Re: [asterisk-users] Advice on Asterisk Conference
On 22 Apr 2012, at 18:34, Steve Edwards wrote: On Sun, 22 Apr 2012, Stuart Elvish - IP Exchange Systems wrote: 1. DO I need a separate server for the conference server? This depends on a few factors: (a) You won't be able to run MySQL alongside Asterisk with conferencing and get good results. If you plan to use a single Asterisk server to do conferencing and other voice functions (for example voicemail) then I wouldn't expect any major issues. It depends on the usage of each voice function of the system. I don't think there was enough information in the OP's post to support the statement that running MySQL and Asterisk on the same box will not yield good results. I prefer to run them on separate boxes. Database servers and 'telco' servers have different resource requirements and seem to need different administration styles but they are not fundamentally incompatible. One of the amusing things about Oracle's XE 'free' database was that it was limited to a single core and 1Gb RAM. This made it perfect to run on the same box as asterisk. On a 4 core 4Gb machine you were pretty much certain that Asterisk was going to get 3 Cores and enough memory to itself, irrespective of what the DB users did :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on Asterisk Conference
Be aware that there are several different conferencing solutions for asterisk. I've used app_meetme in asterisk 1.8 (the LTS release) pretty happily. It is reasonably full featured and well supported. It has 2 drawbacks : 1) it needs a kernel module (Dahdi) to do the mixing and timing 2) it only does narrowband codecs If you like living nearer to the edge, there is app_confbridge in asterisk 1.10 (aka 10) which removes those 2 issues, but is new and so lacks some of the support (e.g. web modules etc) In terms of numbers, I've always found that user management issues kick in before the software/hardware limitations. Your milage may vary. T. On 20 Apr 2012, at 18:20, Mitchell Johnson wrote: We're looking into using Asterisk to do our conferencing. Currently we do all our conferencing using Cisco, we have a router with PVDM modules so we can offload the hardware resources. I'm looking for some best practices on how to set it up. 1. DO I need a separate server for the conference server? 2. Do I need to offload the actual conference to a router with PVDM modules. 3. Does anyone have experience with transitioning from Cisco conferencing to Asterisk? 4. How many participants can participate in a conference? Thanks, Mitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding degradation G711-iLBC
I'm quite fond of GSM610 as a low(ish) bandwidth codec - although it isn't as good as (say) speex or Silk, it is widely supported, and European users have had years of cellphone use to get used to the specific sound of a GSM call. So you can often go from a GSM610 supporting handset all the way through to a GSM supporting ITSP without needing to transcode at all. If at all possible avoid creating a path which involves 2 different lossy codecs - e.g. 729 _and_ GSM the results are significantly worse than either. If you can control all of the call path and have devices that support it, Silk is _lovely_ . It takes a bit of tuning for your expected network (which is unfortunately manual in Asterisk 10) but it is worth it. Tim. On 15 Apr 2012, at 12:15, Gustavo Garcia Bernardo wrote: Is it a good idea to use asterisk transcoding from G711 to iLBC or should I find out any other solution not involving transcoding (f.e. using G.729 that is supported in both sides). I'm worried about voice quality and trying to avoid paying for G.729 licensing. Anybody with experience or quantitative measurements of the voice quality degradation in that scenario? Regards, G Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
On 15 Jun 2011, at 23:29, Kevin P. Fleming wrote: On 06/15/2011 04:40 PM, Elliot Murdock wrote: Hello, Yes, the issue I am having is currently only with Google Talk. Wonder if what development will be made to fix this issue. At some point it will be fixed, and then Google will break it again. Google Talk/Google Voice connections to Asterisk will always be at the mercy of Google changing the protocol, which they do whenever they feel like it and with no warning. In other words, you better not be relying on it for critical communications, and you'll need to be patient when it breaks... because the developers can't just drop everything and fix it when Google changes the protocol. -- A quick (uneducated) look at the packet, I think google have added some jingle compatibility to gtalk. The packet invite now contains 2 nodes - one in the jingle namespace and one in the google/session namespace this confuses asterisk and it passes the call to _neither_ . I'm not up on iksemel - but I think that if it were told to match on either node, not just the first one things might work again The good news is that it supports a load of nice codecs now, including g722 :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
On 23 Jun 2011, at 13:44, randulo wrote: On Thu, Jun 23, 2011 at 1:58 PM, Tim Panton t...@westhawk.co.uk wrote: The good news is that it supports a load of nice codecs now, including g722 :-) And you know what that means? Unfortunately it means it doesn't work (yet). You should probably not mention the voipusersconfere...@gmail.com address this for week's VUC as at the moment the gateway ignores any calls to it. If/when it comes back to life, we can realistically expect wideband through to zipdx. T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium launches flying phone-phone
Huntsville 4/1/11 Digium inc today launched a new flying android desk phone. The new handset includes a small low noise helicopter to assist in call transfers. The device has speech and gesture recognition, so when you don't want to take a call you can either wave it away or say (Kevin will answer that). The flying handset has a megaphone and boom mic for priority call mode where it hovers just out of swatting range so you can't avoid the call. People wonder why we are doing this, but hey, I figured I'd try and combine as many my interests into one project and half the rest of the management are pilots too so it seemed like a good idea. Says Mark Spencer whilst dodging a cloud of his creations T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Scripting Language
Gosh, it depends what you want to do with asterisk. I've been having quite a lot of luck with groovy recently. You can easily wrap it around the (excellent) asterisk-java framework and have clean simple access to AMI and AGI interfaces. Alternatively look at adhearsion - which is a ruby framework for asterisk. But it _really_ does depend on what you are doing. T. On 1 Apr 2011, at 12:57, Gopalakrishnan A.N wrote: Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Scripting Language
I don't think that is quite true - Asterisk-java gives you access to AMI - which can be used to originate calls and to monitor call progress etc. You can even get RTCP call quality events. So I'm pretty sure you could use groovy and asterisk-java together to stress test your asterisk build. You would have to put some sample extensions and dialplan in place, but after that I figure you could do the rest in a nice modern scripting language. I wrote a blog post about groovy and asterisk a week or 2 ago : http://babyis60.wordpress.com/2011/03/14/the-gtalkskypesipirc-asynchronous-uc-mashup/ Tim. On 1 Apr 2011, at 15:59, Gopalakrishnan A.N wrote: Thanks. Asterisk-Java is a framework to build customer application. But my query here is, a testing script where to test a asterisk appliance or application, like stress testing, performance testing and etc. through some scripting language. For example SIPp has its own framework, where If the asterisk device is sending 100 message, SIPp is capable of recognizing that. In that way I am asking. On Fri, Apr 1, 2011 at 8:13 PM, Tim Panton t...@westhawk.co.uk wrote: Gosh, it depends what you want to do with asterisk. I've been having quite a lot of luck with groovy recently. You can easily wrap it around the (excellent) asterisk-java framework and have clean simple access to AMI and AGI interfaces. Alternatively look at adhearsion - which is a ruby framework for asterisk. But it _really_ does depend on what you are doing. T. On 1 Apr 2011, at 12:57, Gopalakrishnan A.N wrote: Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone (else) need an asynchronous asterisk event-action framework ?
I did a google talk,skype, SIP, asterisk, IRC async event driven voice/IM mashup
for the voip user's conference - (see http://wp.me/pgOOh-4a for a description)
I've ended up with a thing that could (with some work) be turned into an
asynchronous
asterisk event-action framework.
The basic premise is that you write filter terms to asynchronously match events
and actions to carry out when an
event matches - like this:
acts += new FilteredEventAction(DialEvent, // what sort of even we
want
{event - event.subEvent == Begin
event.dialString==200901@zipdxn-out}, // filter to just the events we care
about
{event - doSomethingZipDXSpecificHere( event.callerIdNum ) } // do
something
});
If I were to polish this up and make an open source framework of it would
anyone use it ?
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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[asterisk-users] Weird PRI error on an QUAD E1 span: Ring requested on unconfigured channel 255/255
After working fine for a week or so my new Quad E1 asterisk 1.8 system has started rejecting outbound calls from the Nortel BMC 450 it is connected to. The cli fills up with these: sig_pri.c: Ring requested on unconfigured channel 255/255 span 3 Is this likely to be a 1) config error 2) cable issue (I made them) 3) hardware problem with the Digium card 4) software (lib pri) Any clues? Anyone seen this recently (google shows it in 2005 but not since as far as I can see) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird PRI error on an QUAD E1 span: Ring requested on unconfigured channel 255/255
On 8 Mar 2011, at 15:48, Thorsten Göllner wrote: After working fine for a week or so my new Quad E1 asterisk 1.8 system has started rejecting outbound calls from the Nortel BMC 450 it is connected to. The cli fills up with these: sig_pri.c: Ring requested on unconfigured channel 255/255 span 3 Is this likely to be a 1) config error 2) cable issue (I made them) 3) hardware problem with the Digium card 4) software (lib pri) Any clues? Anyone seen this recently (google shows it in 2005 but not since as far as I can see) Which version of libpri? libpri version: 1.4.11.5 Asterisk01*CLI dahdi show version DAHDI Version: 2.4.0 Echo Canceller: Asterisk01*CLI core show version Asterisk 1.8.2.3 built by thp @ Asterisk01 on a x86_64 running Linux on 2011-02-07 14:32:26 UTC Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8.3 - IAX - echo - jitterbuffer
On 8 Mar 2011, at 02:12, sean darcy wrote: On 03/07/2011 05:26 PM, Kevin P. Fleming wrote: On 03/07/2011 04:15 PM, sean darcy wrote: I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On the office side, they hear an echo of _their_ speech, not mine. The office uses sip-providers generally without any echo problem. Where do I start to figure this out? How do I narrow it down? Can I figure out if it is an iaxagent problem? Could using jitterbuffer cause this? This is probably acoustic echo from your phone. The jitterbuffer has nothing to do with this. Yup. Turning down the volume on the call reduces the echo. Of course, now I can barely hear the office! I can keep the volume up on standard calls from the Droid X, which suggests that Android has some echo cancelling on phone calls. I'll try to see if the developer of iaxagent can do anything. BTW, if you haven't, try iaxagent on your phone. It's a very clever use of the iax protocol and leverages iax's strengths. iax makes a lot of sense on mobiles, dealing with the NAT issues from inconsistent access points easily. Thanks for the help. sean Anyone know how iaxagent is accessing the speaker/mic ? In theory the phone should have echo cancellation built-in, but it may only be enabled in certain cases. T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird PRI error on an QUAD E1 span: Ring requested on unconfigured channel 255/255
On 8 Mar 2011, at 16:24, Thorsten Göllner wrote: After working fine for a week or so my new Quad E1 asterisk 1.8 system has started rejecting outbound calls from the Nortel BMC 450 it is connected to. The cli fills up with these: sig_pri.c: Ring requested on unconfigured channel 255/255 span 3 Is this likely to be a 1) config error 2) cable issue (I made them) 3) hardware problem with the Digium card 4) software (lib pri) Any clues? Anyone seen this recently (google shows it in 2005 but not since as far as I can see) Which version of libpri? libpri version: 1.4.11.5 Asterisk01*CLI dahdi show version DAHDI Version: 2.4.0 Echo Canceller: Asterisk01*CLI core show version Asterisk 1.8.2.3 built by thp @ Asterisk01 on a x86_64 running Linux on 2011-02-07 14:32:26 UTC When the error occurs, maybe you can take a look at asterisk -rx dahdi show channels ?! Tricky, by the time I'd got there, they had bypassed the Asterisk and plugged the BMC straight into the BT E1's so all the channels will have been in red alarm. They did say that power-cycling the Asterisk machine cleared the problem for a while. The kernel log has some odd messages too: Mar 4 10:18:08 Asterisk01 kernel: [78444.361778] wct4xxp :07:08.0: Setting yellow alarm span 2 Mar 4 10:18:08 Asterisk01 kernel: [78444.361807] wct4xxp :07:08.0: RCLK source set to span 1 Mar 4 10:18:08 Asterisk01 kernel: [78444.361816] wct4xxp :07:08.0: Recovered timing mode, RCLK set to span 1 Mar 4 10:18:10 Asterisk01 kernel: [78446.092349] wct4xxp :07:08.0: Setting yellow alarm span 1 Mar 4 10:18:10 Asterisk01 kernel: [78446.092366] dahdi: Master changed to TE4/0/3 Mar 4 10:18:10 Asterisk01 kernel: [78446.092423] wct4xxp :07:08.0: RCLK source set to span 4 Mar 4 10:18:10 Asterisk01 kernel: [78446.092430] wct4xxp :07:08.0: Recovered timing mode, RCLK set to span 4 Mar 4 10:18:29 Asterisk01 kernel: [78465.586898] wct4xxp :07:08.0: Setting yellow alarm span 4 Mar 4 10:18:29 Asterisk01 kernel: [78465.586924] wct4xxp :07:08.0: RCLK source set to span 3 Mar 4 10:18:29 Asterisk01 kernel: [78465.586931] wct4xxp :07:08.0: System timing mode, RCLK set to span 3 Mar 4 10:18:31 Asterisk01 kernel: [78467.623700] wct4xxp :07:08.0: Setting yellow alarm span 3 Mar 4 10:18:31 Asterisk01 kernel: [78467.623767] wct4xxp :07:08.0: All spans in alarm : No validspan to source RCLK from Mar 4 10:18:31 Asterisk01 kernel: [78467.623781] wct4xxp :07:08.0: RCLK source set to span 1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Crossover cable for E1 ?
