2006 11:43 AM Subject: Re: [Asterisk-Users] Jingle support - can we test the feature ? On 20 Apr 2006, at 16:39, Robert Rozman wrote: - Original Message - From: Time Bandit [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 20, 2006 4:18 PM Subject: Re: [Asterisk-Users] Jingle support - can we test the feature ? we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Have you looked at Wildfire ? http://www.jivesoftware.org/wildfire/ There is an Asterisk-plugin that update your status automagically when you're on the phone -- Hi, thanks for pointer. I know for that project, but reading about Jingle, Jabber and Asterisk integration it seems not so interesting for me at the moment... Regards, So what aspect of Jingle, Jabber and Asterisk did you mean in your original post ? Well I've read few general interviews and articles about integration of Jingle protocol and Asterisk. There is also IAX version of specification for audio transport. There is asterisk-xmpp effort. The main thing at least in my opinion would be that I could have network of Asterisk servers, and user could use integrated client that would give presence, IM and audio communication in Asterisk compatible way.. So I'm curious if anyone has made and tests or has more info on that Regards, Rob. ___ --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] Jingle support - can we test the feature ?
Hi, we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Thanks in advance, regards, Rob. ___ --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] Dial two extensions at the SAME time and connect them when possible
Hi, I want to start call between A and B. Currently call can be triggerred with either first calling A or B number and then the other number after fist picks up. I'd like to call A and B at the same time and connect them in call when possible... One way would probably be with putting both calls in conference, but maybe there is some more elegant way of doing it? Also is there any specific reason why calls are triggered so only one client is called and then the other ? Thanks , regards, Rob. ___ --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] Jingle support - can we test the feature ?
- Original Message - From: Time Bandit [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 20, 2006 4:18 PM Subject: Re: [Asterisk-Users] Jingle support - can we test the feature ? we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Have you looked at Wildfire ? http://www.jivesoftware.org/wildfire/ There is an Asterisk-plugin that update your status automagically when you're on the phone -- Hi, thanks for pointer. I know for that project, but reading about Jingle, Jabber and Asterisk integration it seems not so interesting for me at the moment... Regards, Rob. ___ --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] most common VOIP echo simulaton for research purposes ?
Hi, I'm speech recognition researcher and would like to do some research on recognition robustness in echo distortion of speech signal. Since VOIP is becoming wide spread, I'd like to simulate (one or more) common echo distortions that mostly appear in voip communications ? Any example, FIR or IIR filter or acoustical system response ? Any other distortion worth researching ? Thanks in advance, regards, Rob. ___ --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] Reading sound in eagi script and recognizing DTMF sounds at thesame time ?
Hi, we've connected Sphinx4 through eagi script (modified eagi example) to Asterisk. Users can now say their wishes - but for gradual evolution we would like to provide older way of DTMF navigation too - can we recognize DTMF while reading sound in eagi ? Any advice or examples ? Thanks in advance, regards, Rob. ___ --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] What SW/HW phones support sendtext feature (trying to send speech recognition results back to user)?
Hi, we've proof of conecpt system for speech recognition on Asterisk. We would like to send results of recognition back to user in standard way. Currently we're considering using sendtext command and it works with Firefly. But I'm curious what soft or hard ip phones that can connect to Asterisk support such feature ? Also what softphone would be most suitable for further work in adding such feature and possibly something more in this field ? Any other good way to send results (text) back to user ? Thanks in advance, regards, Rob. ___ --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] Bristuffed asterisk 1.2.1 on Suse 10 - problem with zaphfc module
Hi, I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel : Linux laps1 2.6.13-15.7-smp #1 SMP Tue Nov 29 14:32:29 UTC 2005 x86_64 x86_64 x86_64 GNU/Linux and Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f . I get this : laps1:~/Voipy/1.2.1/bristuff-0.3.0-PRE-1f/zaphfc # make load make -C /usr/src/linux-2.6 SUBDIRS=/root/Voipy/1.2.1/bristuff-0.3.0-PRE-1f/zaphfc ZAP=-I/root/Voipy/1.2.1/bristuff-0.3.0-PRE-1f/zaptel-1.2.1 modules make[1]: Entering directory `/usr/src/linux-2.6.13-15.7' Building modules, stage 2. MODPOST make[1]: Leaving directory `/usr/src/linux-2.6.13-15.7' modprobe zaptel insmod ./zaphfc.ko insmod: error inserting './zaphfc.ko': -1 Invalid module format make: *** [loadlinux26] Error 1 and this in dmesg : zaphfc: version magic '2.6.13-15.7-smp gcc-4.0' should be '2.6.13-15.7-smp SMP gcc-4.0' What am I doing wrong? Anyone sucessfully using latest Bristuff under Suse ? Thanks in advance, regards, Rob. ___ --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] Reading sound and recognizing DTMF sounds in eagi script ?
Hi, we've connected Sphinx4 through eagi script (modified eagi example) to Asterisk. Users can now say their wishes - but for gradual evolution we would like also to provide older way of DTMF navigation too - can we recognize DTMF while reading sound in eagi ? Any advice or examples ? Thanks in advance, regards, Rob. ___ --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] Can I call another S0 bus device (BRI) locally without taking 2 channels through Telco provider ?
Hi, I have Asterisk connected to BRI interface in parallel to my ordinary ISDN phone. Can I make internal calls between those two without going through telco provider and taking both voice channels ? Thanks in advance, regards, Rob. ___ --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] Sipura 3000 Disconnect Singnel
Hi, I had the same problem... I've solved it by recording desconnect tone line is sending and then do frequency analysis and then you can specify custom disconnect tone on sipura 3000 configuration Procedure is described in more details on voxilla web page.. HTH, regards, Rob. - Original Message - From: Code Lover [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, December 04, 2005 10:25 AM Subject: [Asterisk-Users] Sipura 3000 Disconnect Singnel Hi all, I was testing the FXO system from sipura 3000 with asterisk PERL AGI. But when we hangup the FXO phone the channel is not disconnecting and the destination is continue ringing. even if we try to press the disconnect button for destionations after some seconds again it start to ringing. Is it problem from Sipura itself or i have to do some advance in asterisk configuration. You suggestion will be high appricated. -- Thank You, Code Lover ___ --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] IP GSM Gateway is giving uncomplete SIP signalization to PRI interface - can I somehow avoid that in Asterisk ?
Hi, I have following setup : PBX - Voxip from Parlay -PRI- Asterisk -SIP- SIP IP GSM Gateway (2n) on outgoing call from pbx through Voxip and to IP GSM gateway : latter only responds with SIP session progress but no SIP Ringing message when connection starts to ring, so Voxip is hanging up line on approx 13sec timeout I know we could try simulate ringing with r in dial, but that would be quite wrong, cause GSM gateways sometime take more time to establish connection, so user gets false ringing signal... Can we somehow interfere with Asterisk and generate SIP messages to fool Voxip from hanging up the line ? Thanks in advance, regards, Rob. ___ --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] Call progress from sip gsm gateway to pri interface - doesn't get through
Hi, we have following setup : PBX - Parlay -ISDN PRI- Asterisk -SIP- GSM Gateway Call comes from PBX through Parlay to Asterisk and it routes it over SIP to GSM gateway. GSM gateway gives back call progress (it takes some time to ring or get through), but this info won't get back to Parlay on ISDN PRI interface (Digium PRI card), so Parlay after some timeout disconnects call We guess that this setup should work, but we're not sure. Anyone with working setup like this? Anyone with experience of call progress getting from SIP to PRI or BRI interfaces ? Any advice or pointer to more info ? Thanks in advance, regards, Rob. ___ --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] Anyone using Parlay VoXip SIP Gateway with Asterisk ?
Hi, we're having quite some problems with new hardware we're testing - Parlay Voxip ISDN-SIP gateway... So we're curious if anyone is using this in connection to Asterisk and what are experiences on this HW ? Thanks in advance, regards, Rob. ___ --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 1.2 - Windows Messenger ?
Hi, I've found quite some docs on this, but many of them deprecated... I'm curious what is the latest window messenger version that works as registered client to Asterisk... I've tried 4.7, but it registers only if I leave password empty. Am I missing something or is there any better way to register and use Windows messenger with Asterisk ? Any other sucessful experience with Windows Messenger and Asterisk ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura doesn't get caller id and hangup with Siemens Combiset
Hi, I'm trying to setup Sipura to work with Siemens Combiset 1009 on PSTN line (GSM gateway - produced for germany).. I have two problems: - Sipura doesn't detect Caller ID - Sipura doesn't detect hangup condition I have 3.1.7(GWg) firmware on Sipura and Asterisk 1.0.9... Anyone has settings for German PSTN lines or any other similar that should work ? Any other advice or help ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] groupware + unified messagerie +Asterisk
Hi, I guess you know this project, but just in case: http://jivesoftware.org/asterisk-im/ IMHO, Egroupware would be best groupware solution to start on, but they have little interest in doing that (searching their mailing list for voip returned 2 hits...). We will gradually start working on merging java sip client with Asterisk-IM client and see what will come out Regards, Rob. - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 10, 2005 5:25 AM Subject: Re: [Asterisk-Users] groupware + unified messagerie +Asterisk harry gaillac wrote: it's no what i expect the easier solution you provide the more customers you get ! Indeed. However, I tend to be of the opinion that you should have enough money in the bank for a full year of wages for someone if you take on extra staff. While this may make my growth slower, at least I can honestly guarantee my staff's continued employment! So, to cut a long story short, I don't have enough staff to write an infinitely configurable one, as I currently have my books pretty crammed with jobs. If you have any questions though and want to develop one yourself, I'm more than happy to help you! :D -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do you handle situation with Grandstream occasionally losing registration with Asterisk ?