I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1 card. Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable? If so, any clues where I might buy one in the UK? The Digium card sellers don't seem to stock such a thing. Thanks. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Funky IAX behavior between 1.4 and 1.8
On 5 Nov 2010, at 15:04, Danny Nicholas wrote: Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. I can SIP into all 4 machines and life is great. When I try to IAX from the live machine to Machine 3, I get lags/pauses on Background/Playback commands. I play files and groups of files that last from 1-45 seconds, so I can press keys and proceed, but I don’t expect my end-users to know to do this. Any clues? Do I need to open a tracker issue on this one? Thanks Danny Nicholas -- There is an open bug on this - or something very like it - https://issues.asterisk.org/view.php?id=18110 The work around seems to be to set internal_timing = yes in asterisk.conf and noload = res_timing_dahdi.so ;noload = res_timing_pthread.so noload = res_timing_timerfd.so in modules.conf Which forces asterisk to use the (older less efficient/accurate) pthreads timer. The bug looks to be being worked on, so I'm optimistic it will be fixed soon. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird stalling of playback on IAX2 channels on 1.8 svn
I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird stalling of playback on IAX2 channels on 1.8svn
On 8 Oct 2010, at 15:37, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Friday, October 08, 2010 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Weird stalling of playback on IAX2 channels on 1.8svn I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk Are both Asterisk's 1.8? I had unhappy results doing IAX between 1.4 and 1.6 (1.8 is built on 1.6???) The far end is our voip supplier's asterisk - no Idea what version. But it also happens when talking to our Java IAX stack which isn't asterisk at all, so it isn't specific to a particular asterisk :-) What's more, if a call makes it past the announcement and gets bridged, it works fine. I've had several half hour calls through it. So it seems to me that it is an interaction between playback and iax2. T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird stalling of playback on IAX2 channels on 1.8 svn
On 8 Oct 2010, at 16:03, Bryant Zimmerman wrote: Tim I am actually seeing this on a 1.6.2.13 box as well. For some reason durring prompt playbacks they some times stall mid file. The call stays up but no audio comes in. Bryant From: Tim Panton t...@westhawk.co.uk Sent: Friday, October 08, 2010 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Weird stalling of playback on IAX2 channels on 1.8 svn I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Bryant, That's a relief, I thought it was just me ! Perhaps you can add something to https://bugs.digium.com/view.php?id=18110 T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 - Separate Signaling and Media?
On 24 Aug 2010, at 04:30, Tim Nelson wrote: - Tim Nelson tnel...@rockbochs.com wrote: Greetings all- Here's an odd question. Supposedly, IAX2 now has the ability to operate with signaling and media in separate streams, very much like SIP. I've read about this feature here[1] and there[2], but I have yet to see how to actually implement or test it. There are no options in the iax.conf sample configs with Asterisk. All suggestions welcome, except those telling me to jump off a bridge because separated signaling and media makes IAX pointless when compared to SIP. :-) Ugh, and let me specify references as originally intended: [1] http://tools.ietf.org/search/rfc5456 [2] http://www.voip-info.org/wiki/view/IAX+versus+SIP --Tim I think that the only implementation that does this is in asterisk. It only does it in a limited way by setting: transfer=mediaonly in iax.conf I've never tried it, but I'd be happy to co-operate on an experiment :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can't build resODBC on SUSE 11.3
What is menuselect actually looking for when it blocks me from selecting res_odbc ? I've got unixOdbc installed and working. I also have /usr/lib64/libltdl.so.3 - so I'm confused as it is claiming these are the pre-requisites ? How can I best track down what it _thinks_ is missing ? (This is on asterisk 1.8 svn trunk - but I don't think that is important, I think it is a package number issue) Thanks in advance, Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk + openBTS
Last time I looked, no OpenBTS does not (yet) support handoff between base stations during a call. Handoff between calls can be done using SIP registrations to a central asterisk. Tim. Sent from my iPhone On 23 Aug 2010, at 13:42, equis software equissoftw...@gmail.com wrote: Do you know if OpenBTS support handoff? Thanks On Fri, Aug 20, 2010 at 12:32 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Fri, Aug 20, 2010 at 10:41 AM, Tim Panton t...@westhawk.co.uk wrote: On 19 Aug 2010, at 20:59, Randy R wrote: On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) alansli...@gmail.com wrote: On 19/08/10 18:20, equis software wrote: I want to know about asterisk and openBTS This island runs it's GSM network on OpenBTS: http://www.niueisland.com/ This was the place he presented about. Read the blog here: http://openbts.sourceforge.net/NiuePilot/ and more about the installation here: http://vuc.me/2010/island-telephony-adventure/ I was part of the team that went to Niue to install OpenBTS, I'm happy to answer questions if you have them, although I'm not the radio guy - asterisk is more my thing :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk In all reality, Asterisk could be substituted with any other platform. All the magic happens in the USRP, OpenBTS, and the cellular phones. Asterisk is merely handling the routing and voice, same as it ever was. It is just the top of the stack. I have two USRPs and a handful of daughter boards, and yes I have two flex 800s that have been physically altered so they can also be flex 1800s with a simple command line. These are the boards you want for GSM (Cellular). There is also a project to be able to listen into phone calls (thanks to the French making encryption so weak) besides a ton of other applications that can be dreamed up. You can do passive radar, track people that have cell phones powered on, RFID (Free tolls anyone?), WiFi, heck, you can even kill people with certain types of pacemakers. While OpenBTS is cool and is on topic with Asterisk, read up on GNURadio and all the projects and applications you can come up with. It is really cool technology. Start here http://gnuradio.org/redmine/projects/show/gnuradio but you can easily find things like this http://tech.mit.edu/V128/N30/subway/Defcon_Presentation.pdf or come up with your own with a bit of imagination and skillz. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't build resODBC on SUSE 11.3
On 23 Aug 2010, at 18:07, Warren Selby wrote: On Mon, Aug 23, 2010 at 11:34 AM, Tim Panton t...@westhawk.co.uk wrote: What is menuselect actually looking for when it blocks me from selecting res_odbc ? I've got unixOdbc installed and working. I also have /usr/lib64/libltdl.so.3 - so I'm confused as it is claiming these are the pre-requisites ? How can I best track down what it _thinks_ is missing ? (This is on asterisk 1.8 svn trunk - but I don't think that is important, I think it is a package number issue) Thanks in advance, Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk You need to install the -devel packages of libtool-ltdl and unixODBC. Ah, libtool was what I was missing - thanks! Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk + openBTS
On 19 Aug 2010, at 20:59, Randy R wrote: On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) alansli...@gmail.com wrote: On 19/08/10 18:20, equis software wrote: I want to know about asterisk and openBTS This island runs it's GSM network on OpenBTS: http://www.niueisland.com/ This was the place he presented about. Read the blog here: http://openbts.sourceforge.net/NiuePilot/ and more about the installation here: http://vuc.me/2010/island-telephony-adventure/ I was part of the team that went to Niue to install OpenBTS, I'm happy to answer questions if you have them, although I'm not the radio guy - asterisk is more my thing :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to configure xorcom on Suse 11
On 29 Mar 2010, at 08:13, Tzafrir Cohen wrote: On Sun, Mar 28, 2010 at 09:16:48PM -0500, Tim Panton wrote: On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote: On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote: I'm having trouble getting a xorcom set up. A large part of the problem is that the box is a _long_ way away and I can't get to/at it easily, so while I could probably fix this in a few hours if the machine were in front of me, I'm struggling over a slow unreliable laggy link. Ok, enough whining from me. I have a new Xorcom plugged into the usb of a Suse 11 machine I built Dahdi from trunk (last thursday) # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools dahdihardware -v sees the box but no spans. Generally '/etc/init.d/dahdi start' . Or more specifically, 'dahdi_registration on' . See also: http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_installation_scenarios I've must be missing something here - this is what I see now. sh-4.0# dahdi_hardware -v usb:001/020 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware LABEL=[usb:X1037246] connect...@usb-:00:1a.7-4 Shouldn't I see spans ??? I think the box (I've never seen it, but I know what I asked for) has 8fxs+8fxo+2E1 . Yes, you should. Any relevant kernel messages? 2010-03-29T02:50:36.515445-11:00 pbx kernel: [467270.047734] FIRMWARE: : XPD=0-0 (0x0) OP=0x22 LEN=14 BYTES: 0E 00 22 00 02 02 08 00 09 00 04 00 00 00 2010-03-29T02:50:36.515511-11:00 pbx kernel: [467270.047734] NOTICE-xpp: XBUS-00(00): FIRMWARE ERROR_CODE: category=2 errorbits=0x02 (rate_limit=1669611216) 2010-03-29T02:50:36.515577-11:00 pbx kernel: [467270.047734] FIRMWARE: : XPD=0-0 (0x0) OP=0x22 LEN=14 BYTES: 0E 00 22 00 02 02 08 00 09 00 04 00 00 00 2010-03-29T02:50:36.515645-11:00 pbx kernel: [467270.047734] NOTICE-xpp: XBUS-00(00): FIRMWARE ERROR_CODE: category=2 errorbits=0x02 (rate_limit=1669611217) 2010-03-29T02:50:36.515711-11:00 pbx kernel: [467270.047734] FIRMWARE: : XPD=0-0 (0x0) OP=0x22 LEN=14 BYTES: 0E 00 22 00 02 02 08 00 09 00 04 00 00 00 2010-03-29T02:50:36.515789-11:00 pbx kernel: [467270.047734] NOTICE-xpp: XBUS-00(00): FIRMWARE ERROR_CODE: category=2 errorbits=0x02 (rate_limit=1669611218) 2010-03-29T02:50:36.515859-11:00 pbx kernel: [467270.047734] FIRMWARE: : XPD=0-0 (0x0) OP=0x22 LEN=14 BYTES: 0E 00 22 00 02 02 08 00 09 00 04 00 00 00 2010-03-29T02:50:36.515926-11:00 pbx kernel: [467270.047734] NOTICE-xpp: XBUS-00(00): FIRMWARE ERROR_CODE: category=2 errorbits=0x02 (rate_limit=1669611219) 2010-03-29T02:50:36.515993-11:00 pbx kernel: [467270.047734] FIRMWARE: : XPD=0-0 (0x0) OP=0x22 LEN=14 BYTES: 0E 00 22 00 02 02 08 00 09 00 04 00 00 00 2010-03-29T02:50:36.516060-11:00 pbx kernel: [467270.047734] If not: try: rmmod xpp_usb modprobe xpp_usb What new messages do you then see in /var/log/messages ? Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to configure xorcom on Suse 11
On 27 Mar 2010, at 21:48, JD Austin wrote: Xorcom hardware uses three layers; you must resolve issues in the following order: 1. USB 2. Dahdi 3. Asterisk I suspect you're having trouble with the usb layer. Run lsusb It will display a line like this if the firmware isn't loaded: Bus 001 Device 004: ID e4e4:1161 If it is e4e4:1162 then the firmware is loaded. You can manually load the firmware like this: /usr/share/dahdi/xpp_fxloader load or /usr/share/dahdi/xpp_fxloader usb It seems to load (some) usb firmware ok, as you can see from the syslog, but I suspect it is loading the wrong version. I got e4e4:1164 (I think - I've lost contact with the box for the moment). Thanks for the explanation too. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to configure xorcom on Suse 11
On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote: On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote: I'm having trouble getting a xorcom set up. A large part of the problem is that the box is a _long_ way away and I can't get to/at it easily, so while I could probably fix this in a few hours if the machine were in front of me, I'm struggling over a slow unreliable laggy link. Ok, enough whining from me. I have a new Xorcom plugged into the usb of a Suse 11 machine I built Dahdi from trunk (last thursday) # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools dahdihardware -v sees the box but no spans. Generally '/etc/init.d/dahdi start' . Or more specifically, 'dahdi_registration on' . See also: http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_installation_scenarios I've must be missing something here - this is what I see now. sh-4.0# dahdi_hardware -v usb:001/020 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware LABEL=[usb:X1037246] connect...@usb-:00:1a.7-4 Shouldn't I see spans ??? I think the box (I've never seen it, but I know what I asked for) has 8fxs+8fxo+2E1 . Thanks, Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trying to configure xorcom on Suse 11