Hi, I have Grandstream 100 as only ever present extension for my Asterisk AMP home setup. Incoming call comes to ring group and then proceeds to voicemail. But Grandstream 100 occasionally loses registration (have anyone found any solution to this ?) and then AMP's dialparties.agi won't proceed call even to voicemail, but will end it imediately (cause dialparties checks if SIP extension is present - otherwise it deletes it from call group...).. How do you handle this problem ? Thanks in advance., regards, Rob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection
Tole spada v DTMF zgodbo... - Original Message - From: Ryan [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 23, 2005 6:35 AM Subject: Re: [Asterisk-Users] DTMF detection On Sat, Oct 22, 2005 at 12:53:39PM -0600, Ryan exclaimed: snip I just setup telasip and I'm having the same issue. I captured some RTP packets and realized that when I get duplicate numbers it is because an RTP packet has arrived out of order. In all my test cases it was just one packet coming 1 packet too late, but the sequence number was correct. It seems that * instead of putting the packets back in order (using the seq numbers) makes a duplicate digit. I'm not sure if this is a bug or not (I haven't read the rfc). I found this in mantis at: http://bugs.digium.com/view.php?id=4659 Unfortunately this will require upstream providers to patch asterisk before this will work (which will happen over time). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection
Sorry, went on wrong address Regards, Rob. - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 9:22 AM Subject: Re: [Asterisk-Users] DTMF detection Tole spada v DTMF zgodbo... - Original Message - From: Ryan [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 23, 2005 6:35 AM Subject: Re: [Asterisk-Users] DTMF detection On Sat, Oct 22, 2005 at 12:53:39PM -0600, Ryan exclaimed: snip I just setup telasip and I'm having the same issue. I captured some RTP packets and realized that when I get duplicate numbers it is because an RTP packet has arrived out of order. In all my test cases it was just one packet coming 1 packet too late, but the sequence number was correct. It seems that * instead of putting the packets back in order (using the seq numbers) makes a duplicate digit. I'm not sure if this is a bug or not (I haven't read the rfc). I found this in mantis at: http://bugs.digium.com/view.php?id=4659 Unfortunately this will require upstream providers to patch asterisk before this will work (which will happen over time). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Overlap dial and match as you go = how to implement early dial on telco line
Hi, I have Asterisk between PBX and telco line. PBX is reporting number in overlap dial manner. I'd like to early connect to telco line as soon as I get for instance two numbers, that distinguish telco calls. But the problem is if I receive 3 numbers at once, then two numbers dialplan rule will not be matched I've found some references to similar problems, but I'm not sure which solution was included in Asterisk (if any) So I'd kindly ask if anyone has working solution or has idea how to do this on recent Asterisk to describe it... Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2beta and te411p: incorrectly reporting sometimes all channels busy
Hi, we have strange problem on our new card. Sometimes it reports all channels busy, so call cannot be made (it doesn't even appear in log). We've contacted Digium support, but received no usable answer (they've told us that this card should work on stable Asterisk version - AFAIK this is not correct)... Any advice, what to check and what are possible cause of such behaviour ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone using Java SIP communicator with Asterisk ?
Hi, this java video softphone claims it can operate with Windows messenger. It's also mentioned on this web page http://www.voip-info.org/wiki/view/SIP+COMMUNICATOR But I couldn't find any more info on how to set it up with Asterisk and how compatible is with other video softphones... Anyone with such experience or working installation ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E400P vs te410p vs te411p
Hi, I found E400P quad PRI card quite cheap (749USD): http://www.govarion.com/product_info.php?cPath=1products_id=2osCsid=68cdd6e3d08754 in comparison to te410p (approx 1500 USD ) http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TE410P Now newer generation with HW echo canceling emerged (te411p). I'm not sure in what things those two cards differ and what would be best option to buy (I believe there is big performance gap between them, but don't know how big and if it's worth of money) Also how do you find HW echo canceling in te411p ? Any advice, help ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firefly 3rd party - it hangs on Initialising and exits with error
Hi, I spot weird behaviour of latest Firefly 3rd party on my laptop. Sometimes it comes to state that it won't start (hangs on Initializing ) and it again works after system restart... Didn't yet figured out how to recreate it. Any similar experience ? Also - how can I force Firefly to make outgoing calls (also sip or iax calls) through Asterisk ? I'd like to make outgoing iax calls through Asterisk or other registered pbx so I can correct caller id, register outgoing call and other things Any advice ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to properly handle incoming SIP and IAX calls, so user can call back and how to properly make outgoing sip/iax calls through Asterisk ?
Hi, I'm aware that incoming and outgoing calls are going fine when isdn channels are involved - caller id properly identifies calling party, so user can call back But how to properly handle this for iax, sip calls I have few questions : - BTW, what to type for instance in remote firefly to make standalone calls to Asterisk default context or particular extension ? - If I receive incoming sip or iax call and is then saved as for instance in Firefly. Now Firefly would like to call back that caller, but call goes not through Asterisk... Why ? How to do this properly? - Outogoing calls: how to properly send outgoind iax or sip calls through asterisk, so each calling extension gets proper caller id, so can be called back ? Any experience or existing solution to this problem? Any advice ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Closest dialplan language equivalent for dialparties.agi ?
Hi, I'm using AMP and its dialparties.agi as most important script in system. I'd like to port configuration to more embedded system, where I don't have Perl available. So I'd like to implement dialparties.agi functionality as closest as possible with dialplan language. Are there any existing dialplan scripts-examples that are close related to dialparties.agi functionality ? Is it possible to use compiled Perl AGI script in binary form also as AGI script ? How to ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech Recognition
Hi, I'm not sure if DTMF is convenient solution for user that has cellular on his ear Regards, Rob. - Original Message - From: Dean Collins [EMAIL PROTECTED] To: Ed Greenberg [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 08, 2005 4:15 PM Subject: RE: [Asterisk-Users] Speech Recognition Ed can I ask you a question, Not trying to influence you one way or the other but why deal with the 'issues' of speech recognition when what you are looking to achieve is easily met with dtmf codes. Dtmf, works, is easy to manage and well established. Speech should only be used when you need to enter complex controls with more than '9' easy options etc. Just a thought. Cheers, Dean -Original Message- From: Ed Greenberg [mailto:[EMAIL PROTECTED] Sent: Friday, 8 July 2005 9:32 AM To: Dean Collins; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Speech Recognition Tell me probably is excessive. I just really need to recognize Yes, No, One, Two, Three and Four. The Sphinx suggestion should help though. /edg --On Friday, July 08, 2005 8:27 AM -0400 Dean Collins [EMAIL PROTECTED] wrote: Hi Ed, Did you read the wiki comment on Tellme? Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech Recognition
- Original Message - From: Richard Koch [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, July 08, 2005 4:38 PM Subject: [Asterisk-Users] Speech Recognition Ed, Check this out: http://turnkey-solution.com/asterisk-sphinx.html That got me up in running in no time. -Rick What are you experiences with recognition accuracy and user acceptance ? Any more info you're willing to share will help out others Regards, Rob. -Original Message- From: Ed Greenberg [mailto:edg at greenberg.org] Sent: Friday, 8 July 2005 9:32 AM To: Dean Collins; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Speech Recognition Tell me probably is excessive. I just really need to recognize Yes, No, One, Two, Three and Four. The Sphinx suggestion should help though. /edg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *-Euroisdn Italy
- Original Message - From: Emanuele Pucciarelli [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 24, 2005 11:12 PM Subject: Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *-Euroisdn Italy Robert Rozman wrote: I wanted to do this (it's principle I always follow) , but we even haven't received offer to pay for the stuff (we applied twice for offer of two cards), so bought where we actually could buy something... A customer of mine has had the same problem with the Italian dealer: they behaved as though they didn't want to sell :( I had this experience with original company No answer for 14 days... So I got a little precausious, how would SW-drivers support look like, if someone even doesn't want to sell HW... Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *- Euroisdn Italy
- Original Message - From: Emanuele Pucciarelli [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 24, 2005 5:32 PM Subject: Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *- Euroisdn Italy Robert Rozman wrote: I'm pulling my hair down and getting bold :-) . I have Asterisk between Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff Asterisk) (hint: spend the extra $$ and support who's written the software!) Hi, I wanted to do this (it's principle I always follow) , but we even haven't received offer to pay for the stuff (we applied twice for offer of two cards), so bought where we actually could buy something... What is your experience of authors of software ( I guess we all know whom we talk about) Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy
Hi, I'm pulling my hair down and getting bold :-) . I have Asterisk between Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff Asterisk) I'm trying to do just plain transfer of call from pbx to ISDN through Asterisk... It seems like PBX hangsup, when call is progressing with no apparent reason. I'd kindly ask for any advice or some working example for this On isdn side I also have a problem. Asterisk quite often says that it cannot create ZAP channel, although partticular span is reported up and active. I've also tried to connect loop between NT and TE port and call doesn't get through I'd really appreciate if anyone has any advice on this problem, or any experience or working example for italian ISDN and particular Panasonic PBX. Thanks in advance, regards, Rob. I'm getting this : Jun 22 16:25:13 VERBOSE[5536]: -- Accepting overlap voice call from '432575513' to '000' on channel 0/2, span 4 Jun 22 16:25:21 VERBOSE[5536]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mZap/11-1[0;37;40m, [1;35;40mZAP/g1/38670613063|60[0;37;40m) in new stack Jun 22 16:25:21 VERBOSE[5536]: -- Called g1/38670613063 Jun 22 16:25:32 DEBUG[5536]: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/2 span 1 Jun 22 16:25:32 VERBOSE[5536]: -- Zap/2-1 is making progress passing it to Zap/11-1 Jun 22 16:25:32 DEBUG[5536]: Received AST_CONTROL_PROGRESS on Zap/11-1 Jun 22 16:25:32 DEBUG[5536]: Dunno what to do with control type 15 Jun 22 16:25:34 VERBOSE[5536]: -- Channel 0/2, span 4 got hangup Jun 22 16:25:34 DEBUG[5536]: Set option AUDIO MODE, value: ON(1) on Zap/2-1 Jun 22 16:25:34 DEBUG[5536]: Hangup: channel: 2 index = 0, normal = 33, callwait = -1, thirdcall = -1 Jun 22 16:25:34 DEBUG[5536]: Not yet hungup... Calling hangup once with icause, and clearing call Jun 22 16:25:34 DEBUG[5536]: disabled echo cancellation on channel 2 Jun 22 16:25:34 DEBUG[5536]: Set option TDD MODE, value: OFF(0) on Zap/2-1 Jun 22 16:25:34 DEBUG[5536]: Updated conferencing on 2, with 0 conference users Jun 22 16:25:34 DEBUG[5536]: Set option AUDIO MODE, value: OFF(0) on Zap/2-1 Jun 22 16:25:34 DEBUG[5536]: disabled echo cancellation on channel 2 Jun 22 16:25:34 VERBOSE[5536]: -- Hungup 'Zap/2-1' Jun 22 16:25:34 DEBUG[5536]: Exiting with DIALSTATUS=CANCEL. I have zapata.conf: [channels] switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres=yes callerid=asreceived overlapdial=yes usecallingpres=yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 ;--- ; p2p TE mode (for connecting ISDN lines in point-to-point mode) signalling = bri_cpe ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) ;signalling = bri_cpe_ptmp context=from-isdn group = 1 ; S/T port 1-4 (first quadBRI, or lower ports of an octoBRI) channel = 1-2 channel = 4-5 channel = 7-8 ;channel = 10-11 ;--- ; p2p NT mode (for connecting an ISDN PBX in point-to-point mode) ;signalling = bri_net ; p2p NT mode (for connecting an ISDN PBX in point-to-multipoint mode) signalling = bri_net_ptmp context=from-pbx group = 2 ;overlapdial=no ; S/T port 5-8 (second quadBRI, or upper ports of an octoBRI) channel = 10-11 ;channel = 13-14 ;channel = 16-17 ;channel = 19-20 ;channel = 22-23 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?
Hi, I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, errors (duplicates) on more), when transferred inband from gsm gateway to NT port of quadbri under bristuffed Asterisk. Since Asterisk is claimed to have good dtmf recognizer, I suspect there are some settings to workarouned... I've tried dtmf relax, but didn't help, so I suspect gain settings Is there any other possible cause of unreliable dtmf inband recognition ? Where can I set gain on voice channel (I guess majority of settings under bristuff in zaptel.conf are dummy) ? Any other advice on this problem or similar experience ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to dimension Asterisk - that is used solely as callback server - only sending untranscoded voice between two ISDN channels on PRI ?
Hi, I wonder how I could dimension Asterisk system that will be used solely as callback server : - when user calls it registers ring, hangup and calls back - it gives him a dial signal and calls dialed number on another ISDN channel out that means plain transfer between two ISDN channels - no transcoding or any other stuff... I guess using Asterisk in this way I could dimension for higher number of parallel calls - but how many ? General rule is to put 1 octo PRI card per PC, but could I add another one or more if used in described way ? Any similar examples of dimensioning ? Do I get any better with cluster for such purpose ? Any other advice ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - whatsettings work ?
Hi, thanks for response I have following in zapata.conf, so I guess point to multipoint setting is right ? Is framing and coding (ami,ccs) right for Italy ? Thanks in advance, regards, Rob. zapata.conf: [channels] switchtype = euroisdn ;pridialplan = dynamic je delalo pridialplan = unknown ;prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres=yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 callerid=asreceived overlapdial=yes ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp context=from-isdn group = 1 ; S/T port 1-3 (first quadBRI, or lower ports of an octoBRI) channel = 1-2 channel = 4-5 channel = 7-8 ;--- - Original Message - From: Matteo Brancaleoni [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 09, 2005 10:45 AM Subject: Re: [Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - whatsettings work ? You're connected to a p2mp bri, switch to bri_cpe_p2mp Matteo. Il giorno mer, 08-06-2005 alle 19:54 +0200, Robert Rozman ha scritto: Hi, I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy with octobri card from Beronet. I use bristuff and have following zaptel.conf... # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of 1. For a secondary, use 2, and so on. # To not use this as a sync source, just use 0 # loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,0,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 I get this on bri intense debug... Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=64864 [ fc ff 03 0f fd 60 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=39384 [ fc ff 03 0f 99 d8 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=38343 [ fc ff 03 0f 95 c7 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Thanks very much in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - what settings work ?
Hi, I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy with octobri card from Beronet. I use bristuff and have following zaptel.conf... # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of 1. For a secondary, use 2, and so on. # To not use this as a sync source, just use 0 # loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,0,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 I get this on bri intense debug... Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=64864 [ fc ff 03 0f fd 60 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=39384 [ fc ff 03 0f 99 d8 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=38343 [ fc ff 03 0f 95 c7 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Thanks very much in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disa - how it returns on user not dialing anynumbers ?
- Original Message -
From: Peter Svensson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, June 06, 2005 9:30 AM
Subject: Re: [Asterisk-Users] Disa - how it returns on user not dialing
anynumbers ?
On Mon, 6 Jun 2005, Robert Rozman wrote:
I'd like to use DISA properly for my case - I'd like to handle it right,
if
user when in DISA doesn't dial any number - how does Asterisk return from
DISA cmd ?
The file app_disa.c is hardwired to hang up the call if too many incorrect
passwords are attempted and when no valid extensions has been entered
before the digittimeout expires.
To change it the block under the reorder: label in app_disa.c is
probably the easiest. Instead of playing tones and all that it could set a
channel variable based on the k variable which seem to be the main state
variable. The dialplan could then handle the various exit cases.
Peter
Hi,
thanks for info... I wanted to add n+101 behaviour if disa doesn't get
proper password or extension
I'm total newbie and would like to ask some more experienced users whether
following change is proper to do that.
I've changed in that block :
reorder:
/*ast_indicate(chan,AST_CONTROL_CONGESTION);*/
/* something is invalid, give em reorder for several seconds */
/*time(rstart);
while(time(NULL) rstart + 10)
{
if (ast_waitfor(chan, -1) 0)
break;
f = ast_read(chan);
if (!f)
break;
ast_frfree(f);
}
ast_playtones_stop(chan);*/
LOCAL_USER_REMOVE(u);
/* return -1;*/
/* Dodano timeout koda:*/
if (ast_exists_extension(chan, chan-context, chan-exten, chan-priority +
101, chan-callerid))
chan-priority+=100;
return 0;
}
Thanks in advance,
regards,
Rob.
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[Asterisk-Users] Disa - how it returns on user not dialing any numbers ?
Hi, I'd like to use DISA properly for my case - I'd like to handle it right, if user when in DISA doesn't dial any number - how does Asterisk return from DISA cmd ? I'd like to dial some default number if user doesn't dial anything or give him some message - but I don't know what gets executed after DISA if nothing is dialed I'm reading this on wiki, but don't understand what following sentence means. Any help ? ... If login is successful, the application parses the dialed number in the specified (or default) context, and returns 0 with the new extension context filled-in and the priority set to 1, so that the PBX may re-apply the routing tables to it and complete the call normally. ... Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disa - how it returns on user not dialing anynumbers ?
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, June 06, 2005 1:27 AM
Subject: Re: [Asterisk-Users] Disa - how it returns on user not dialing
anynumbers ?
Try the t extension (stands for timeout).
Hi, thanks for advice. But it seems that t extension doesn't get executed. I
have it in context from where I call DISA and also in DISA context, but no
go
After dialing nothing and some time it just gets hungup, no t extension
triggering .
Any advice ?
Thanks in advance,
regards,
Rob.
I have :
exten = s,14,DigitTimeout(5)
exten = s,15,ResponseTimeout(20)
exten = s,16,DBget(temp=USER/${CALLERIDNUM})
exten = s,17,SetAccount(${temp})
exten = s,18,DISA(no-password|from-bri-user-disa)
exten = t,1,DBget(temp=DYNAMIC/${CALLERIDNUM})
exten = t,2,DBdel(DYNAMIC/${CALLERIDNUM})
exten = t,3,Dial(Local/[EMAIL PROTECTED]/n)
exten = t,102,Goto(from-pstn,s,1) ;
[from-bri-user-disa]
include = outbound-allroutes-custom
include = outrt-001-Mednarodni
include = ext-local
include = outrt-003-zunanja
exten = s,1,NoOp(DISA: Waiting for BRI user to enter number)
exten = s,2,DigitTimeout(5) ; Vhodni
exten = s,3,ResponseTimeout(20) ;
exten = t,1,DBget(temp=DYNAMIC/${CALLERIDNUM})
exten = t,2,DBdel(DYNAMIC/${CALLERIDNUM})
exten = t,3,Dial(Local/[EMAIL PROTECTED]/n)
exten = t,102,Goto(from-pstn,s,1) ;
On 6/5/05, Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
I'd like to use DISA properly for my case - I'd like to handle it right,
if
user when in DISA doesn't dial any number - how does Asterisk return from
DISA cmd ?
I'd like to dial some default number if user doesn't dial anything or give
him some message - but I don't know what gets executed after DISA if
nothing
is dialed
I'm reading this on wiki, but don't understand what following sentence
means. Any help ?
... If login is successful, the application parses the dialed number in
the specified (or default) context, and returns 0 with the new extension
context filled-in and the priority set to 1, so that the PBX may
re-apply the routing tables to it and complete the call normally. ...
Thanks in advance,
regards,
Rob.
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Re: [Asterisk-Users] Disa - how it returns on user not dialinganynumbers ?
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, June 06, 2005 2:40 AM
Subject: Re: [Asterisk-Users] Disa - how it returns on user not
dialinganynumbers ?
What is the CLI output?
Hi,
I get this:
-- Executing SetAccount(Zap/10-1, 250) in new stack
-- Executing DigitTimeout(Zap/10-1, 5) in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout(Zap/10-1, 20) in new stack
-- Set Response Timeout to 20
-- Executing DBget(Zap/10-1, temp=USER/041461620) in new stack
-- DBget: varname=temp, family=USER, key=041461620
-- DBget: set variable temp to robi
-- Executing SetAccount(Zap/10-1, robi) in new stack
-- Executing DISA(Zap/10-1, no-password|from-bri-user-disa) in new
stack
== Spawn extension (from-bri-user, s, 18) exited non-zero on 'Zap/10-1'
-- Executing NoOp(Zap/10-1, Out od DISA command:hangup) in new stack
-- Executing Macro(Zap/10-1, hangupcall) in new stack
-- Executing ResetCDR(Zap/10-1, w) in new stack
-- Executing NoCDR(Zap/10-1, ) in new stack
-- Executing Wait(Zap/10-1, 2) in new stack
-- Executing Hangup(Zap/10-1, ) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/10-1'
in macro 'hangupcall'
== Spawn extension (from-bri-user, h, 2) exited non-zero on 'Zap/10-1'
-- Hungup 'Zap/10-1'
It seems like ti goes to hangup priority in context it was called from - but
I'd like to continue with other actions in dialplan - how to do that ?
Regards,
Rob.
On 6/5/05, Robert Rozman [EMAIL PROTECTED] wrote:
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, June 06, 2005 1:27 AM
Subject: Re: [Asterisk-Users] Disa - how it returns on user not dialing
anynumbers ?
Try the t extension (stands for timeout).
Hi, thanks for advice. But it seems that t extension doesn't get executed.
I
have it in context from where I call DISA and also in DISA context, but no
go
After dialing nothing and some time it just gets hungup, no t extension
triggering .
Any advice ?
Thanks in advance,
regards,
Rob.
I have :
exten = s,14,DigitTimeout(5)
exten = s,15,ResponseTimeout(20)
exten = s,16,DBget(temp=USER/${CALLERIDNUM})
exten = s,17,SetAccount(${temp})
exten = s,18,DISA(no-password|from-bri-user-disa)
exten = t,1,DBget(temp=DYNAMIC/${CALLERIDNUM})
exten = t,2,DBdel(DYNAMIC/${CALLERIDNUM})
exten = t,3,Dial(Local/[EMAIL PROTECTED]/n)
exten = t,102,Goto(from-pstn,s,1) ;
[from-bri-user-disa]
include = outbound-allroutes-custom
include = outrt-001-Mednarodni
include = ext-local
include = outrt-003-zunanja
exten = s,1,NoOp(DISA: Waiting for BRI user to enter number)
exten = s,2,DigitTimeout(5) ; Vhodni
exten = s,3,ResponseTimeout(20) ;
exten = t,1,DBget(temp=DYNAMIC/${CALLERIDNUM})
exten = t,2,DBdel(DYNAMIC/${CALLERIDNUM})
exten = t,3,Dial(Local/[EMAIL PROTECTED]/n)
exten = t,102,Goto(from-pstn,s,1) ;
On 6/5/05, Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
I'd like to use DISA properly for my case - I'd like to handle it right,
if
user when in DISA doesn't dial any number - how does Asterisk return
from
DISA cmd ?
I'd like to dial some default number if user doesn't dial anything or
give
him some message - but I don't know what gets executed after DISA if
nothing
is dialed
I'm reading this on wiki, but don't understand what following sentence
means. Any help ?
... If login is successful, the application parses the dialed number in
the specified (or default) context, and returns 0 with the new extension
context filled-in and the priority set to 1, so that the PBX may
re-apply the routing tables to it and complete the call normally. ...
Thanks in advance,
regards,
Rob.
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[Asterisk-Users] Does Debian Bristuffed Asterisk work ignore Beronet cards ?
Hi, we've purchased new Beronet Octobri card and have problems loading modules for stock Asterisk for Debian Sarge (it has bristuff patches in it). when loading qozap it says that no multibri card was found although lspci shows it... There were quite some rumours about bristuff not liking other than junghanns cards, but don't know if something happened Anyone recently used this card and Debian Asterisk and can confirm that this is working ? Any advice or hint, what should be done to get it into working state ? Can qozap output some more debug info ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Debian Bristuffed Asterisk work ignoreBeronet cards ?
- Original Message - From: Emanuele Pucciarelli [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 02, 2005 7:44 PM Subject: Re: [Asterisk-Users] Does Debian Bristuffed Asterisk work ignoreBeronet cards ? Robert Rozman wrote: when loading qozap it says that no multibri card was found although lspci shows it... There were quite some rumours about bristuff not liking other than junghanns cards, but don't know if something happened http://www.beronet.com/download/card_installation_guide_en.pdf On page 37 you'll find that bristuff must be patched in order to recognize other cards. Hi, thanks for info. I've read that but on quadbri Beronet card we purchased 2 months ago everything worked without any changes... Do you know where to find those patches and if they are really necessary ? Thanks in advance, regards, Rob. Anyone recently used this card and Debian Asterisk and can confirm that this is working ? Any advice or hint, what should be done to get it into working state ? I've never tried, but in similar situations I've changed the PCI ids in the drivers. This will surely make any driver recognise the card; it doesn't mean that it will surely work, though :) -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unreliable DTMF detection with DISA on incoming Zap channel on bristuffed * and GSM gateway
Hi, I'm getting unusable DTMF detection with DISA on incoming ZAP channel (bristuffed *) on quadbri from GSM gateway. DTMF detection works ok in normal ISDN incoming line. How can I check what's going on ? What settings to check ? Anyone with more experience on such scenarios ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I hide caller id on the fly (per each usesetting) on Bristuffed * and quadbri
- Original Message - From: Peer Oliver Schmidt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 04, 2005 11:14 PM Subject: Re: [Asterisk-Users] Can I hide caller id on the fly (per each usesetting) on Bristuffed * and quadbri Robert Rozman wrote: I wonder if I can hide caller id for just certain users. Can I override caller id setting for show or hide on the fly from dialplan ? Did you try setcallerid()? -- I tried but this will work if calling internal line. I'm after dynamically hiding caller id on QuadBRI outgoing ISDN calls... I guess this is possible with settings in zapata.conf, but only per channel - I wonder if it is possible to set this up by user or do it from dialplan with some command Thanks in advance, Rob. Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I hide caller id on the fly (per each use setting) on Bristuffed * and quadbri
Hi, I wonder if I can hide caller id for just certain users. Can I override caller id setting for show or hide on the fly from dialplan ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any workaround for long DISA timeout before it actually dials ?
Hi, I'm finding long timeout before DISA really calls extension user entered annoying. I wonder what workarounds are you using for this ? Playtones is one possibility , but it won't stop when user starts entering numbers... Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fritz+chan_misdn - any working example ?
Hi, I'd kindly ask if anyone can provide working configuration examples for Asterisk-Fritz-mISDN combo. Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream : low bandwidth codec (ilbc doesn't work, any other ? )
Hi, I'm trying to setup one of free low bandwidth codecs for Grandstream (ilbc, g726, ...), but with ilbc I just hear engine running in handset. Is anyone using ilbc sucessfully with Grandstream? Quality ? Any other alternative ? I use Bristuffed Asterisk Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quadbri bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?
Hi, I have problem with Quadbri and bristuffed Asterisk - I guess this is only configuration trick. I'd like Asterisk to respond only to 1 number on BRI interface and do nothing on other. Right now, even if I leave out that number in incoming context, Asterisk takes out and rejects call as number is non existant. I'd like that Asterisk wouldn't respond, so other ISDN phone can answer it I did this with chan_capi, where you can determin incoming MSNs on which Asterisk responds, but wonder how to do that with Zap channels Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323: Sending CallerID to H323 voip provider...
Hi, I'm trying to make a OH323 call with Grandstream, Asterisk and H323 voip provider. Everything seems to work fine, except callerid is set to some value for all calls (10100), despite setcallerid statements in Asterisk. Are there any special considerations or tricks to get this working ? I've contract that voip provider will accept custom CallerID Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why 's' doesn't take over unknown extension in context ?