I'm having trouble getting a xorcom set up. A large part of the problem is that the box is a _long_ way away and I can't get to/at it easily, so while I could probably fix this in a few hours if the machine were in front of me, I'm struggling over a slow unreliable laggy link. Ok, enough whining from me. I have a new Xorcom plugged into the usb of a Suse 11 machine I built Dahdi from trunk (last thursday) # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools dahdihardware -v sees the box but no spans. I get a syslog _full_ of this: 2010-03-25T21:35:22.338865-10:00 pbx kernel: [185556.006494] INFO-xpp_usb: XUSB: Xorcom LTD -- Astribank -- FPGA 2010-03-25T21:35:22.338878-10:00 pbx kernel: [185556.006695] INFO-xpp: XBUS-00: [usb:X1037246] Activating 2010-03-25T21:35:59.930296-10:00 pbx kernel: [185593.410290] INFO-xpp: XBUS-00: [usb:X1037246] Disconnecting 2010-03-25T21:35:59.930374-10:00 pbx kernel: [185593.410305] INFO-xpp: XBUS-00: [usb:X1037246] Deactivating 2010-03-25T21:35:59.930527-10:00 pbx kernel: [185593.410324] INFO-xpp: XBUS-00: [usb:X1037246] Release XPDS 2010-03-25T21:35:59.930696-10:00 pbx kernel: [185593.410423] INFO-xpp: XBUS-00: [usb:X1037246] Atribank Remove 2010-03-25T21:35:59.930776-10:00 pbx kernel: [185593.410455] INFO-xpp: XBUS-00: [usb:X1037246] Astribank Release 2010-03-25T21:35:59.930854-10:00 pbx kernel: [185593.410730] INFO-xpp_usb: xusb-0 (usb-:00:1a.7-4) [X1037246]: now disconnected 2010-03-25T21:36:14.944476-10:00 pbx kernel: [185608.613407] INFO-xpp: revision Unknown MAX_XPDS=64 (8*8) 2010-03-25T21:36:14.944488-10:00 pbx kernel: [185608.613422] INFO-xpp: FEATURE: without BRISTUFF support 2010-03-25T21:36:14.944493-10:00 pbx kernel: [185608.613436] INFO-xpp: FEATURE: with PROTOCOL_DEBUG 2010-03-25T21:36:14.944498-10:00 pbx kernel: [185608.613601] INFO-xpp: FEATURE: with sync_tick() from DAHDI 2010-03-25T21:36:14.946463-10:00 pbx kernel: [185608.615753] INFO-xpp_usb: revision Unknown 2010-03-25T21:36:15.162342-10:00 pbx kernel: [185608.831539] INFO-xpp_usb: XUSB: Xorcom LTD -- Astribank -- FPGA 2010-03-25T21:36:15.166858-10:00 pbx kernel: [185608.834398] INFO-xpp: XBUS-00: [usb:X1037246] Activating 2010-03-25T21:41:37.626186-10:00 pbx kernel: [185931.095914] INFO-xpp: XBUS-00: [usb:X1037246] Disconnecting 2010-03-25T21:41:37.626263-10:00 pbx kernel: [185931.095923] INFO-xpp: XBUS-00: [usb:X1037246] Deactivating 2010-03-25T21:41:37.626417-10:00 pbx kernel: [185931.095941] INFO-xpp: XBUS-00: [usb:X1037246] Release XPDS 2010-03-25T21:41:37.626579-10:00 pbx kernel: [185931.096024] INFO-xpp: XBUS-00: [usb:X1037246] Atribank Remove 2010-03-25T21:41:37.626659-10:00 pbx kernel: [185931.096126] INFO-xpp: XBUS-00: [usb:X1037246] Astribank Release 2010-03-25T21:41:37.626744-10:00 pbx kernel: [185931.096588] INFO-xpp_usb: xusb-0 (usb-:00:1a.7-4) [X1037246]: now disconnected 2010-03-25T21:41:47.445482-10:00 pbx kernel: [185941.114843] INFO-xpp: revision Unknown MAX_XPDS=64 (8*8) 2010-03-25T21:41:47.445492-10:00 pbx kernel: [185941.114855] INFO-xpp: FEATURE: without BRISTUFF support 2010-03-25T21:41:47.445497-10:00 pbx kernel: [185941.114868] INFO-xpp: FEATURE: with PROTOCOL_DEBUG 2010-03-25T21:41:47.445503-10:00 pbx kernel: [185941.115024] INFO-xpp: FEATURE: with sync_tick() from DAHDI 2010-03-25T21:41:47.447466-10:00 pbx kernel: [185941.117112] INFO-xpp_usb: revision Unknown 2010-03-25T21:41:47.665867-10:00 pbx kernel: [185941.333559] INFO-xpp_usb: XUSB: Xorcom LTD -- Astribank -- FPGA Any hints as to what I'm doing wrong would be much appreciated. (here's some project background for the curious http://babyis60.wordpress.com/2010/02/25/the-island-phone-system-adventure/ ) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk
]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) -- Skype/rexesbposolutions-084159e8 Playing 'queue' (language 'en') [Dec 29 17:27:43] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) [Dec 29 17:27:43] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore format back to 2 == Spawn extension (sky, s, 2) exited non-zero on 'Skype/rexesbposolutions-084159e8' [Dec 29 17:27:43] NOTICE[4997]: core.cpp:2133 sfa_call_hangup: ending call Kindly resolve this issue ASAP. With Regards Vijay Goyal (Software Engineer VOIP) Alliance Infotech Private Limited - Mobility,Convenience,Realization (An ISO 9001: 2000 certified company) B 254 First Floor, Okhla Industrial Area-I, New Delhi 110 020 (India) | Tel: +91 11 40517731, 2637 1851 | Fax: +91 11 2637 1852, 2981 0953 Digium Select Partner | Dialogic Partner | Microsoft Certified PartnerCRM Computer Telephony solutions | Speech Enabled IVRS | Unified Communications | Voice loggers | Audio Conferencing | Web Enabled solutions ___ -- 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 It looks to me as if you are running out of 729 licenses. A single call may (sometimes) need more than one license. You can probably avoid this problem by either: 1) buying more 729 licenses (just a few more than active channels should do) 2) using Ulaw in chan_skype (instead of 729) 3) downloading the soundfiles in 729 (you currently only have GSM) Do 3) anyway - gsm transcoded to 729 always sounds horrible. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] GSM and Wav format
On 2 Nov 2009, at 12:11, ABBAS SHAKEEL wrote: Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these recorded files on WEB. We have following options 1. Record in GSM and send to central Server. Which will convert to it to WAV format using some code / any other thing. The issue in this is that CPU will get very busy in this case. Because GSM Files can be very frequent. 2. Recored in Wav and send to central server. In this case we may face Network Bandwidth problem.(Even we create VPN). QUESTION IS: Is there any other format in which we can record using the record application provided its is small in size and directly playable on WEB. Directly playable is a complex question - what are you assuming your web users have? Browser only: HTML 5 browsers mostly support Oggvobis natively Browser+quicktime: gsm,mp3,wav etc Browser+flash: MP3 (and perhaps speex) Browser+java: Pretty much any format you like I wrote (and opensourced) a little java applet that plays .gsm files see http://www.westhawk.co.uk/software/playGSM/PlayGSM.html Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] IAX2 order
On 18 Sep 2009, at 19:26, David A. Bandel wrote: Folks, I've been fighting with this seems like forever now and can't make it work in 1.6.x. In 1.4.x, I could make sure a particular voip provider was always first in the list by making him as an #include and putting it last. Now in 1.6.x I can never get this to take. I really don't want to run TWO asterisk servers just for some IAX trunking. Real problem: my voip provider doesn't send username authentication, just their secret. If they are either the first or the only, then all works because * checks just the first iax2 entry in the show iax2 peers list. If they are anything but first, all my incoming calls from them fail (identification failure). Is there ANY way to make them first? In 1.6.x they are always 3rd. TIA, Do you have an IAX packet trace of that exchange ? It seems _really_ unlikely - and possibly a protocol violation - for them to be sending a secret without a username. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] AST-2009-006: IAX2 Call Number Resource Exhaustion
Is there a version of this patch for 1.6.2 - or did the recent 1.6.2 rc1 drop include it ? Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] G.722 problems with IAX
On 4 Sep 2009, at 07:53, Armin Schindler wrote: On Thu, 3 Sep 2009, Tilghman Lesher wrote: On Thursday 03 September 2009 02:47:05 Armin Schindler wrote: Hello, I try to move our asterisk installation (3 Asterisk servers in different offices connected using IAX and a lot of SIP phones, as well as ISDN connections using CAPI) to use G.722 instead of G.711. Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one, which solves the gain problem). So SIP-to-SIP and to ISDN there is no problem. G.722 itself works and transconding to G.711 for ISDN also works good. But when I make a connection through IAX to another asterisk (having allow=g722 to really use G.722 in IAX) the voice is 'broken'. I also work on G.722 for twinklephone and encountered a special thing about G.722: It has a sample rate of 16000, but it announced as 8000 in SDP. Since I have similar problem with my G.722-twinkle implementation, it looks like the RTP and/or jitterbuffer code has a problem with that. Did I miss something here or is this really a bug? You missed that the IETF has a typo in the specification, stating that G.722 is to be stated as 8000, even though it's 16000. This will remain, due to backwards compatibility concerns. Please see RFC 3551, section 4.5.2. http://www.apps.ietf.org/rfc/rfc3551.html#sec-4.5.2 No, I didn't miss that. See my text. I mentioned this because I think this might be the reason of the problem and the incorrect handling in jitterbuffer, if it is the jitterbuffer. It is just a guess, since everything else seems to work good. The question is why does G.722 via IAX has problems. Is anyone using it and can say it works in his setup? Armin I've got g722 running through 1.4.22.2 with the patch set that targets 1.4.7 Calls from our java iax softphone come in as IAX2 in g722 and leave via SIP to a g722 conference service. seems to work ok. No transcoding, recording etc, and the jitterbuffer is _off_ since it's a VoIP to VoIP call. (a few folks used it on the VUC conference this afternoon - anyone have problems ?). Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Different codecs for reading and writing
On 1 Aug 2009, at 22:26, Alex Balashov wrote: Elliot Murdock wrote: Thank you...do you know if IAX can do this? The reason for doing is this is to get over the adsl upload/download discrepancy. While G711 gives terrific quality, it is not always that feasible for the upload direction, which has much more limited bandwidth. Accordingly, it would be possible to use G729 for upload, but keep the higher quality codec, G711, for download. I do not know a great deal about IAX so I will defer to the experts for the definitive word on whether it is possible from the point of view of its formal protocol mechanics. However, poking around the various configuration options for IAX peers on voip-info.org and a few other places suggests that there is no option to do that with IAX, either. It's not really something 99.9% of VoIP users want to do. :-) I think you will find that it may work with Asterisk's IAX implementation. The protocol expects the 2 ends to agree a single symmetrical codec as part of the connection setup, but it doesn't define what actually happens if the codec specified in the first (full frame) voice packet isn't what was agreed. I have a vague memory that if the codec is one that is allowed, asterisk does 'the right thing' issues a warning and uses what it was given. But, as Alex says, there is no clear way to define this in the config files. You would probably do better to use Speex in both directions, but set the encoding quality (in codec.conf ) parameters to be different at the 2 ends. The speex decoder should at the far end should be fine with that. see http://www.voip-info.org/wiki/view/Asterisk+config+codecs.conf Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Skype for Asterisk: Public Beta available
I had that too, I cured it by kill -9 'ing the skypeforasterisk process that was left over from the previous version of the beta. Hope that helps. Tim. On 2 Aug 2009, at 11:20, Emrah wrote: I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Skype for Asterisk: Public Beta available
I don't know then. My understanding is that the message is caused by the wrong skypeforasterisk process running. - did you (ever) run it as a different user ? If it is a test box, you could try a full reboot. Tim. On 2 Aug 2009, at 19:35, Emrah wrote: Hi Tim, I don't have any skypeforasterisk process running. I tried to killall -9 asterisk but it did not solve my issue. Any other suggestions? Thanks for your help, Emrah Tim Panton wrote: I had that too, I cured it by kill -9 'ing the skypeforasterisk process that was left over from the previous version of the beta. Hope that helps. Tim. On 2 Aug 2009, at 11:20, Emrah wrote: I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Dialplan strategy suggestions needed
On 1 Aug 2009, at 08:32, Myles Wakeham wrote:
I have a new Asterisk system going into production next week and I'm a
bit stumped as to the best way to handle the Dialplans for it.
The Asterisk system is replacing 4 separate PSTN lines with both SIP
PSTN inputs. The setting up of the dial plan is giving me some design
headaches, which probably means I'm missing something obvious and
doing
this the hard way.
I have separate entire phone 'systems' for each incoming DID. For
example, this one system will handle 4 separate incoming DIDs. The
first (let's call it Company A) takes a call, plays a menu, gets input
from the caller, directs them to the appropriate extension (ie.
Press 1
for sales, 2 for support, etc.).
The second DID has a similar setup with a menu, but its for an
entirely
different company. The incoming extension number I'm getting for this
will be entirely different, therefore will be the definition of this
phone menu structure.
The others are simply numbers that go through a 'Time of Day' check
and
then simply forward to an extension.
I have it all working fine by using:
extern = 212555,1,.
extern = 212555,2,.
etc. for the first number, and then a second set for the second
number like:
extern = 2125551112,1,.
extern = 2125551112,2,.
The problem is when I get to the Background() command to play the
sound
file and get the input from the user for the menu. Since the input
from
the user becomes the extension, I then have a problem that my
multi-faceted dialplan now gets confused with what extension applies
to
what menu, etc.
I need to be able separate these into their own sections so that
extensions won't conflict but I'm not sure how to do this. All the
calls are coming in from one SIP provider, so I have only one context
that I'm using because of that. I'm not sure if there is a way to
create separate contexts for this and branch to them?
Sure, have a top level context that inbound calls from the ITSP go into:
[from-ITSP]
exten = 2125551112,1,Goto(companya,${EXTEN},1)
exten = 212555,1,Goto(companyb,${EXTEN},1)
; then separate contexts for each company:
[companya]
extern = 212555,1,.
extern = 212555,2,.
[companyb]
extern = 2125551112,1,.
extern = 2125551112,2,.
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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Re: [asterisk-users] Some IAX calls do not disconnect.