Hi,
I always thought that if there is no called extension in context, then 's'
extension is started (I use latest bristuffed Asterisk)
I have context 'from-isdn' :
[from-isdn]
exten = s,1,Wait,2
exten = s,2,NoOp(ISDN call from outside ${CALLERID}: Name: ${CALLERIDNAME},
Number: ${CALLERIDNUM})
exten = s,3,SetCIDName(From ISDN: ${CALLERIDNUM})
exten = s,4,SetCIDNum(0${CALLERIDNUM})
exten = s,5,AGI,callerid_lookup.agi
exten = s,6,NoOp(After callerid_lookup.agi: ${CALLERID}: Name:
${CALLERIDNAME}, Number: ${CALLERIDNUM})
exten = s,7,DBget(temp=DYNAMIC/${CALLERIDNUM})
exten = s,8,DBdel(DYNAMIC/${CALLERIDNUM})
exten = s,9,Background(custom/aa_1)
exten = s,10,Wait,5
exten = s,11,Dial(Local/[EMAIL PROTECTED]/n)
exten = s,108,Goto(from-pstn,s,1) ;
exten = 99,1,Goto(s,1) ;
Now if there is no line 99 on incoming call I get :
-- Extension '99' in context 'isdn-incoming' from '041461620' does
not exist. Rejecting call on channel 0/1, span 1
Why doesn't extension 's' get started if extension 99 is unknown in
context from-isdn?
Thanks in advance,
regards,
Rob.
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Re: [Asterisk-Users] Latest Bristuff crashes on modprobe -r qozap ?
- Original Message - From: Michael Bielicki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 06, 2005 5:54 PM Subject: Re: [Asterisk-Users] Latest Bristuff crashes on modprobe -r qozap ? do a ztcfg -s before doing a modprobe -r qozap On Apr 6, 2005 4:45 PM, Robert Rozman [EMAIL PROTECTED] wrote: uname -a Hi, I'm using latest Bristuffed Asterisk under Suse 9.2 and upgraded kernel. Everything seems to be working fine, except crash when removing qozap with modprobe -r qozap. Hi, I'm still getting same crashes Anything else to try ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why 's' doesn't take over unknown extensionincontext ?
- Original Message - From: Eric Wieling aka ManxPower [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 11, 2005 9:43 PM Subject: Re: [Asterisk-Users] Why 's' doesn't take over unknown extensionincontext ? Steve Mann wrote: I think it is i you want, s is the start for a context, meaning anything coming in through that context will start there, i is invalid, and fires if an invalid extension is keyed in that context. s is run when a call comes in and Asterisk does not know the dialed number. It does NOT mean meaning anything coming in through that context will start there __ Hi, thanks for explanation... I had this working, but for calls from CAPI - I remember that Asterisk said something about back off to default 's' extension or something similar. Obviously chan_capi didn't send called number to Asterisk and this happened, but that's not the case with Zap calls... Thanks, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delayed dial under Asterisk ?
Hi, I'd like to setup delayed dial under Asterisk. That means that at the caller side I set up number *YY and call Asterisk PBX (XXX... is number of Asterisk PBX, * means pause (2 secs), YY is internal number). Has anyone experience with receiving such calls ? How should I setup Asterisk dialplan for that ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Delayed dial under Asterisk ?
- Original Message -
From: Mick Hastings [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 08, 2005 1:07 PM
Subject: [Asterisk-Users] Re: Delayed dial under Asterisk ?
Hi Robert,
I just set this up today for dialing international using a calling card
account.
usually we call 0120 982 433
wait for voice prompt
then dial the number
i set it up so the user only has to prefix with 011 then the number like
this:
[brastel]
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED],,TM(BRASTEL^${EXTEN:3}))
exten = _011.,2,Hangup
[macro-BRASTEL]
exten = s,1,Wait(2)
exten = s,2,SendDTMF(${ARG1})
this way the user dials this: 011 61 3 9556 7787
and asterisk does this:
dials 0120 982 433
waits for connect
then waits 2 seconds
then sends 61 3 9556 7787
seems to work for me just fine.
cheers,
Mick
Thanks for useful info. I still wonder if Asterisk has any problem receiving
delayed dial numbers (like first 8 numbers to get to Asterisk, then wait for
2 secs and then send internal number...).
regards,
Rob.
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[Asterisk-Users] Latest Bristuff crashes on modprobe -r qozap ?
uname -a Hi, I'm using latest Bristuffed Asterisk under Suse 9.2 and upgraded kernel. Everything seems to be working fine, except crash when removing qozap with modprobe -r qozap. Any hint what's wrong ? Thanks in advance, regards, Rob. Linux voip 2.6.8-24.13-smp #1 SMP Fri Mar 18 10:19:42 UTC 2005 i686 i686 i386 GNU/Linux Badness in smp_call_function at arch/i386/kernel/smp.c:555 [c01183c4] smp_call_function+0x114/0x120 [c0158a29] unmap_area_pmd+0x39/0x50 [c011827b] flush_tlb_all+0x1b/0x30 [c0158ea5] remove_vm_area+0x45/0x70 [c011cc9a] iounmap+0x3a/0xc0 [e0d4d118] qoz_shutdownCard+0x118/0x1e0 [qozap] [e0d4ddb0] cleanup_module+0x0/0x90 [qozap] [e0d4ddd7] cleanup_module+0x27/0x90 [qozap] [c0139adf] try_stop_module+0x1f/0x30 [c0139cd7] sys_delete_module+0x157/0x170 [c0155a6a] unmap_vma_list+0x1a/0x30 [c0155e25] do_munmap+0x125/0x190 [c0155ed8] sys_munmap+0x48/0x70 [c0107029] sysenter_past_esp+0x52/0x79 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Suse minimal installation based on Suse Rescue - what to add to be bootable on HD partition ?
Hi, I'm trying to go route some of Asterisk users already proposed for Asterisk minimal system. I've started from Suse Rescue system image - I've put it into HD partition. But since rescue is spawned from working system it has empty /boot directories and is not directly bootable if put on HD. I've tried to transfer or install kernel and grub to this partition, but no success (I first access to partition with chroot to make additions...). I get errors on kernel rpm -ivh installation (I guess there are no directories and dependencies found in chroot) and also get error on grub-install /dev/hda7: Could not find device for /boot: Not found or not a block device Is there anyone more experienced with some advice, howto or example what need to be added to partition to be bootable on HD ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem on outgoing calls (quadbri card and bristuffed Asterisk latest) ?
Hi, I have strange behavior on outgoing calls (I can receive calls and I can make outgoing calls to ISDN lines ok (035778421 and 5778421 for instance - 03 is area code). I use latest bristuffed Ast. under Suse 9.2. My zapata.conf and zaptel.conf are at the end of mail. Any help, advice - I guess there is something wrong with settings... But when I call my cellular on 041 461 620 - exactly as I type on phone, I get this : -- Executing Dial(IAX2/[EMAIL PROTECTED]/6, ZAP/g1/041461620|60) in new stack -- Called g1/041461620 -- Zap/1-1 is making progress passing it to IAX2/[EMAIL PROTECTED]/6 -- Channel 0/1, span 1 got hangup Mar 30 14:40:10 WARNING[9744]: app_dial.c:412 wait_for_answer: Unable to forward voice -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Hangup(IAX2/[EMAIL PROTECTED]/6, ) in new stack == Spawn extension (from-internal, 041461620, 3) exited non-zero on 'IAX2/[EMAIL PROTECTED]/6' -- Hungup 'IAX2/[EMAIL PROTECTED]/6' and under debug span 1: [18 01 89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 133/0x85) (Terminator) Message type: CALL PROCEEDING (2) -- Zap/1-1 is making progress passing it to IAX2/[EMAIL PROTECTED]/4 Protocol Discriminator: Q.931 (8) len=23 Call Ref: len= 1 (reference 133/0x85) (Terminator) Message type: DISCONNECT (69) [08 02 82 83] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: No route to destination (3), class = Normal Event (0) ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] [28 09 4e 4f 20 52 4f 55 54 45 20] Display (len= 9) [ NO ROUTE ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 40 (cs0, Display) -- Channel 0/1, span 1 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 5/0x5) (Originator) Message type: RELEASE (77) [08 02 81 83] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: No route to destination (3), class = Normal Event (0) ] -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Hangup(IAX2/[EMAIL PROTECTED]/4, ) in new stack == Spawn extension (from-internal, 041461620, 3) exited non-zero on 'IAX2/[EMAIL PROTECTED]/4' -- Hungup 'IAX2/[EMAIL PROTECTED]/4' Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 133/0x85) (Terminator) Message type: RELEASE COMPLETE (90) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null **/etc/zaptel.conf loadzone=nl defaultzone=nl # qozap span definitions # most of the values should be bogus because we are not really zaptel #span=1,1,3,ccs,hdb3 #span=2,0,3,ccs,hdb3 #span=3,0,3,ccs,hdb3 #span=4,0,3,ccs,hdb3 span=1,1,3,ccs,ami,crc4 span=2,0,3,ccs,ami,crc4 span=3,0,3,ccs,ami,crc4 span=4,0,3,ccs,ami,crc4 bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 **/etc/asterisk/zapata.conf [channels] switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres=yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 ;callerid=asreceived overlapdial=yes ;--- ; p2p TE mode (for connecting ISDN lines in point-to-point mode) ;signalling = bri_cpe ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp context=isdn-incoming group = 1 ; S/T port 1-3 (first quadBRI, or lower ports of an octoBRI) channel = 1-2 ;channel = 4-5 ;channel = 7-8 ;--- ; p2p NT mode (for connecting an ISDN PBX in point-to-point mode) signalling = bri_net context=pbx-incoming group = 2
[Asterisk-Users] Confused: Qozap is on interrupt 209 alone - is this good or not ?