On 7 Jul 2009, at 05:05, Steve Totaro wrote: On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelsontnel...@rockbochs.com wrote: - Steve Totaro stot...@asteriskhelpdesk.com wrote: Just use SIP and solve all your problems. I seem to be noticing a common element to your posts about IAX. :-) I've been successfully using IAX in a large scale environment with no problems... yet. Can you shed some light on the reasoning behind your obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a usability standpoint (NAT traversal is quick to my mind...). BUT, is it just not robust enough in your experience? Are there inherent problems with the protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the implementation within Asterisk that is the problem? I'm very interested to to know where your disdain comes from. :-) Thanks Steve! --Tim First define large scale. It certainly means different things to different people. Second, It comes from huge amounts of audio problems over many, many years, and many, many implementations. I actually don't have a disdain for it, it has made me a good deal of money by fixing ITSPs/carrier's audio issues by switching them to SIP and still does so I have a fondness for it. Keep up the sub par protocol, it helps with the balance sheet! Third, it will never kill SIP. First of all, Digium owns the name and we have seen what they are willing to do to attack people for trademark or copyright infringement (think about the Google Adwords debacle and the the Open letter to Digium drafted by Trixter that I am not sure was ever fully addressed by Digium.) It would have to be renamed or something. I think the same thing of DAHDI. They want control over the the names Inter Asterisk Exchange and Digium (whatever the heck the rest of it means.) Second, SIP is the industry standard. Only a couple of goofy phones do IAX2 as far as I know, some crappy handsets I wouldn't even bother testing if offered as a free demo unit. SNOM might now, I am not sure but I think I read interest in it or it was actually accomplished. SNOM is OK but I was never a big fan. When I see it on a Polycom, Cisco, NEC, 3Com, or any other major vendor's phones or platforms, then I may rethink my ideas. If 3Com and Digium are partnered up now, how come the NBX for V3000 doesn't support IAX2? They do have SIP. Second, there are work arounds for just about every downfall of SIP, like NAT traversal and the like. Third, ALL REAL TIME VOICE traffic is on a single port. There is a big issue there, I won't elaborate, but just think about it. SIP is here to stay until some other protocol comes about, but certainly not IAX2. It will be along the evolution of H323 to SIP to X., but not IAX,lol. Do you realize that most providers are dropping IAX2 support, even IAX.cc recommends SIP, gotta wonder why? Maybe it is all good now, but I won't bank my reputation on it. I use what I know works well, period. Even unnamed Digium Employees have poo pooed IAX2, albeit a year or two ago. It looks good on paper, didn't perform well historically, and now just like anything that I have lost trust in, it has to earn my trust back and that is not easy. -- Obviously Steve and I don't agree about this. There are places where IAX can go that SIP just can't. When Steve says just use SIP, what he is actually recommending is to use SIP/STUN/SDP/RTP/IPSEC to get the same result. (at a 50% bandwidth overhead) i.e. replace a single 100 page RFC with something like 100 RFCs :-) In a big organization where you control the network infrastructure, that is an entirely viable solution, but when you want to get calls through a messy network without having to fill out an infinite number of change requests to the firewall team you should consider IAX. The mess that SIP makes is reflected in the number of bugs and the code size. I'm currently working with a SIP stack that is about 10x the size of the comparable IAX codebase, which matters in some environments. As to the 'everything over a single port' issue, this is no longer such a big deal. (And it is exactly this feature which provides IAX's firewall penetration) Most modern Linuxes support multiple threads reading datagrams from a single datagram socket. The current IAX implementation in Asterisk doesn't support it, but that's an implementation issue, not the protocol itself. Also IAX now supports redirecting the media - which could be used to send it to a separate port on the same box. Various Digium employees have also badmouthed SIP (I think we all have after a bad day at the SDP coalface), so you can't take such remarks too seriously. I overheard a senior Cisco employee saying So you were right all along about IAX to a very senior Digium employee, which also proves nothing much :-) Competition is a good thing - even amongst protocols. T. Tim Panton - Web/VoIP consultant and implementor
Re: [asterisk-users] Some IAX calls do not disconnect.
Ah, and you are using iax trunking - which depends on the realtime clock. I'm no expert on virtualization, but I think I read that the usb based zaptel clock was a better choice in a virtualized system. T. On 6 Jul 2009, at 06:44, Rajkumar S wrote: Hi, The servers B C are running in a virtual machine (linux kvm) and uses ztdummy for timing. Server A has a digium card. I am not sure if this is the cause of the problems I am facing. raj On Fri, Jul 3, 2009 at 5:57 PM, Rajkumar Srajkum...@gmail.com wrote: On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: I'd try adding transfer=no in the B iax.conf This does not help, I still have some ghost calls in B a16-in1*CLI core show channels Channel Location State Application(Data) IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-12174 outbo...@inbound-cal Up Dial(iax2/a16-in1- sangoma-flip IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-7161outbo...@inbound-cal Up Dial(iax2/a16-in1- sangoma-flip IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-14813 s...@queue:20 Up Dial(iax2/a16-in1- a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-4485s...@queue:20 Up Dial(iax2/a16-in1- a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-10115 s...@queue:20 Up Dial(iax2/a16-in1- a16-q1/queue 10 active channels 5 active calls raj ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Some IAX calls do not disconnect.
On 3 Jul 2009, at 07:18, Rajkumar S wrote: Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP clients connect. SIP clients can also dial outside and call goes like C - B - A - PSTN. Every day evening I find that there are about 30 calls in B which is not disconnected. This comprise of both calls from B - A as well as B - C. There are no such lingering calls in A or C. Every day I manually disconnect the calls, shown below are two example with first one from B - C and second B - A. a16-in1*CLI soft hangup IAX2/a16-in1-11080 Requested Hangup on channel 'IAX2/a16-in1-11080' -- Hungup 'IAX2/a16-in1-a16-q1-16420' == Spawn extension (queue, s, 20) exited non-zero on 'IAX2/a16- in1-11080' -- Hungup 'IAX2/a16-in1-11080' a16-in1*CLI soft hangup IAX2/a16-in1-903 Requested Hangup on channel 'IAX2/a16-in1-903' -- Hungup 'IAX2/a16-in1-sangoma-flip-outgoing-16393' == Spawn extension (inbound-calls, outbound, 1) exited non-zero on 'IAX2/a16-in1-903' -- Hungup 'IAX2/a16-in1-903' in iax.conf of B the entries are like: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no [a16-in1-a16-q1] type=peer host=192.168.79.176 auth=plaintext secret=password username=a16-q1 qualify=yes trunk=yes in C the corresponding entry is: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no [a16-q1] type=user auth=plaintext secret=password context=inbound-calls qualify=yes trunk=yes I do not know where even to start. Any idea to resolve this would be much appreciated. raj I'd try adding transfer=no in the B iax.conf I'm guessing the box in the middle (B) is somehow transferring itself out of the call but retaining a ghost call entry. It would be interesting to know what state those ghost calls are in - iax2 show netstats on the CLI might tell you something interesting. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Skype for Asterisk. Any return of experience ?
On 27 Jun 2009, at 10:06, Olivier wrote: Hi, As many remember, almost one year this Skype for Asterisk extension program was announced. Has anyone tried it ? Is there any available pricelist ? I've just had a talk on Skype for Asterisk accepted at Astricon (www.astricon.net ), so if you can wait that long, you come along and I'll try and tell you what SFA can do. In the meanwhile - it often crops up on the voipusers conference (www.vuc.me ) on a Friday. In fact I've been running an experiment allowing people to call the conference from Skype (using SFA of course). Feel free to call in and try it this Friday. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] IAX for internet file transfer?
On 27 Jun 2009, at 11:27, Maris wrote: guarantee delivery?, not to mention that IAX2 does not use RTP. Are you suggesting to change the protocol to support such transfers? When it makes sense, yes - see below, otherwise the idea can get into the waste paper backet. ... But why does he want to do it ? Share secret / illegal files LOL ? Transfer files and/or logging data to/from computers anywhere in the intranet of organizations - over the internet. Due to restrictions this computer may not have server functionality. For the purpose, an IAX client can be installed on the remote computer. Of course, such client-client communication can be solved using an intermediate server which two clients that exchange data connects to. The specific features of IAX (NAT transparency) could help, provided that simple TCP channels initiated by the clients can posess problems in establishing connections under certain weird network constellations - it goes beyond my knowledge to judge that. ... to the other side and decode it there Asterisk (or just about any VoIP software) will opt for timely delivery rather than a reliable delivery. Encoding digital data into audio in order to transfer it as digital audio data packets makes no sense for me. Packet problems can be overcome with other methods, as pointed out by other contributors. Rob Maris Hardware developer You should read the protocol spec. http://www.rfc-editor.org/authors/rfc5456.txt It already supports a couple of 'data' transports, including the one that was used to upgrade the IAXy firmware. I don't think you would have to change much (if anything) in the protocol to make it work. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] IAX2 trunking with Older Asterisk, version ?
Given that he is using plaintext as the auth method, I guess anyone
who wants that
password can have it by snooping anyhow. :-)
T.
On 1 Jun 2009, at 07:18, Rob Hillis wrote:
The clue in the log is no authority found. Something in the
configuration at the other end doesn't match the configuration at this
end - almost certainly the username and password.
Why are you including the IP address when dialling the trunk? If your
peers are set up with IP addresses (which they are) it should not be
necessary.
By the way, it's a *very* bad idea to post passwords in a public
forum.
Tharanga wrote:
my sip phone registered on 1.6, when i dial 4567 from 1.6 version,
it wont go to 1.6 voice mail. it says
== Using SIP RTP CoS mark 5
-- Executing [4...@sip:1] Dial(SIP/312-09f9a720, IAX2/trun...@147.120.203.98
/4567,10,t) in new stack
-- Called trun...@147.120.203.98/4567
[Jun 1 11:01:18] WARNING[8178]: chan_iax2.c:8991 socket_process:
Call rejected by 147.120.203.98: No authority found
-- Hungup 'IAX2/trunk14-9738'
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/312-09f9a720' status is
'CHANUNAVAIL'
[trunk14]
type=friend
host=147.120.203.98
auth=plaintext
secret=XX
context=sip,sip2,sip3
;keyrotate=off
permit=0.0.0.0/0.0.0.0
1.6 EXTENSIONS.CONF
[globals]
TRUNKIAX14=IAX2/trun...@147.120.203.98
[sip]
;exten = 4567,1,Dial(${TRUNKIAX14}/${EXTEN}|10|t)
exten = 4567,1,Voicemail(${EXTEN},u)
~
1.2 EXTENSIONS.CONF
[Jun 1 05:20:31] NOTICE[9536]: chan_iax2.c:8782 socket_process:
Rejected connect attempt from 147.120.203.71, who was trying to
reach '4567@
[trunk14]
type=friend
host=147.120.203.71
auth=plaintext
secret=Mah
context=sip,sip2,sip3
;keyrotate=off
permit=0.0.0.0/0.0.0.0
[globals]
TRUNKIAX14=IAX2/trun...@147.120.203.71
[sip]
exten = s,1,wait(1) ; Answer the line
exten = s,n,BackGround(demo-congrats)
exten = s,n,ResponseTimeout,5
exten = s,n,Dial(SIP/${EXTEN},20,t)
;exten = s,n,BackGround(goodbye)
exten = s,n,Hangup
exten = 4567,1,Dial(${TRUNKIAX14}/${EXTEN},10,t)
Asterisk versions may differ. I do IAX trunk successfully even
between Asterisk 1.0.2 and 1.4.xx
please post your Dial command.
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Re: [asterisk-users] playing media(moh,prompts) from flash player
We did an opensource Java Applet that plays GSM files _very_ simply if that helps. I'd accidentally removed it from our website, but it is back now - improved with a javascript interface supporting load, play and pause actions. http://www.westhawk.co.uk/software/playGSM/PlayGSM.html The demo is pretty basic, but that at least keeps the html simple :-) Tim. On 21 May 2009, at 12:41, marek cervenka wrote: hi, i'm searching solution for playing media(moh,prompts,voicemail,recordings - wav format) from adobe flash player (web browser) flash cannot play wav directly (imho) i must convert files to any other format on-the-fly - i cannot use mp3 because of royalties - next option is swf (with ffmpeg), supported free audio codecs (http://en.wikipedia.org/wiki/Flash_Video#Format_details) * uncompressed PCM * ADPCM * AAC can you someone recommend solution/combination which works? tnx --- Marek Cervenka === ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Rusting Snoms?
Christian, thanks, I'd never run pcap in a phone before - cool. The trace shows jitter - but in a weird way. some of the packets have delta's of 20 ms but always a multiple of 10 so 50 and 30 occur, as do 10 and 0. Is that normal ? Tim. On 9 May 2009, at 11:04, Christian Stredicke wrote: Because the phone is a digital system, I would suspect that it is a problem with the switch. Run a quick PCAP trace to see where the jitter comes from. Depending on the firmware version, you can do that from the web interface. CS -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com ] Im Auftrag von Tim Panton Gesendet: Samstag, 9. Mai 2009 11:46 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Rusting Snoms? This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of them didn't work well. The symptom is that outgoing RTP audio is garbled - like the packets are pulsed. Inbound is fine. This isn't true for all of the phones, just some of them. (The all run the same SNOM firmware) To be fair, they are on a new network, so it could be the cables or new 1Gb switches, except that the problem moves with the phone if you relocate it from one desk to another. I've tried a fresh asterisk install, but that didn't help either. So I am forced to conclude that something went 'bad' in those (old) phones while they were switched off. Has anyone got any clues for me? Thanks! Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Rusting Snoms?
On further investigation - it may well be that the switch doesn't like the phones (or vice-versa) I tried daisy-chaining one phone off the second port of the other and got distinctly better audio. It's a new netgear fvs 318 with autosensing 100/10 ports. Any clues ? Thanks. Tim On 9 May 2009, at 11:04, Christian Stredicke wrote: Because the phone is a digital system, I would suspect that it is a problem with the switch. Run a quick PCAP trace to see where the jitter comes from. Depending on the firmware version, you can do that from the web interface. CS -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com ] Im Auftrag von Tim Panton Gesendet: Samstag, 9. Mai 2009 11:46 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Rusting Snoms? This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of them didn't work well. The symptom is that outgoing RTP audio is garbled - like the packets are pulsed. Inbound is fine. This isn't true for all of the phones, just some of them. (The all run the same SNOM firmware) To be fair, they are on a new network, so it could be the cables or new 1Gb switches, except that the problem moves with the phone if you relocate it from one desk to another. I've tried a fresh asterisk install, but that didn't help either. So I am forced to conclude that something went 'bad' in those (old) phones while they were switched off. Has anyone got any clues for me? Thanks! Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Rusting Snoms?