Hi, I'm confused whether I setup PC for Asterisk right or not. Module qozap is alone (yet not sharing) in interrupt 209 (isn't this too high for native interrupt). Is this good state or not? If not, how to setup better ? Thanks in advance, regards, Rob. voip:~ # cat /proc/interrupts CPU0 CPU1 0:2874244 0IO-APIC-edge timer 1: 30 0IO-APIC-edge i8042 9: 0 0 IO-APIC-level acpi 12: 78 0IO-APIC-edge i8042 14: 8862 1IO-APIC-edge ide0 15: 52 0IO-APIC-edge ide1 177: 483517 0 IO-APIC-level Intel ICH5 185: 10531 0 IO-APIC-level SysKonnect SK-98xx 209:1951868 0 IO-APIC-level qozap NMI: 0 0 LOC:28742352874318 ERR: 0 MIS: 0 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as gateway with oh323 channel to VOIP provider that can provide gateway or gatekeeper feature ?
Hi, sorry for my h323 dumbness. VOIP provider terminates H323 calls - it can be used as gatekeeper or gateway (they claim so). What option and what setup is best to connect Asterisk to this provider ? Any working examples ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting quadbri to EuroISDN with 2 TE and 2 NT ports - what cables and settings ?
Hi, I'm trying to connect quadbri between powered ISDN phone and ISDN line: ISDN ---1--- TE - * - NT --2-- Phone I use quadbri, suse 9.2 and latest 0.2.0-RC7k bristuff. I've used sample settings provided with package, but do get strange error (I think that I have wrong setting for P2P or P2MP setting and cables 1 and 2). If I connect phone to ISDN with straight cable it works. I've put quadbri in between, and connected ISDN to span1 in TE mode, and phone in NT mode on span4. Did configuration (added at the end). I get errors: qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 I'm not sure what cables to use. I use straight for -1- and -2-. Is this right ? I'm in European community (EuroISDN) so I guess I should have some pretty standard connections... I have ISDN line with 2 MSNs and two connectors on NT termination - so I guess this is P2MP. Is this OK ? Does anyone have working example for Germany ? There is also one strange thing: I get this in dmesg when loading qozap module, although I have specified 4th port to be NT - and pri show span 4 shows it in netowork mode: Zapata Telephony Interface Unloaded module zaptel unsupported by SUSE/Novell, tainting kernel. Zapata Telephony Interface Registered on major 196 module qozap unsupported by SUSE/Novell, tainting kernel. PCI: Enabling device :02:0c.0 ( - 0003) ACPI: PCI interrupt :02:0c.0[A] - GSI 20 (level, low) - IRQ 209 qozap: S/T ports: 4 [ TE TE TE TE ] qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Registered tone zone 3 (Netherlands) Thanks in advance, regards, Rob. #- /etc/zaptel.conf: loadzone=nl defaultzone=nl # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 #- /etc/asterisk/zapata.conf [channels] switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres=yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 ;--- ; p2p TE mode (for connecting ISDN lines in point-to-point mode) ;signalling = bri_cpe ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp context=isdn-incoming group = 1 ; S/T port 1-3 (first quadBRI, or lower ports of an octoBRI) channel = 1-2 channel = 4-5 channel = 7-8 ;--- ; p2p NT mode (for connecting an ISDN PBX in point-to-point mode) signalling = bri_net context=pbx-incoming group = 2 ; S/T port 4 (second quadBRI, or upper ports of an octoBRI) channel = 10-11 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff-0.2.0-RC7k: error on loading qozap : qozap: Unknown symbol zt_xxxxx
Hi, I had problems described in another thread so went from a start, but now have problems when loading qozap module. I get : # insmod qozap.ko ports=9 insmod: error inserting 'qozap.ko': -1 Unknown symbol in module and in /var/log/messages: module qozap unsupported by SUSE/Novell, tainting kernel. qozap: disagrees about version of symbol zt_receive qozap: Unknown symbol zt_receive qozap: disagrees about version of symbol zt_ec_chunk qozap: Unknown symbol zt_ec_chunk qozap: disagrees about version of symbol zt_transmit qozap: Unknown symbol zt_transmit qozap: disagrees about version of symbol zt_unregister qozap: Unknown symbol zt_unregister qozap: disagrees about version of symbol zt_register qozap: Unknown symbol zt_register I did start from clean Suse 9.2 : cd /usr/src/linux make clean make mrproper make cloneconfig make prepare-all ln -s /usr/src/linux-2.6.8-24.13/ /usr/src/linux-2.6 cp /usr/src/linux-2.6.8-24.13-obj/i386/smp/Module.symvers /usr/src/linux // cause of warning when compiling zaptel echo # Section for zaptel device /etc/udev/rules.d/50-udev.rules echo KERNEL=\zapctl\, NAME=\zap/ctl\ /etc/udev/rules.d/50-udev.rules echo KERNEL=\zaptimer\, NAME=\zap/timer\ /etc/udev/rules.d/50-udev.rules echo KERNEL=\zapchannel\, NAME=\zap/channel\ /etc/udev/rules.d/50-udev.rules echo KERNEL=\zappseudo\, NAME=\zap/pseudo\ /etc/udev/rules.d/50-udev.rules echo KERNEL=\zap[0-9]*\, NAME=\zap/%n\ /etc/udev/rules.d/50-udev.rules echo /etc/udev/rules.d/50-udev.rules echo zap/*:root:root:660 /etc/udev/permissions.d/50-udev.permissions echo /etc/udev/rules.d/50-udev.rules cd zaphfc/ wget http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC7k.tar.gz tar zxvf bristuff-0.2.0-RC7k.tar.gz cd bristuff-0.2.0-RC7k/ ./download.sh ./compile.sh #clearing SuSE deprecated SuSE modules for module in /lib/modules/`uname -r`/misc/*; do rm -i /lib/modules/`uname -r`/extra/$(basename $module); done #Loading the drivers (quadBRI): cd qozap modprobe zaptel #insmod qozap.o (for kernel 2.4) insmod qozap.ko ports=9 (for kernel 2.6) //This is where I got errors... ztcfg Any advice, what's wrong ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RSA interasterisk IAX problems ?
Hi, I'd like to setup oneway connection - so asteriskB can place calls on asteriskA and be safely authenticated with rsa keys. I just don't get any response on asteriskA. I've generated pair of keys: name.key, name.pub and put them on both servers - is it right to only have name.key on asteriskA and name.pub on asteriskB ? I get everybody is busy ... on asteriskB, and none response on asteriskA. What am I doing wrong? Does anyone have working example of this ? I have following setup (UDP 5036 ports are opened and forwarded on both Asterisk, they are both behind NATs): 1. asteriskA *iax.conf: [asteriskA] type=user host=voip.xxx.xx username=asteriskA auth=rsa inkeys=name context=default accountcode=asteriskA 2. asteriskB *iax.conf: [asteriskA] type=peer host=xxx.xxx.xxx.xxx auth=rsa outkey=name username=asteriskA *extensions.conf: exten = _00[34][01].,3,Dial(IAX/asteriskA:[EMAIL PROTECTED]/[EMAIL PROTECTED],30) or exten = _00[34][01].,3,Dial(IAX/asteriskA/[EMAIL PROTECTED],30) but nogo. Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RSA interasterisk IAX problems ?
Hi, I'd kindly ask if anyone can provide working example of RSA authentication and IAX ? Thanks in advance, regards, Rob. - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 24, 2005 12:16 PM Subject: [Asterisk-Users] RSA interasterisk IAX problems ? Hi, I'd like to setup oneway connection - so asteriskB can place calls on asteriskA and be safely authenticated with rsa keys. I just don't get any response on asteriskA. I've generated pair of keys: name.key, name.pub and put them on both servers - is it right to only have name.key on asteriskA and name.pub on asteriskB ? I get everybody is busy ... on asteriskB, and none response on asteriskA. What am I doing wrong? Does anyone have working example of this ? I have following setup (UDP 5036 ports are opened and forwarded on both Asterisk, they are both behind NATs): 1. asteriskA *iax.conf: [asteriskA] type=user host=voip.xxx.xx username=asteriskA auth=rsa inkeys=name context=default accountcode=asteriskA 2. asteriskB *iax.conf: [asteriskA] type=peer host=xxx.xxx.xxx.xxx auth=rsa outkey=name username=asteriskA *extensions.conf: exten = _00[34][01].,3,Dial(IAX/asteriskA:[EMAIL PROTECTED]/[EMAIL PROTECTED],30) or exten = _00[34][01].,3,Dial(IAX/asteriskA/[EMAIL PROTECTED],30) but nogo. Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RSA interasterisk IAX problems ?
- Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 25, 2005 2:13 AM Subject: Re: [Asterisk-Users] RSA interasterisk IAX problems ? Hi, I'd kindly ask if anyone can provide working example of RSA authentication and IAX ? Thanks in advance, regards, Rob. - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 24, 2005 12:16 PM Subject: [Asterisk-Users] RSA interasterisk IAX problems ? Hi, I'd like to setup oneway connection - so asteriskB can place calls on asteriskA and be safely authenticated with rsa keys. I just don't get any response on asteriskA. I've generated pair of keys: name.key, name.pub and put them on both servers - is it right to only have name.key on asteriskA and name.pub on asteriskB ? I get everybody is busy ... on asteriskB, and none response on asteriskA. What am I doing wrong? Does anyone have working example of this ? I have following setup (UDP 5036 ports are opened and forwarded on both Asterisk, they are both behind NATs): 1. asteriskA *iax.conf: [asteriskA] type=user host=voip.xxx.xx username=asteriskA auth=rsa inkeys=name context=default accountcode=asteriskA 2. asteriskB *iax.conf: [asteriskA] type=peer host=xxx.xxx.xxx.xxx auth=rsa outkey=name username=asteriskA *extensions.conf: exten = _00[34][01].,3,Dial(IAX/asteriskA:[EMAIL PROTECTED]/[EMAIL PROTECTED],30) or exten = _00[34][01].,3,Dial(IAX/asteriskA/[EMAIL PROTECTED],30) but nogo. Thanks in advance, regards, Rob. Hi, I've found solution to my own problem. There is major lapsus on wiki on page for rsa authentication - example line starts with IAX/ instead of IAX2/. That solved my problem. But I needed to copy both keys (public and private) to both asterisks. Is this right ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] te110p sometimes green, sometimes stays red on stable cvs ?
Hi, we've installed te110p with Suse 9.2 on Siemens primergy. We're connecting to voxsteam i60 to test PRI interface. We have problems, after reboot sometimes it goes green, otherwise stays blinking red. How could we debug this situation ? Are there any common advices what to check ? Are CVS Head drivers better than CVS stable ? Why ? Do you also experience such unstable behaviour ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems loading zapata module under suse 9.2 (cvs stable from 5 days ago) ?
Hi, I've compiled Asterisk cvs stable (few days ago) unde Suse 9.2 without any problems. We're using te110p and wcte11xp module that is autoloaded by Suse 9.2. Card goes green after reboot, but this meesages appear in logs: Mar 22 11:28:51 linux kernel: Zapata Telephony Interface Registered on major 196 Mar 22 11:28:51 linux modprobe: FATAL: Error inserting torisa (/lib/modules/2.6.8-24.11-smp/extra/torisa.ko): Unknown symbol in module, or unknown parameter (see dmesg) Mar 22 11:28:51 linux modprobe: FATAL: Error running install command for torisa Mar 22 11:28:51 linux kernel: module torisa unsupported by SUSE/Novell, tainting kernel. Mar 22 11:28:51 linux kernel: torisa: disagrees about version of symbol zt_receive Mar 22 11:28:51 linux kernel: torisa: Unknown symbol zt_receive Mar 22 11:28:51 linux kernel: torisa: disagrees about version of symbol zt_ec_chunk Mar 22 11:28:51 linux kernel: torisa: Unknown symbol zt_ec_chunk Mar 22 11:28:51 linux kernel: torisa: disagrees about version of symbol zt_transmit Mar 22 11:28:51 linux kernel: torisa: Unknown symbol zt_transmit Mar 22 11:28:51 linux kernel: torisa: disagrees about version of symbol zt_rbsbits Mar 22 11:28:51 linux kernel: torisa: Unknown symbol zt_rbsbits Mar 22 11:28:51 linux kernel: torisa: disagrees about version of symbol zt_unregister Mar 22 11:28:51 linux kernel: torisa: Unknown symbol zt_unregister Mar 22 11:28:51 linux kernel: torisa: disagrees about version of symbol zt_register Mar 22 11:28:51 linux kernel: torisa: Unknown symbol zt_register Mar 22 11:28:51 linux kernel: torisa: disagrees about version of symbol zt_alarm_notify Mar 22 11:28:51 linux kernel: torisa: Unknown symbol zt_alarm_notify Any idea what's wrong and where to start digging ? Also we have specified slot with PRI card to interrupt 9 (bios setting in Siemens Primergy Econel server), but we don't get quite expected results when : linux:~ # cat /proc/interrupts CPU0 0: 39855639IO-APIC-edge timer 1: 11IO-APIC-edge i8042 2: 0 XT-PIC cascade 15: 38IO-APIC-edge ide1 137: 0 IO-APIC-level uhci_hcd 153: 130872 IO-APIC-level libata, eth0 161: 0 IO-APIC-level uhci_hcd 169: 39804328 IO-APIC-level t1xxp 177: 0 IO-APIC-level ehci_hcd NMI: 0 LOC: 39858292 ERR: 0 MIS: 0 Is this OK ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Te110P initial installation problems ?
- Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 19, 2005 12:36 AM Subject: [Asterisk-Users] Te110P initial installation problems ? Hi, thank you for last info. we've tried to use te110p but failed. We're quite surprised that cable wasn't included with the card as any documentation, at least on HW setup and installation, yet cable pinout for connection to PRI interfaces 1. We have followed instructions on your site and from Beronet guide, but card just keeps blinking and nothing happens (also no useful info in logs). Then we suspected cable, but couldn't find out what pinout does PRI interface on te110p have ? I'd kindly ask if anyone knows pinout of te110p connector to help me... 2. Also we've spotted weird behaviour of wcte11xp module. If I do modprobe -r wcte11xp Siemens primergy econel server under Suse 9.2 freezes hard... Is this normal behaviour ? Do we have any debug options on loading module ? How to track this problems ? span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=fr defaultzone=fr Anyone ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Te110P initial installation problems ?
Hi, thank you for last info. we've tried to use te110p but failed. We're quite surprised that cable wasn't included with the card as any documentation, at least on HW setup and installation, yet cable pinout for connection to PRI interfaces 1. We have followed instructions on your site and from Beronet guide, but card just keeps blinking and nothing happens (also no useful info in logs). Then we suspected cable, but couldn't find out what pinout does PRI interface on te110p have ? 2. Also we've spotted weird behaviour of wcte11xp module. If I do modprobe -r wcte11xp Siemens primergy econel server under Suse 9.2 freezes hard... Is this normal behaviour ? Do we have any debug options on loading module ? How to track this problems ? span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=fr defaultzone=fr Thanks in advance, regards, Robert Rozman. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What cable to connect TE110P to telco PRI ?
Hi, call me stupid, but cable is not delivered with te110p. What cable can I use to connect to telco NT PRI line ? Is it same as for BRI interface ? Thanks, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons?
Hi, I'd also like to see alternative op_style.cfg. Can we establish some place to share them ? I've also one with smaller buttons (but will have to count them :-) ... Regards, Rob. - Original Message - From: Nicolás Gudiño [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 16, 2005 1:26 PM Subject: Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons? Hi Ronald, I have setup flash pannel, ... looks nice, but so far I could not configure it to get more than 4x7 buttons. I tried to make the buttons smaller, but than just the entire picture is smaller. What did you change in op_style.cfg? You can have literally hundred of buttons per screen, or multiple 'context' to split your buttons into several screens. I wll send you an alternate op_style.cfg with smaller buttons offlist. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stable CVS or Head CVS for using TE110P ?
Hi, I'd like to know which version of Asterisk performs best and most stable with TE110P. I don't need any other features (it'll just terminate interasterisk calls without any other feature - so there is no need for CVS Head features or ? ). Any info on setting up secure interasterisk IAX connections (only one way) ? With IAX authentication by certificates ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: [Asterisk-Dev] SetVarCDR
Hi,
could anyone provide any working examples? I have same problem, I do
redirect to s-${Dialstatus} when making outgoing calls and I also get s-
into CDRs.
Thanks,
Rob.
- Original Message -
From: William M. Sandiford [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, March 13, 2005 5:19 AM
Subject: [Asterisk-Users] RE: [Asterisk-Dev] SetVarCDR
I don't know...now I have a _X. in my CDR.
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Saturday, March 12, 2005 8:05 PM
To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com
Subject: Re: [Asterisk-Dev] SetVarCDR
You must have some fux0red config 'cause using _X. works fine here. I
haven't had an 's' in my CDRs for over several months now.
(continue this on the -users list.)
-Matthew
From: William M. Sandiford [EMAIL PROTECTED]
Reply-To: Asterisk Developers Mailing List
asterisk-dev@lists.digium.com Asterisk-Dev@lists.digium.com
Date: Sat, 12 Mar 2005 13:41:30 -0500
To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com
Subject: RE: [Asterisk-Dev] SetVarCDR
Yes, but putting showing s in the dst is not what I want, I'm trying
to alter that. Also your suggestion of using _X. didn't work either.
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Saturday, March 12, 2005 11:20 AM
To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com
Subject: Re: [Asterisk-Dev] SetVarCDR
*buzzer* That's the sound I make when somebody lies to me.
- Dr. Cox, Scrubs
I replied to your post on the -users list about this. Showing 's' in
the dst is programaticly correct behavior.
-Matthew
From: William M. Sandiford [EMAIL PROTECTED]
Reply-To: Asterisk Developers Mailing List
asterisk-dev@lists.digium.com Asterisk-Dev@lists.digium.com
Date: Fri, 11 Mar 2005 18:30:03 -0500
To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com
Subject: RE: [Asterisk-Dev] SetVarCDR
This was asked to the -users list with no replies. Since the
original post came from the CVS list and was obviously a recent
development change, I thought this list was appropriate. My
apologies if it was not.
Out of curiosity, what are CDR variables for then? Are they
read-only?