It isn't POE - its using the original power brick that came with the phone. Swapping to a dumb hub (without uplink-autosensing) seems to fix it. Would a firmware upgrade (from 3,56m 6154) help ? Tim. On 19 May 2009, at 13:28, Christian Stredicke wrote: With cheap PoE devices Ethernet can easily get on the edge - or over the edge. If you have another switch/different model, a quick try will help isolating the problem. CS -Ursprüngliche Nachricht- Von: Tim Panton [mailto:t...@westhawk.co.uk] Gesendet: Dienstag, 19. Mai 2009 13:46 An: Asterisk Users Mailing List - Non-Commercial Discussion; Christian Stredicke Betreff: Re: [asterisk-users] Rusting Snoms? On further investigation - it may well be that the switch doesn't like the phones (or vice-versa) I tried daisy-chaining one phone off the second port of the other and got distinctly better audio. It's a new netgear fvs 318 with autosensing 100/10 ports. Any clues ? Thanks. Tim On 9 May 2009, at 11:04, Christian Stredicke wrote: Because the phone is a digital system, I would suspect that it is a problem with the switch. Run a quick PCAP trace to see where the jitter comes from. Depending on the firmware version, you can do that from the web interface. CS -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com ] Im Auftrag von Tim Panton Gesendet: Samstag, 9. Mai 2009 11:46 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Rusting Snoms? This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of them didn't work well. The symptom is that outgoing RTP audio is garbled - like the packets are pulsed. Inbound is fine. This isn't true for all of the phones, just some of them. (The all run the same SNOM firmware) To be fair, they are on a new network, so it could be the cables or new 1Gb switches, except that the problem moves with the phone if you relocate it from one desk to another. I've tried a fresh asterisk install, but that didn't help either. So I am forced to conclude that something went 'bad' in those (old) phones while they were switched off. Has anyone got any clues for me? Thanks! Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Rusting Snoms?
This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of them didn't work well. The symptom is that outgoing RTP audio is garbled - like the packets are pulsed. Inbound is fine. This isn't true for all of the phones, just some of them. (The all run the same SNOM firmware) To be fair, they are on a new network, so it could be the cables or new 1Gb switches, except that the problem moves with the phone if you relocate it from one desk to another. I've tried a fresh asterisk install, but that didn't help either. So I am forced to conclude that something went 'bad' in those (old) phones while they were switched off. Has anyone got any clues for me? Thanks! Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Asterisk and ODBC
On 2 May 2009, at 19:30, Vela Sivasankaran wrote: Hi, I am using a 64-bit RHEL 5 machine. I built Asterisk latest 1.6 branch. The system has ODBC and Postgres installed. psql, isql and odbc work fine. Asterisk make menuselect for some reason does not see the installed packages and refuses to build res_odbc and other packages. How do I force it to do that? Is there a way to modify the output file from menuselect and make it build these modules? Can I donwload these modules from anywhere if it is not possible from menuselect? Is there a sample Asterisk 1.6 full build 64-bit RHEL rpm available anywhere? Make sure you have the odbc headers installed (is that unix-odbc- devel ?) Then re-run ./configure before you do a make menuselsect Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Compact, fanless appliance?
On 26 Apr 2009, at 09:17, Paul Chambers wrote: Vincent wrote: www.voip-info.org/wiki/view/Asterisk+embedded+systems Thanks Steve. I knew about this list, but I wanted to make sure there weren't other, more complete sources about the subject. So at this point, it seems like it boils down to this: Soekris PCEngines Atcom (IP01: £160+VAT) Herologic (HL-463: $259) uCpbx (235.00 EUR) AstBoxes (168.00 EUR) Gumstix HP Thinclient t5720 Thank you. EdgePBX is another option - they offer two, eight and twelve port Astfin (Asterisk on Blackfin) boxes, similar to those from uCpbx Atcom. The FX02 and FX08 have SD slots, the FX12 has a CF slot. None have PCI slots or USB, just Ethernet. The FX02 (two port) is $150, plus $25 per module, though any compatible TDM400-style module will work. I have an FX08 with Digium and NetX86 modules in it. It's been solid for me (just a customer, no connection to the company). I'm running asterisk 1.4 on an NSLU2 , only a couple of channels and minimal transcoding, but it seems fine and stable. £80 + usb storage I built 1.4 from sources on the NSLU2 which took a while :-) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Skype for SIP
On 23 Mar 2009, at 19:42, Gordon Henderson wrote: Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon There are fewer limitations to SFA than SFS. SFA gets presence and full user info, plus it can make calls to Skype users, which SFS cant. I'm hoping that Digium will extend this difference by adding support for text and perhaps video... Here's an example of something SFS can't do: sfa.westhawk.co.uk/skype/call.xsql?key=echo123 (a quick demo I knocked up with the SFA beta) Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] usb-phones
On 24 Mar 2009, at 09:52, Gordon Henderson wrote: On Mon, 23 Mar 2009, Hans Witvliet wrote: While reading the thread about recommending usb-phones... Once in a while, i'm in a data-centre, no normal phones, and too much concrete shielding wireless phones. So i was thinking to use one of those usb-phones, and plug it into one of my servers there. But what i read from the thread, i seems that you need a graphical environment, while all of the servers are strictly cli-only. Is there a cli-based phone (besides the asterisk-console), that can use a usb-audio-device? Afaicr,those usb-phones present themselves as an plain usb-audio device. I asked here a while back about a command-lime VoIP client. Got no- where interesting - other than people suggesting I run a full-blown asterisk, and alas I've not had time to do it myself. It *should* be relatively straightforward using the existing IAX libraries, I'd have thought, however... Gordon It would be pretty easy to take the Mexuar Corraleta Java IAX source and make a commandline Jar from it. Such a jar would work on Linux, Mac and Windows. T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] url in dial command: how does it work?
Use IAX :-) In principle chan_skype could also support it. T. On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: Hi, Does anybody knows where I can find some docs about how to make the URL parameter inside the Dial command work? I tried to make some tests with a sip phone without success: the sip debug shows no URL inside sip packets. :( Any hint appreciated. :) Thank you Giorgio ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] url in dial command: how does it work?
Oh sorry, I wasn't clear. The IAX protocol has a frame type for sending this URL info. Skype has an attribute for it. The intention is (I think) to be able to forward the URL for the customer (in the corporate CRM system) to the agent answering a call on a softphone. Some of the IAX softphones support this. What were you planning to do with it. Tim. On 16 Mar 2009, at 13:04, Giorgio Incantalupo wrote: Hi Tim, ok, but I think the big question is...what is the URL for? It seems I need a special device...but which? What kind of device do you use? Thanks. Giorgio Tim Panton wrote: Use IAX :-) In principle chan_skype could also support it. T. On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: Hi, Does anybody knows where I can find some docs about how to make the URL parameter inside the Dial command work? I tried to make some tests with a sip phone without success: the sip debug shows no URL inside sip packets. :( Any hint appreciated. :) Thank you Giorgio ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] asterisk and ericsson e1 connection how to??
You should be able to get support from the people who sold you the card. You need to configure 2 files (I'm looking at an old system, so they have the zaptel style names). My files are below - the thing to note is the span 1,1,0, the second 1 tells you that the span is a timing source, externally clocked. Depending on the mode that your Ericsson is in, you may need to change signalling=pri_cpe to signalling=pri_net /etc/asterisk/zapata.conf: ; Configuration file [channels] ; ; Default language ; language=en context=ntl switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes group=1 callgroup=1 pickupgroup=1 ;echocancel=256 ;channel = 1-6 channel = 1-15,17-31 and /etc/zaptel.conf : span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone = uk On 16 Mar 2009, at 18:11, Oguzhan Kayhan wrote: Hello, I am trying to install my E1 card to make a conection with an Ericsson MD-110 PBX. I installed dahdi drivers as: dahdi_hardware pci::04:08.0 wcte12xp-d161:8000 Wildcard TE121 ran dahdi_genconf and it created all my e1 ports. On the other side i also configured the pbx to communicate with TE121. On ericsson side, i have no error messages. On asterisk side, no error messages. But when i try to create a dahdi trunk, and dial it from asterisk , no call can be made. and also, when i try to call from ericsson side, i get line busy message as soon as i dial the number. Is there any guide that can help me in installing that card? PS: Whatever i made in SPAN config, everytime the only thing i see was Internal clock on dahdi_tool . How can i make my e1 card master (or slave whatever) instead of internal clock?? and other thing i wonder, if i create a span like span=1,0,0,ccs,hdb3 is it zap/g1 in zaptel(dahdi) conf menu in asteriskgui???(or freepbx) ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Asterisk/Skype update
On 23 Feb 2009, at 15:13, Dean Collins wrote: Asterisk/Skype update available here - http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/ …. It’s definitely an update that updates absolutely nothing J, more news at 11 :P John Todd and I discussed this at some length on the VoIP user conference on friday (I'm on a jittery hotel wifi so a bit garbled.) http://recordings.talkshoe.com/TC-22622/TS-198841.mp3 Also briefly covered in my blog on ecomm : http://tinyurl.com/b60-ecomm Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Credit Card processing machines
On 17 Feb 2009, at 19:20, David Gibbons wrote: snip We will be testing the ADT connection heavily this week. The modem connections to my understanding are 2400 baud. Over G.711U and a T1 I don't see why this wouldn't be as solid as a POTS line, but our tests will tell! /snip We do *fax* in this way and it works like a charm. We can hit much more than 2400 baud I think too. --Dave Our creditcard company's small print _insists_ on a direct analog exchange line with no other devices in between. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] IAX Java Softphone?
On 15 Jan 2009, at 07:30, Wolfgang Pichler wrote: Hi all, thanks Tim and Mexuar for releasing this here... I have already taken the source - and compiled a little java applet which is self signed to test the whole thing. That was quick :-) I will put it on my site (and allow users to enter host/user/pass/Calling Number,Calling Name,Number to dial...) for demo usage I would be happy to get some feedback about problems - because i am interessted to integrate it in my callcenter project Tim - can you tell me which audio features it does have - as far as i can see there is alaw and gsm - is there also an echo canceller - jitter buffer ? I don't think the GSM codec is actually in there, from memory it does ULAW/ALaw and Slin There is a jitterbuffer of sorts. I never managed to get the echo canceller to work, although the code for it is in the codebase. I will post it here as soon as i have the page up ... If you plan to do significant work on it, please could you put it on sourceforge so others can chip in ? (That's kinda the point of GPLing it) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] IAX Java Softphone?
On 15 Jan 2009, at 10:06, Wolfgang Pichler wrote: Hi, there is no gsm codec - thats correct - i must have seen something else... (is there a gsm - or other - codec implementation available for free use ?) I think there is an LGPL gsm implementation in java. I will test it further - and if it fits my needs - then i will put some work into it... I will put it on sourceforge if you want - but i will also have no problem if you will create it as new project on sourceforge... (i think you would be the better project owner) My friends tell me that googlecode is good too. For personal reasons I'm not keen to be the project owner, but I will contribute when I can. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] IAX Java Softphone?
I'm delighted to be able to say that as part of the agreement on my departure from Mexuar, the Corraleta applet source code Westhawk Ltd wrote for them has been released under the GPL. it is available for download at : http://www.mexuar.com/files/corraleta_sdk.rar Tim. On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
Re: [asterisk-users] IAX Java Softphone?
It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. On 14 Jan 2009, at 15:09, Dean Collins wrote: Wow very cool - what is required for novices to install this application on their websites? Will you be making available some kind of easy install app? Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Wednesday, 14 January 2009 9:39 AM To: em...@mattruby.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Java Softphone? I'm delighted to be able to say that as part of the agreement on my departure from Mexuar, the Corraleta applet source code Westhawk Ltd wrote for them has been released under the GPL. it is available for download at : http://www.mexuar.com/files/corraleta_sdk.rar Tim. On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ -- 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
Re: [asterisk-users] IAX Java Softphone?
On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote: Thank you for getting that code contributed to the community. Is there a spec somewhere of the features supported by that applet? A version history? Docs of the SDK it's distributed as? All I have is the link. I should emphasise that I no longer have any relationship with Mexuar so I'm in the dark as to exactly what their plans are as far as supporting this code is concerned. I'm just one of the original authors and an open-source proponent. I guess it would make sense for someone to open a sourceforge project for it and add those things. Tim. ___ -- 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] IAX Java Softphone?
On 14 Jan 2009, at 17:07, Roberto Fichera wrote: Tim Panton ha scritto: It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. Really great stuff! Could you please explain how to use it in a java application? Thanks in advance. I designed it as a Java applet, so the top level needs Javascript and DHTML from the browser to provide a UI. That said, It wouldn't be very hard to write an application class and some UI classes to turn it into a stand-alone application , but that depends on the complexity of the UI you want. ___ -- 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] IAX Java Softphone?
On 14 Jan 2009, at 18:02, Roberto Fichera wrote: Tim Panton ha scritto: On 14 Jan 2009, at 17:07, Roberto Fichera wrote: Tim Panton ha scritto: It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. Really great stuff! Could you please explain how to use it in a java application? Thanks in advance. I designed it as a Java applet, so the top level needs Javascript and DHTML from the browser to provide a UI. That said, It wouldn't be very hard to write an application class and some UI classes to turn it into a stand-alone application , but that depends on the complexity of the UI you want. I'm interested to use it as IAX2 API within my UI, so something like: - open IAX2 channel - call 123456 - answer a call - close IAX2 channel It is definitely capable of that with an added class or 2. - but remember it is GPL, so you would 'taint' the rest of your code - if it isn't already GPL. - Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] IAX Java Softphone?
On 14 Jan 2009, at 18:11, Matthew Rubenstein wrote: On Wed, 2009-01-14 at 17:38 +, Tim Panton wrote: On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote: Thank you for getting that code contributed to the community. Is there a spec somewhere of the features supported by that applet? A version history? Docs of the SDK it's distributed as? All I have is the link. I should emphasise that I no longer have any relationship with Mexuar so I'm in the dark as to exactly what their plans are as far as supporting this code is concerned. I'm just one of the original authors and an open-source proponent. I guess it would make sense for someone to open a sourceforge project for it and add those things. Do you know if there are at least hooks in there for the applet to do video over IAX? No, there aren't. We didn't even implement the video frame classes. I don't think it would be hard to add support for a simple video codec transport. The problem is the renderer. Java basically doesn't promise to deliver any video codecs. You are at the mercy of what happens to be installed on the OS or by 3rd parties (eg Quicktime, DiVX etc). (Caveat - I haven't investigated this for a while, it may be that JavaFX changes this picture) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] IAX Java Softphone?
On 14 Jan 2009, at 19:53, Josiah Bryan wrote:
Tim -
Do you have any minimal docs or hints on what hooks the DHTML/JS
methods
are available for scripting? Something like a quickstart javascript
example?
I'm great with javascript, but I havn't read thru the Java to figure
out
the hooks yet - if thats whats needed, I dont mind, but I'd rather
hear
from the guy who knows best.
I'm assuming something like:
applet id=xyz ...
script
var applet = [get applet ref];
function onDialButtonClick()
{
var number = myFunctionGetPhoneNumber();
applet.connectToServer(my.iax.server.com,user,pass);
applet.dial(number);
[update UI]
}
function onHangupClick()
{ applet.hangupCall();applet.disconnectServer() }
/script
Something like that?
-josiah
It's up to Mexuar to decide if they want to release any pre-existing
documentation
(and since it isn't in the .rar I guess they don't intend to at the
moment).
The easiest thing would be to run JavaDoc over the applet class and
see what public methods exist.
Tim.
--
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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Re: [asterisk-users] IAX Java Softphone?
On 14 Jan 2009, at 18:36, Roberto Fichera wrote: Tim Panton ha scritto: On 14 Jan 2009, at 18:02, Roberto Fichera wrote: Tim Panton ha scritto: On 14 Jan 2009, at 17:07, Roberto Fichera wrote: Tim Panton ha scritto: It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. Really great stuff! Could you please explain how to use it in a java application? Thanks in advance. I designed it as a Java applet, so the top level needs Javascript and DHTML from the browser to provide a UI. That said, It wouldn't be very hard to write an application class and some UI classes to turn it into a stand-alone application , but that depends on the complexity of the UI you want. I'm interested to use it as IAX2 API within my UI, so something like: - open IAX2 channel - call 123456 - answer a call - close IAX2 channel It is definitely capable of that with an added class or 2. Could you point me in the proper source code so I can have a look in? ./corraleta/protocol/netse/BinderSE.java Has a Main method used to test the protocol that would be a good place to start. - but remember it is GPL, so you would 'taint' the rest of your code - if it isn't already GPL. I generally follow the rule than if the library is GPL and if the end user ask for the source code I'll provide the source code as it should. If I made some changes in the GPL code, it will be always released to the original author. In all cases the GPL libraries are always mentioned as they are in our custom applications. We generally use jfreechart, jasper report and so on in our applications with this rules. Wouldn't be sufficient for you ;-)? Not my copyright - not my decision ;-) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] Authorize Microsoft SQL
One way to do this would be using
func_odbc.conf
This allows you to define dialplan functions that are based on ODBC
queries.
Like this, which looks up a meetme room number based on the project
and the 'space' number within that project (sub-project if you like).
[SPACE]
prefix=MEETME
dsn=my_oracle_xe_dsn
read=SELECT confno from meetme where project = '${SQL_ESC(${ARG1})}'
and spaceno = '${SQL_ESC(${ARG2})}'
I use this function in the dial plan like this:
exten = _[0-9].,n,Set(CNO=${MEETME_SPACE(${PROJECT},${EXTEN})})
exten = _[0-9].,n,NoOp(Conferenece no for ${EXTEN} is ${CNO}.)
exten = _[0-9].,n,SetMusicOnHold(dumbout)
exten = _[0-9].,n,meetme(${CNO},1nTM)
If you are using a proper database (like MsSQL or Oracle),
then you can hide some of the business logic
by using a database view instead of messy dialplan logic.
Tim.
On 19 Dec 2008, at 04:42, Steve Wofford wrote:
There is some code somewhere on the Asterisk/Linux box getting the SQL
data, be it a program, script or batch file.
There is something initiating the T-SQL code...
SELECT * FROM supportcases WHERE id = 123456789
This code comes from the client, not the server. The Asterisk box will
have the database drivers (ODBC...), but that just allows a
connection,
there is something that tells the server to return data (via the
query).
You are going to have to write the script (middleman) and pass it on
from SQL to Asterisk. I don't know of anything like this ready-made.
1. DialPlan collect @number from caller
2. Call script, program etc and use the @number as a parameter
3. The script, program etc will the create the SQL Query to query the
database:
SELECT COUNT(*) FROM supportcases WHERE @number = 123456789
4. The script, program etc will then get the number of rows returned,
hopefully 1 or 0 and assign it as a variable.
5. Your script, program etc with then use the following logic:
If @variable = 0 Then
Play enter your case again Voice Prompt
ElseIf @variable = 1 Then
Connect to Agent...
HTH,
Steve Wofford
www.uctrlit.com
P.(949)743-0233 Ext. 200
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gregory
Malsack
Sent: Thursday, December 18, 2008 20:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Authorize Microsoft SQL
Steve, my friends setup does not utilize perl/php code. His
communication is directly between asterisk and mysql, there is no
middle
man. This is what I was hoping for with ms sql. But it doesn't sound
like that will be the case.
Thanks for everything!
Greg
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Wofford
Sent: Thursday, December 18, 2008 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Authorize Microsoft SQL
This is exactly what you need. Get your friends perl/php script and
the
SQL code will be near identical, or at least you will have no problem
changing it yourself even if you don't know SQL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gregory
Malsack
Sent: Thursday, December 18, 2008 20:13
To: f...@teamforrest.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Authorize Microsoft SQL
This much I already know. This information is easily found through a
simple google search. What I'm looking for is if anyone knows what a
dialplan would look like that would perform an ODBC query to an ODBC
database. I've seen minuet documentation on ODBCget, which is what I'm
thinking will do the trick, but as I said the documentation on this is
so vague that I'm not quite understanding it.
There's also the possibility that there is another option here that
I'm
not seeing. One idea Steve gave me, was to create a perl/php script
that
does the query and returns a result code. Basically acting like a
middle
man between asterisk and the MS SQL database.
Greg
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred
Posner
Sent: Thursday, December 18, 2008 9:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Authorize Microsoft SQL
All you need is odbc and freetds. Then it will integrate very
smoothly.
Fred Posner
f...@teamforrest.com
Direct: +1 (503) 914-0999
-Original Message-
From: Steve Wofford s...@uctrlit.com
Date: Thu, 18 Dec 2008 19:46:36
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Authorize Microsoft SQL
Re: [asterisk-users] IAX trunk mixing
If you set IAX2 debug on the HUNGARIAN machine and send the console
output
(or a wireshark output) I'll take a look.
At a guess it is a problem with your iax.conf file.
I generally find it clearer to have separate user and peer definitions
for
each system rather than relying on 'friend' which can be confusing.
Tim.
On 6 Dec 2008, at 20:14, Tóth Csaba wrote:
Hi List,
Help me pls, or you think this can be an asterisk bug and should i
make
a bug report?
thanks,
Csaba
Tóth Csaba írta:
hi,
i have a problem, and i am completely stuck with it, i hope someone
can
point out where is my config wrong.
I have three server, connect together with IAX trunking. The server
are
at romania (10.0.4.23, V1.4.22), hungary (10.0.1.23, V1.4.20) and
serbia
(10.0.3.4, V1.4.22). I have a hardphone (6251) connected to the
romanian
server, i dial a hungarian telephone number, the call goes to the
hungarian server well, but that server recognise the call come from
serbia.. and everything is mixed inside..
the phone starts at context do-phoning on the romanian server.
i called 003620XXX from the phone, and as you see, the romanian
server starts the call in good IAX trunk, but the hungarian server
identifies it badly..
Here is the message on the HUNGARIAN asterisk console about it:
-- Accepting AUTHENTICATED call from 10.0.4.23:
requested format = speex,
requested prefs = (gsm),
actual format = gsm,
host prefs = (),
priority = caller
-- Executing [EMAIL PROTECTED]:1]
MixMonitor(IAX2/telsrv-husrb-1541,
om_1228466966.19588_6251.wav) in
new stack
== Begin MixMonitor Recording IAX2/telsrv-husrb-1541
-- Executing [EMAIL PROTECTED]:2]
Macro(IAX2/telsrv-husrb-1541, kitelco|0620XXX) in new stack
-- Executing [EMAIL PROTECTED]:1] Set(IAX2/telsrv-husrb-1541,
telszam=0620XXX) in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(IAX2/telsrv-husrb-1541,
ZAP/g2/0620XXX) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/0620XXX
-- Zap/37-1 is proceeding passing it to IAX2/telsrv-husrb-1541
here is ROMANIAN console:
[Dec 5 08:51:34] -- Executing [EMAIL PROTECTED]:1]
Set(SIP/6251-00c888c0, telszam=0620XXX) in new stack
[Dec 5 08:51:34] -- Executing [EMAIL PROTECTED]:2]
Set(SIP/6251-00c888c0, ~~EXTEN~~=s) in new stack
[Dec 5 08:51:34] -- Executing [EMAIL PROTECTED]:3]
Dial(SIP/6251-00c888c0, IAX2/telsrv-huro/0620XXX) in new
stack
[Dec 5 08:51:34] -- Called telsrv-huro/0620XXX
[Dec 5 08:51:34] -- Call accepted by 10.0.1.23 (format gsm)
[Dec 5 08:51:34] -- Format for call is gsm
[Dec 5 08:51:35] -- IAX2/telsrv-huro-16384 is proceeding
passing it
to SIP/6251-00c888c0
[Dec 5 08:51:35] -- Hungup 'IAX2/telsrv-huro-16384'
[Dec 5 08:51:35] == Spawn extension (macro-kitelsrvhu, s, 3)
exited
non-zero on 'SIP/6251-00c888c0' in macro 'kitelsrvhu'
[Dec 5 08:51:35] == Spawn extension (macro-kitelsrvhu, s, 3)
exited
non-zero on 'SIP/6251-00c888c0'
here are the snippets of the config files:
ROMANIAN server
iax.conf:
[telsrv-huro]
type=friend
host = 10.0.1.23
user = telsrv-huro
secret = xxx
bandwidth=low
qualify=yes
trunk=yes
timezone=Europe/Budapest
context=incoming-hu
[telsrv-rosrb]
type=friend
host = 10.0.3.4
user = telsrv-rosrb
secret = xxx
bandwidth=low
qualify=yes
trunk=yes
timezone=Europe/Bucharest
context=incoming-srb
extensions.ael:
context do-phoning {
includes {
do-nationalcall;
}
}
abstract context do-nationalcall {
_0036. = kitelsrvhu(06${EXTEN:4});
_6[2-8]XX = kitelsrvhu(${EXTEN});
_7[2-8]XX = kitelsrvhu(${EXTEN});
_00381. = kitelsrvsrb(${EXTEN:4});
_51[567]X = kitelsrvsrb(${EXTEN});
}
context incoming-hu {
includes {
template-companynumbers;
template-spec;
template-helyi;
template-mobil;
template-orszagos;
}
}
context incoming-srb {
includes {
template-companynumbers;
template-spec;
template-helyi;
template-mobil;
template-orszagos;
}
}
macro kitelsrvhu(telszam) {
Dial(IAX2/telsrv-huro/${telszam});
switch(${DIALSTATUS}) {
case CHANUNAVAIL:
Playback(/var/lib/asterisk/sounds/beeperr);
case CONGESTION:
Playback(/var/lib/asterisk/sounds/beeperr);
case BUSY:
Busy();
Wait(5);
};
Hangup();
}
macro kitelsrvsrb(telszam) {
Dial(IAX2/telsrv-srbro/${telszam});
switch(${DIALSTATUS}) {
case CHANUNAVAIL:
Playback(/var/lib/asterisk/sounds/beeperr);
case CONGESTION:
Playback(/var/lib/asterisk/sounds/beeperr);
case BUSY:
Busy();
Wait(5);
};
Re: [asterisk-users] func_odbc questions
On 1 Dec 2008, at 13:38, Giedrius Augys wrote:
2008/12/1 Tilghman Lesher [EMAIL PROTECTED]
On Monday 01 December 2008 06:15:15 Giedrius Augys wrote:
I'm working with asterisk 1.6. And I have success using
func_odbc with
one row query results (SELECT source,destination from cc
WHERE ... ):
exten = s,1,Ringing
exten = s,n,Wait(4)
exten = s,n,Answer
exten =
s,n,Set(ARRAY(NUMBER,REALNUMBER1,REALNUMBER2,STATUSAS)=$
{ODBC_GETVARIABLES(
${NUMERIS})}) exten = s,n,Verbose(1| ${NUMERIS}, ${REALNUMBER1}
${REALNUMBER1}, ${STATUSAS})
But I don't know how to retrieve data, if query returns a lot of
rows. In
documentation I read that need to use in config file:
mode=multirow,
and use function ODBC_FETCH. But how to get result-id variable and
use
ODBC_FETCH?
The initial result in mode=multirow is not data at all, but a
query_id that
may be used with ODBC_FETCH to return the first row of data and every
subsequent row of data (up to the max number of rows, if any).
And another question is, if I execute not SELECT , but stored
procedure,
and this procedure will return two, three tables? Is it possible
retrieve
these data from couple tables?
If you're talking about a JOIN, then yes. As long as the fields
have distinct
names, then you can retrieve each row in turn, same as any other
query.
--
Tilghman
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Thanks for your reply, but I don't get it... Is there any
documentation or simple examples how to use ODBC_FETCH and so on.
For your own sanity (if nothing else) I'd wrap the stored procedure in
a view, then
get FUNC_ODBC to query that view.