Regards,
Bill
-Original Message-
From: Tilghman Lesher [mailto:[EMAIL PROTECTED]
Sent: Friday, March 11, 2005 6:07 PM
To: Asterisk Developers Mailing List
Subject: Re: [Asterisk-Dev] SetVarCDR
On Friday 11 March 2005 16:40, William M. Sandiford wrote:
I found a reference to the application SetVarCDR in the following
post but I don't seem to have this available to me in my version of
*.
This is a -users question. Please do not post usage questions to the
developers list.
I would like to change the value of the src and dst variables in the
CDR as I sometimes find that they don't have entirely accurate
information. For example my dst field quite often has a value of
s because I do my call processing in the s extension. This is no
good to me.
This isn't what CDR variables are for. In fact, unless you have
coded your own custom cdr_*.c module, you probably shouldn't be using
them at all (at least until somebody codes a generic implementation).
--
Tilghman
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[Asterisk-Users] Asterisk Fritz Capi isdn PBX integration : Can I dial out on any MSN I declare ?
Hi, I'm integrating Asterisk to legacy PBX via ISDN router. If I want to call legacy PBX internal extension I need to specify MSN as caller id and local number to call. I wonder if I can cal out via Fritz CAPI on any msn I want, or are there any limitations - I've read something about 5 MSNs limitation and wonder if it still holds ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type IAX2
I don't know if this is still true, but Iax clients had problems when you check them with qualify (set latter to no)... HTH, Rob. - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, March 05, 2005 9:05 AM Subject: RE: [Asterisk-Users] Unable to create channel of type IAX2 And when it does work, the console says: Mar 5 02:07:08 NOTICE[9962]: chan_iax2.c:7065 iax2_poke_noanswer: Peer 'akralliax' is now UNREACHABLE! Time: 5 Mar 5 02:07:18 NOTICE[9962]: chan_iax2.c:6420 socket_read: Peer 'akralliax' is now REACHABLE! Time: 3 The iaxcomm phone is on the same LAN, so why can it be coming and going? Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sábado, 05 de Marzo de 2005 01:55 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Unable to create channel of type IAX2 Guys.. Im trying to setup a fotphone using iaxcomm and when I dial that softphones extension, * complains of this: Mar 5 01:54:54 NOTICE[9962]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) Any hints? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wrong CVS version ?
Hi, I've updated my Asterisk 3 times with : cvs checkout -r v1-0 zaptel asterisk asterisk-addons and then do cd asterisk make clean make make install make samples make progdocs and then when I run Asterisk I get : Asterisk CVS-v1-0-02/11/05-01:46:25, Copyright (C) 1999-2004 Digium. Is this a bug in CVS handling or am I doing something wrong ? How to check which version of CVS I have from commandline ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wrong CVS version ?
- Original Message - From: Adnan Ahmed [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 1:28 AM Subject: Re: [Asterisk-Users] Wrong CVS version ? you are compiling in wrong sequence first zaptel then asterisk and after that asterisk-addons . hope this helps ??? How this influence on cvs version of asterisk binary ??? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why ${EXTEN} variable changes after Goto ?
Hi,
I'm trying to implement dynamic routing of incoming calls to local extension
if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand why ? How to
workaround on this one ?
Thanks in advance,
regards,
Rob.
[outbound-capi-ISDN]
exten = _0.,1,NoOp(Calling ISDN number ${EXTEN:1} on CAPI/7104370 from
${CALLERIDNUM})
exten = _0.,2,Dial,CAPI/7104370:b${EXTEN:1}|10|Tt
exten = _0.,3,Goto(s-${DIALSTATUS},1)
exten = _0.,103,NoOp(Calling ISDN number ${EXTEN:1} on CAPI/7104371)
exten = _0.,104,Dial,CAPI/7104371:b${EXTEN:1}|30|Tt
exten = _0.,105,Goto(s-${DIALSTATUS},1)
exten = _0.,205,Macro(outisbusy)
exten = s-NOANSWER,1,NoOp(NOANSWER - Setting dynamic autoroute for ISDN
number ${EXTEN:1} to local ext. ${CALLERIDNUM})
exten = s-NOANSWER,2,DBput(DYNAMIC/${EXTEN}=${CALLERIDNUM})
exten = s-NOANSWER,3,Congestion
exten = _s-.,1,Congestion
exten = _s-.,2,Macro(hangupcall)
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Re: [Asterisk-Users] Why ${EXTEN} variable changes after Goto ?
- Original Message -
From: Umar Sear [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 03, 2005 11:01 PM
Subject: Re: [Asterisk-Users] Why ${EXTEN} variable changes after Goto ?
On Thu, 3 Mar 2005 22:11:23 +0100, Robert Rozman [EMAIL PROTECTED]
wrote:
Hi,
I'm trying to implement dynamic routing of incoming calls to local
extension
if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand why ? How to
workaround on this one ?
Thanks in advance,
regards,
Rob.
[outbound-capi-ISDN]
exten = _0.,1,NoOp(Calling ISDN number ${EXTEN:1} on CAPI/7104370 from
${CALLERIDNUM})
exten = _0.,2,Dial,CAPI/7104370:b${EXTEN:1}|10|Tt
exten = _0.,3,Goto(s-${DIALSTATUS},1)
exten = _0.,103,NoOp(Calling ISDN number ${EXTEN:1} on CAPI/7104371)
exten = _0.,104,Dial,CAPI/7104371:b${EXTEN:1}|30|Tt
exten = _0.,105,Goto(s-${DIALSTATUS},1)
exten = _0.,205,Macro(outisbusy)
exten = s-NOANSWER,1,NoOp(NOANSWER - Setting dynamic autoroute for ISDN
number ${EXTEN:1} to local ext. ${CALLERIDNUM})
exten = s-NOANSWER,2,DBput(DYNAMIC/${EXTEN}=${CALLERIDNUM})
exten = s-NOANSWER,3,Congestion
exten = _s-.,1,Congestion
exten = _s-.,2,Macro(hangupcall)
Save the original extension to a variable like ...
exten = _0.,1,SetVar(myvar=${EXTEN})
Hi,
thanks for info. But another question arises - will this variable be unique
if for instance two or more calls happen in the same time ? I guess this is
more general Asterisk behaviour question...
Regards,
Rob.
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Re: [Asterisk-Users] Why ${EXTEN} variable changes after Goto ?
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, March 04, 2005 11:45 AM
Subject: Re: [Asterisk-Users] Why ${EXTEN} variable changes after Goto ?
On Thursday 03 March 2005 04:11 pm, Robert Rozman wrote:
Hi,
I'm trying to implement dynamic routing of incoming calls to local
extension if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand why ? How to
workaround on this one ?
exten = 42,1,SetVar(SAVED_EXTEN=${EXTEN})
exten = 42,2,Goto(marvin,27,1)
Hi,
thanks for help. I'd just like to be sure what happens if there is more than
one concurrent calls. Is variable set up for each of them or is necessary to
make variable that is somehow unique to each call ???
Regards,
Rob.
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[Asterisk-Users] Secure IAX Interasterisk authentication ?
Hi, I wonder if I can securely authenticate two Asterisk servers with IAX connection. I know for RSA authentication for IAX2 channel, but that seems to be meant for peer authentication... Has anyone done RSA (or any other secure way) authentication between two Asterisk servers ? Any example ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium BRI or quad BRI
- Original Message - From: Michael Bielicki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 26, 2005 5:01 PM Subject: Re: [Asterisk-Users] Digium BRI or quad BRI Hmm don't know about you but I rarely wait more than a couple of hours or answers from junghanns. We use junghanns cards in quite some setups and they work quite fine. Hi, can you reveal email address you use ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Beronet BN4S0 (quad BRI) card, echo cancel, zaptel timing, bristuff ...
Hi, I guess I'd need to run Beronet quad and octo bri cards under bristuff to get zaptel features (echo canceling, timing source) Am I right or could I achieve this also with chan_misdn - their native driver ? Running bristuff on Beronet cards is unsupported. Has anyone succesfully run Beronet quad BRI cards under bristuff recently ? Do they work ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium BRI or quad BRI
Hi, I had similar questions. I've emailed few questions and got no response in 10 days from junghanns. So I decided to try Beronet cards (they will arrive shortly). I just cannot imagine to have support from someone that is not able to answer few simple technical questions about their cards in one week. HTH, regards, Rob. - Original Message - From: Brett, Gary [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, February 23, 2005 2:49 PM Subject: RE: [Asterisk-Users] Digium BRI or quad BRI Thanks Florian, that's great, is this card (junghanns QuadBRI) really stable with * ? Do you or anybody else have any experiences with this card and also is it ok to run multiple cards in one machine cheers -Original Message- From: Florian Overkamp [mailto:[EMAIL PROTECTED] Sent: 23 February 2005 12:58 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Digium BRI or quad BRI Hi, -Original Message- Hi there, quick question...do digium make any BRI cards (ISDN2) or even better a quad port BRI, maybe im going blind, but I cant see any on their website They don't. If you are in need of a european ISDN2 type, see if http://www.junghanns.net/asterisk/page17.html helps you out. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What happens if quadbri or octobri loses power - do they have power failure feature ?
Hi, I mistakenly posted this to Dev list I wonder if I use quadbri or octobri cards to insert Asterisk between ISDN PBX and ISDN line - if power of Asterisks fails - will those card connect PBX directly to ISDN line ? If not are there any other simple switching devices, that would do this (in power fail it will connect ISDN PBX to ISDN lines directly) ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I exchange datas between two Asterisk servers ?
Hi, I'd like to establish way to exchange data between two remote Asterisk server. Something like call over IAX and send some structured data. Any advice ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