Tim.___
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Re: [asterisk-users] pick up IAX2 calls
I think it doesn't work across channel types. So it works (if I recall correctly) in IAX or in SIP or in ZAP, but not in mixture. I think that if you have a Dial() that rings several extens, then any of the technologies involved can pickup with *8 So if you have Dial(IAX/fredSIP/billzap/mark) then someone in the same group as fred can pickup with IAX and someone in the same group as bill can pickup with SIP etc. So it's an asterisk thing, not an IAX thing per-se. Tim. P.S. (you could try putting in a dummy 'fred' entry into Dial and iax.conf.) T. On 25 Nov 2008, at 01:09, Bruno Castelo Branco wrote: hi thanks Luis , but doesn't work. For SIP extensions works well *8, but for IAX a tried *8 and ** + iax extension and didn't works Luis Morales wrote: Try with ** + iax extension Regards, Luis Morales On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco [EMAIL PROTECTED] wrote: Hi Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. Any idea? thanks ___ -- 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 ___ -- 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] Large Asterisk installarions (~10, 000 extensions), preferably at universities
On 22 Nov 2008, at 00:06, Michael Collins wrote: Date: Fri, 21 Nov 2008 16:20:28 -0600 From: Terry Wilson [EMAIL PROTECTED] Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes Yehavi Bourvine wrote: OK, but I still did not get a reply to my original question: Why using SIP registrar in front of Asterisk and not simply use bare Astersik? can't it handle the load? (remember - in my case it doesn't handle the RTP, only signalling). Can't it handle so much registrations? (I am using realtime DB, it is has any relevance). My experience has shown that using a dedicated registrar for large installs is more effective; it doesn't tie up resources on the Asterisk box with all those registration refreshes, for one. A product built to be a high-throughput standalone registrar will handle the concurrency requirements and perform better. I've looked at doing various things to chan_sip to improve signaling performance (hash tables for call lookups, etc.) I gave up when I realized that the overhead of handling the RTP was so far above the overhead of processing SIP signaling that it didn't really matter much. The only reason I have ever had to use a SIP registrar (OpenSER in my case) was if I needed to load balance calls across multiple asterisk servers. If most of the phones are not separated by a NAT from Asterisk (as would be the case in something like a University network), the registration timeout could be set to a relatively high value w/o causing any problems which would cut down on some of the SIP traffic from registrations. In fact, I just ran some tests using SIPp and w/o any audio, using realtime w/ 10k accounts I can register 100/second while doing 10 calls/second. If you are looking just at registrations every 15 minutes or so, that is 90k devices that could register to asterisk. This was using 1.6.0.1 on my little HP amd64 development box--not anything near the kind of machine that you would probably install in a large installation. Asterisk just gets faster and faster. Some of the it isn't good at x stuff comes from experiences with older releases. In a HA and/or high volume scenario I worry about stuff like this that has been in tree since 1.0 or earlier and is in 1.6, channel.c lines 3825~3828: /* XXX This is a seriously wacked out operation. We're essentially putting the guts of the clone channel into the original channel. Start by killing off the original channel's backend. I'm not sure we're going to keep this function, because while the features are nice, the cost is very high in terms of pure nastiness. XXX */ That's not something I want in my high-end, high-capacity, high-availability production system! Actually that's exactly the kind of comment I _do_ want to see in an opensource platform. It is honest, open and an encouragement to others to think of a better fix. Discourage poor coding, critique the design etc - but please don't discourage this kind of commenting, it is the kind of thing that helps one find a bug _infinitely_ faster that you could without the clue the original author left for you. Tim. ___ -- 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] MozIAX - Mozilla IAX2 soft-phone 3sec delay
On 21 Nov 2008, at 21:12, Joseph wrote: Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone. http://moziax.mozdev.org/ I tried it yesterday on eee pc, connected to asterisk on local LAN and the performance is terrible! The delay is about 2sec or 3sec. and very bad echo. I think it is the implementation of their IAX2 in their add on, as I have tried external mic. and the same delay problem. As a comparison I've tried DIAX over dial-up connection and the voice quality was acceptable with very little delay. Sounds to me as if you'll need to tweak the audio settings on the eee . DIAX probably does that for you, it might be worth looking in /proc/snd while DIAX is running to see how it configures the audio device, then getting moziax to do the same. I'm surprised you got reasonable response over a dialup connection - which codec are you using ? Tim. ___ -- 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] SIP to IAX2 with delayed echo
Ok, I'll bite, what possible IAX bugs/shortcomings/features can cause echo ? Tim. On 20 Nov 2008, at 18:47, Steve Totaro wrote: Simple tests. Change from the highly touted IAX2 to SIP, but before that, download X-Lite and see if you have the same delay. If you don't then look at your Polycoms, if you do, then switch to SIP. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Thu, Nov 20, 2008 at 1:39 PM, Dave Fullerton [EMAIL PROTECTED] wrote: There are also settings which will turn on local echo cancellation for the handset, headset and/or speaker phone. I don't recall their names at the moment. They are off by default on the handset and headset unless you're using a very recent (3.0+) SIP app. Tim Nelson wrote: I'm not sure about the 3 second delay, but I've seen plenty of echo issues on Polycom phones when the gain has been changed on the handset. Check the voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're not too high. You also may want to make sure there aren't any system resource constraints such as high CPU usage or memory usage... :-) Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - c james [EMAIL PROTECTED] wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what they wanted to use!) ___ -- 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
Re: [asterisk-users] tired of midget packet received warnings
On 7 Nov 2008, at 08:49, Louis-David Mitterrand wrote: On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. T. ___ -- 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] tired of midget packet received warnings
On 7 Nov 2008, at 09:57, Louis-David Mitterrand wrote: On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. Granted, the monitoring app is simple minded: it only checks if a port is open. In that respect is does a hell of a good job: I hear a beeping alarm as soon as an asterisk instance goes south. Yep, but it won't tell you that the single IAX thread is blocked in a database access, so asterisk is ignoring your packets, it just hasn't closed the port. So what you are saying is that all monitoring apps should speak native iax, else they are bad? Simply checking if a port is open means it's misconfigured or badly written? I wouldn't go so far. Small generic port-monitoring apps should be allowed to check on asterisk without raising such spurious warnings. You know what happens when crying wolf to often, no one listens after a while. A midget packet is not corrupted, I do have a stateful firewall (fiaif) to intercept those. Kinda, certainly I'd be inclined to write a little plug-in that sends a valid POKE packet. Tell me what your monitor supports and I'll help you craft a valid packet. rant AFAIK the onus is on asterisk to adapat: I've suffered too long of the infamous iax2 port-clogging bug that would and render a server 'unreachable' for no good reason. So much so that I went off iax2 entirely and use SIP exclusively for inter-asterisk communication. So much for the muched touted new and advanced pbx communication protocol the iax2 was sold for! This deal-breaker bug went unfixed for years until recently, despite numerous asterisk users reporting iax2 anomalies month after month. A I bitter? yes. Do I trust Digium folks to know their stuff about what is correct or not in networking protocols? I'll let you guess the answer. /rant Yeah, that one took _way_ too long to fix, I think the problem was that IAX was undocumented so not many people could fix it, that and the fact that it required a major re-code to get chan_iax2 multithreaded. Ed Guy et al have done loads of work on the RFC, to the point where it is actually possible to implement IAX without looking at the asterisk code :-) so the situation is better now. I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. 'fraid I stand by that bit Tim. ___ -- 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] Advice on ISDN and Asterisk in the UK
On 24 Oct 2008, at 17:00, Phil Knighton wrote: Hello all What I'm looking for is some plain speaking advice on ISDN. Currently using 4 analog lines connecting via a four port TDM400P FXO card. We need to physically move our installations, and on requesting the analog lines be moved - our telco (BT) is suggesting we replace our analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us six voice channels. They've even offered us free installation of the lines (as opposed to a £560 charge for moving the analog lines!) What hardware would you recommend in the Asterisk box? I don't mind admitting I'm a newb and a lot of the info I've found is over my head. I've been looking at a TE410P - would this achieve what I want which is to connect the 3 ISDN2 connections, giving me six voice channels? Assuming the TE410P is what I'm looking for (or an equivalent - suggestions?) what are the basic points for what I would need to change in my current config? Any help or suggestions would be gratefully appreciated :-) Cheers Phil Couple of things to look out for : 1) FAX! If you currently have a fax on any of the 4 analog lines, then moving to ISDN will require you to do a major dance to get it working with any degree of reliability. (same goes for dialup modems, creditcard processing machines, alarm systems, sky boxes etc) 2) There really isn't any competition in the ISDN BRI market in the UK - once down that road you are tied to BT. If you do want to go to ISDN, think seriously of getting a PRI (30 channels) with only a few channels 'lit'. The normal minimum is 8 out of the 30, but I once persuaded NTL (as was) to put up a 6 line PRI. All the telcos have PRI offerings, and the cards for asterisk are cheaper than the equivalent BRI card. In your situation I'd be taking 2 analog lines and an ADSL, use the ADSL to make voip calls through a good VoIP provider and the 2 analog lines for emergencies, faxes and failover. Tim.___ -- 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] How Secure Is Asterisk
On 20 Oct 2008, at 20:01, Steve Anness wrote: I am sure this has been discussed prior, however, I am sitting here and being asked this very question by my superiors. They are loving what I have done with our two Asterisk servers here; however, they keep asking me if it is secure or not. Of course, as with anything, I suspect that on a secure network they can be reasonably safe. However, realistically if I am using the asterisk server to make internal calls and discussion very private matters, how possible is it for someone to listen to calls? How good is the encryption if any over an IAX trunk? The IAX encryption (encryption=yes in iax.conf) is actually pretty good from what I can see. 3 things though: 1) you can't tell if it has happened - if the far end changes config to encryption=no nothing breaks, your calls just go through un-encrypted - I'd like a must_encrypt setting. 2) The keys are as strong as your iax passwords and the quality of / dev/random on your box. 3) The dialed number, caller id etc all go in the clear, the call setup is unencrypted. Only the body of the call is covered by the encryption. Also there are _no_ endpoints that implement it (except asterisk and our phonefromhere.com softphone) so the last yards to your user will not be protected. Tim. ___ -- 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] How Secure Is Asterisk
On 22 Oct 2008, at 07:23, Nikolai Lusan wrote: On Mon, 2008-10-20 at 14:01 -0500, Steve Anness wrote: However, realistically if I am using the asterisk server to make internal calls and discussion very private matters, how possible is it for someone to listen to calls? How good is the encryption if any over an IAX trunk? There is no encryption on SIP or IAX. If you are only making internal calls (i.e. there is no external exposure of *) then you could put the There is in IAX - set encryption=yes in iax.conf at both ends of a link and all md5 auth'd IAX calls between them will be AES encrypted. Very cool, very easy. Tim ___ -- 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] WebCall application
On 22 Oct 2008, at 10:44, voip crazy wrote: Hello list, Does anybody know any free WebCall solution to let our customer call us directly via our web site? Any clue will be welcomed. Yep, take a look at our offering on www.phonefromhere.com Tim. ___ -- 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] WebCall application
On 22 Oct 2008, at 14:28, Rob Hillis wrote: Tim Panton wrote: Does anybody know any free WebCall solution to let our customer call us directly via our web site? Any clue will be welcomed. Yep, take a look at our offering on www.phonefromhere.com A per-minute charge does not constitute a free solution. Please read requests before spouting off about your own products. oops, sorry, take a look at this instead http://code.google.com/p/njiax/ Of course someone still pays for the bandwidth but. Tim. ___ -- 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] Softphone Framework or Libraries
Yep, we can probably help you, if you are interested send an email to [EMAIL PROTECTED] and someone will get back to you to discuss it. Tim. On 13 Oct 2008, at 18:58, Dean Collins wrote: Tim Panton from Phone From Here was able to implement this functionality when he was at Mexuar so I would check with him. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Ricardo Melendez Sent: Monday, 13 October 2008 1:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Softphone Framework or Libraries Hi to all, I have a project for Customer Relationship Management interfaced with asterisk, I need send the CallerID to my application (via http or tcp/ip), When the phone rings I need to launch a pop-up windows to the Call Center Agent to display customer info, do you know a framework/libraries to make this, if is possible a softphone embedded into html page for the same function. I need to choice one to suit it to my needs Thanks in advance. Ricardo Melendez ___ -- 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
Re: [asterisk-users] Astricon people please post the announcement
On 26 Sep 2008, at 11:17, Grygoriy Dobrovolskyy wrote: I have tryed skip2pbx 580€ yeastar 60 €, the quality is the way behind of a good sip provider, thay are simply not suitable for business, i hope it would not be the case of asterisk addon. Also i wonder if skype auto relay will be disabled (bandwith), wait and see... The Asterisk team said that a) the skype for asterisk code does not act as a supernode - i.e. it only routes traffic for local users, this was one of their requirements. b) they _think_ that in the case where both ends of a skype to skype call are 'local' the huge majority of the bandwidth remains local. c) there will be configuration options controlling which of the transport methods skype for asterisk will use. So you can disable skype over port 443 if you want to ensure that port is available for your ssl webserver (for example) Tim. ___ -- 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] Astricon people please post the announcement
On 26 Sep 2008, at 04:36, Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. I asked Mark about that. They expect to have text to work right, when associated with a voice call. It is less clear what happens it it is _just_ a text session. Olle tells me that 1.6 can do text only calls (he's been working on an asterisk for the deaf project) so there is a decent chance they will get it to work. Tim. ___ -- 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] Astricon people please post the announcement
It's essentially a channel driver. Licensed per channel in the same way that the g729 codec is. Limited private beta opening soon. Tim. On 25 Sep 2008, at 17:47, Steve Anness wrote: So does this mean that my users who currently have skype running on their systems won't have to install anything new once I get things rolling on the Asterisk server? Steve On 9/25/08 11:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- 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 ___ -- 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 ___ -- 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] Astricon people please post the announcement
They demoed it - everyone seems pretty confident it works as advertized. No wide-band codec (yet) Tim. On 25 Sep 2008, at 17:55, randulo wrote: I know a lot of linux and open source people think it's superfluous, but a pseudo chan_skype is huge (assuming it works as advertised). It means anyone with Skype can connect to your server presence. And presumably you can call people via Skype. And use Skype out, etc. On Thu, Sep 25, 2008 at 6:47 PM, Steve Anness [EMAIL PROTECTED] wrote: So does this mean that my users who currently have skype running on their systems won't have to install anything new once I get things rolling on the Asterisk server? Steve On 9/25/08 11:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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] chan_iax2.c: No more space
On 17 Sep 2008, at 23:50, Philipp Kempgen wrote: Tim Panton schrieb: On 17 Sep 2008, at 14:57, Philipp Kempgen wrote: Just a quick question ---cut--- [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 - Circuit/channel congestion) [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: Unable to create call [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 - Circuit/channel congestion) ---cut--- Any idea what causes the No more space warning? There's a comment in chan_iax2.c which says We've still got lock held if we found a spot but that doesn't really clue me in either. Most of the time it works perfectly. In the 1.6 beta code I have to hand, it seems to mean that you have 'used' all the iax call numbers between 1 and 0x4000 in the last 60 seconds, which seems unlikely unless someone is DOS'ing you pretty aggressively. Could that be the case ? More than 16000 channels in 60 seconds is highly unlikely especially since bridging fax transmissions from PRI channels to iaxmodems is all the machine does. IAX is not even used for public communication. Which asterisk version is it running ? Tim. ___ -- 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] chan_iax2.c: No more space
On 17 Sep 2008, at 14:57, Philipp Kempgen wrote: Just a quick question ---cut--- [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 - Circuit/channel congestion) [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: Unable to create call [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 - Circuit/channel congestion) ---cut--- Any idea what causes the No more space warning? There's a comment in chan_iax2.c which says We've still got lock held if we found a spot but that doesn't really clue me in either. Most of the time it works perfectly. In the 1.6 beta code I have to hand, it seems to mean that you have 'used' all the iax call numbers between 1 and 0x4000 in the last 60 seconds, which seems unlikely unless someone is DOS'ing you pretty aggressively. Could that be the case ? Tim. ___ -- 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] Amazing show uptime
On 12 Sep 2008, at 09:20, Michiel van Baak wrote: On 09:59, Fri 12 Sep 08, Stephen Davies wrote: xx-montague-gardens*CLI show uptime System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11 seconds Amazing. Especially considering: [EMAIL PROTECTED]:/var/log uptime 09:58:14 up 18:42, load average: 0.21, 0.09, 0.02 Steve Did ntp/rdate set the clock forward for 38 years right after boot ? I'd guess the battery on your motherboard has died so it is going back to 1970 at boottime. Watchout, because this can also mean that your BIOS is about to loose all settings too which can cause it to forget how to talk to the harddrive :-( T. ___ -- 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] SIP to IAX?
On 9 Sep 2008, at 20:19, Mattias Andersson wrote: Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy for their own phones (Running Widows XP). The reason is that the are using laptops and travel, some are already using softphons and IAX bout some don't like softphons for some reason. If it is not any proxy out their, the will I write o of my own. (Of cause giving it out for free), I think Asterisk for Windows would be overkill. Sorry for my poor English. Regards You might find that a decent USB handset makes your reluctant users happier to use a softphone. Often the objection is about the 'feel' of the thing. Also Tesco (and Freshtel) have some IAX devices. If you really want to put together a SIP to IAX proxy, then a simple asterisk config would do. (You might want to look at Freeswitch or Yate , as their Windows support is a little more 'enthusiastic' than Asterisk's :-) - they both support IAX and SIP ) Tim. ___ -- 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] IAX vs SIP
On 7 Sep 2008, at 21:34, Edgar Guadamuz wrote: Hello, I have been testing a trunk IAX and another SIP, using sipp to generate SIP calls to a Asterisk box. The testing dialplan just connects to another Asterisk box, who answers the call and playback some files. I noticed that the cpu load is higher when I use an IAX, about 90% for 25 simultaneous calls. In the other hand, with a SIP trunk the cpu load was about the half or less. In both cases the Asterisk box was in the middle of the RTP path, and both the trunk and the sip client had the same codec, ulaw. Does it make sense? Why is IAX demanding so much cpu load? Which Asterisk version are you running? There was a specific version (1.4.20 I think) that had made IAX super-expensive. The most recent versions of asterisk _should_ have IAX being roughly equivalent in CPU usage as SIP Incidentally if anyone has comparative numbers for IAX vs SIP on 1.6 betas (or hyper-recent 1.4) I'd love to have them for a talk I'm doing at astricon. Tim. ___ -- 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
[asterisk-users] IAX2 was Re: Problems with 2 Asterisk servers on same LAN
On 8 Sep 2008, at 13:12, Steve Totaro wrote: On Sun, Sep 7, 2008 at 9:57 AM, Michiel van Baak [EMAIL PROTECTED] wrote: On 08:24, Sun 07 Sep 08, Steve Totaro wrote: Maybe the problem is that IAX2 is not as set in stone as the RFCs for SIP? Who is to say it is or isn't compliant to the guidelines? Which of the 100+ SIP/STUN/RTP RFCs with all their optional appendixes are we talking about ? Standardization really _isn't_ SIP's strong point. (Wide adoption is ) Some of us (Ed Guy in particular) are making an effort to get the IAX2 RFC into shape. The draft is pretty usable now. http://www.ietf.org/internet-drafts/draft-guy-iax-04.txt It has the huge advantage that as it is 108 pages, you at least stand a chance of knowing what should happen :-) It is true that the implementation in asterisk has changes and there have been the odd snafu's but that's been true of all the features. I think I would spend a day or two getting SIP working properly, now, rather than spending days trying to figure out audio issues and having to revisit and get SIP working properly in the future. After people are actually relying on the system and already have a bad experience/opinion associated with the New Phone System. I think they should go with the Tech their DID provider prefers. That way you will get the best support from them if something goes wrong. I sort of agree although I doubt I would use a provider that pushed IAX2 except as an option. It has also been my experience that when IAX2 does not work well, the providers do not offer much support beyond the initial configs, failing that, they suggest using SIP as well. Like I said to the OP, if your provider supports it, then it will probably be simpler to implement at your end in this specific case (multiple asterisks behind a single NAT supporting incoming calls). Tim. ___ -- 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] Problems with 2 Asterisk servers on same LAN
On 7 Sep 2008, at 08:38, Gordon Henderson wrote: On Sat, 6 Sep 2008, hugolivude wrote: OS = CentOS 5 Asterisk = 1.4.21 Router = WhiteRussian 0.9 Not sure whether I have a problem w/ Asterisk or White Russian config, so I'm posting to both lists. I have 2 Asterisk servers running behind a Linux router w/ White Russian. I'm having a lot of trouble with REGISTER. The servers are set up this way: 192.168.2.160, SIP 5060, RTP 1-2 192.168.2.170, SIP 5070 RTP 21000-25000 On the 192.168.2.170 server I set rtp.conf, with the 21000-25000 ports and I set bindport=5070 in sip.conf. I _think_ I have the ports forwarded correctly on my router. I set DESTINATION ports for the SIP RTP ports above such that ports 5060 1-2 go to 192.168.2.160 while ports 5070 21000-25000 got to 192.168.2.170. Frankly I find the Firewall GUI a little unintuitive ? here's what /etc/config/firewall looks like: forward:proto=udp dport=5060:192.168.2.160 forward:proto=udp dport=1-2:192.168.2.160 forward:proto=udp dport=5070:192.168.2.170 forward:proto=udp dport=20001-25000:192.168.2.170 This doesn't work though. I cannot get DIDs on the 192.168.2.170 to register. Ethereal indicates that the message gets sent and the server responds. The server seems to be responding on the right port 5070, but it gets a 401 from (one of) my machine(s)! Here's the weirdest part for me. While trouble shooting, I tried port forwarding everything to 192.168.2.170: forward:proto=udp dport=5060:192.168.2.170 forward:proto=udp dport=1-2:192.168.2.170 forward:proto=udp dport=5070:192.168.2.170 forward:proto=udp dport=20001-25000:192.168.2.170 The DiDs on 192.168.2.170 still don't register, but the one on 192.168.2.160 continues to work. How's that possible if the ports aren't forwarding there?!! Do the remote devices know to contact you on port 5070 rather than the default of 5060? Gordon This is one of those cases where it is almost certainly simpler to use IAX2 not SIP. You will need zero config on the router and it will 'just work' - assuming your provider supports IAX that is. ___ -- 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
[asterisk-users] Bridge 2 incoming calls
I think I've forgotten something obvious I've got 2 incoming calls, I want to bridge them - how can I do this ? (assume I somehow know which calls should be paired up...) I could dump them both in a meetme - but that seems wasteful as i _know_ there will only ever be 2 parties. (And I need DTMF to flow through). I may want to record the bridged call, but that isn't vital. I'm thinking of dialing chan_local with a call-id but I'm sure I am missing something simpler. Tim. ___ -- 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] Bridge 2 incoming calls
I knew I'd forgotten something. Doh! On 5 Sep 2008, at 14:57, Andreas Brodmann wrote: Tim, you may want to try: 1) Park call 1 2) Pickup call 1 with call 2 (using ParkedCall) Regards, Andreas 2008/9/5 Tim Panton [EMAIL PROTECTED] I think I've forgotten something obvious I've got 2 incoming calls, I want to bridge them - how can I do this ? (assume I somehow know which calls should be paired up...) I could dump them both in a meetme - but that seems wasteful as i _know_ there will only ever be 2 parties. (And I need DTMF to flow through). I may want to record the bridged call, but that isn't vital. I'm thinking of dialing chan_local with a call-id but I'm sure I am missing something simpler. Tim. ___ -- 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 ___ -- 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 ___ -- 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] Bridge 2 incoming calls
On 5 Sep 2008, at 15:50, Steve Murphy wrote: On Fri, 2008-09-05 at 12:27 +0100, Tim Panton wrote: I think I've forgotten something obvious I've got 2 incoming calls, I want to bridge them - how can I do this ? (assume I somehow know which calls should be paired up...) I could dump them both in a meetme - but that seems wasteful as i _know_ there will only ever be 2 parties. (And I need DTMF to flow through). I may want to record the bridged call, but that isn't vital. I'm thinking of dialing chan_local with a call-id but I'm sure I am missing something simpler. Not in 1.4, but in trunk,(and 1.6.x) there is a the Bridge manager command you can call via the manager interface, which takes two required args, the names of the two channels to bridge, and an optional arg, that will send a tone to the second channel. see main/features.c murf Thats good to know. Will the xml-over http manager interface be able to do it too? (pretty please?) Tim. ___ -- 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] Asterisk 1.6 beta
On 1 Sep 2008, at 17:34, Rob Hillis wrote: VoIP Cyprus wrote: Can you share with me your experiences with Asterisk 1.6? Is it stable enough for commercial service? No. No matter how good some people may tell you it is, 1.6 is still beta software and software is rarely beta for no good reason. Don't even THINK about running 1.6 until it leaves beta and RC stage unless you are truly desperate for the features and are willing to accept random crashes, unusual behaviour and the possibility of things changing before the final release. The company I worked for up until June this year was still selling 1.2 systems until late April because we hadn't worked through all the changes and tested things fully. If your company will depend on your phone system for customer service, don't take the risk. I agree with the advice (i.e. don't use a Beta for commercial service.) But Rob is mixing 2 issues - porting an existing set up to a new version (of anything) is always a pain, there are always unexpected gotchas so once you have service running there is a _huge_ disincentive to upgrading. On the other hand, if you are building a new service, you should go for as new and crispy a version as you dare. The question you to ask is - will it be stable by the time I want to launch?. Doing this can save you at least one round of upgrades in the next year or so. Tim. ___ -- 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] security on localhost connections
On 31 Aug 2008, at 01:15, David Burgess wrote: Asterisk Users - We are presently try to operate a hybrid GSM/Asterisk cellular basestation at the Burning Man Festival in the Nevada desert. (See http://openbts.sourceforge.net). The architecture is basically one where cell phones are presented to Asterisk as SIP users, using the IMSI as the SIP user ID for convenience. (It's running off of a wind turbine is the middle of a dust storm as my alkali-abused hands type this.) When we first got this system running, we were getting hammered with service requests from phones that people left turned on. We tried sending the magic GSM codes for no roaming here, but some of them just kept coming back. It was like a denial of service attack. We figured out that the best way to shut those phones up was just to accept their registrations. We'd send a corresponding SIP registration to Asterisk, that would fail, but we'd report success to the GMS handset anyway so that it would think it had service and stop retrying the registration. Now we've discovered a new problem: Asterisk lets these non-existent make calls even though they are not listed as users in sip.conf. We suspect that is happening because they are all localhost connections, and therefore bypassing some kind of authentication check. These calls also show up in the CDR, but with the SIP ids of real, provisioned SIP users instead of the IMSIs of the phones that are actually making the calls. Any ideas how this is happening or how to fix it? I'm not a SIP expert, but registration is about ensuring that the registering sip endpoint will be able to _receive_ calls so asterisk knows it is 'available' and how to route to it. In the case of an incoming call from these phones, the SIP header tells asterisk enough to help it route the traffic. Asterisk will look up the user and (as Tilghman mentioned) match them against the first password-less user. In IAX (dunno about SIP) the best thing is to add a catchall user which points to a context which rejects all calls immediately. Tim. ___ -- 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] A Suggestion To Asterisk Appliance Developers
On 22 Aug 2008, at 14:55, randulo wrote: On Fri, Aug 22, 2008 at 4:23 AM, Johansson Olle E [EMAIL PROTECTED] wrote: Just some friendly advice if you really want a discussion. Of course, I clicked, read and commented ;-) If this is a way we can get you to say something, Olle, I'm for it! :) This said, I think Michael was trying to work within the idea that those interested in this particular post might want to read it. Michael, if you (or anyone) want to post particular articles here, I'm sure everyone is ok with that. With the exception of spam, obviously. The problem too though is if there are images and links in the post, spam filters, etc. So in the long run, I'd say it's probably better just to do what he did :) I'm more with Olle on this one. I often read this list offline (during my commute) and articles which reference a web page without at least summarizing the content are frustrating :-) Also the archives of this list are a valuable searchable resource (again available offline as spotlight indexes them for me on the my mac). So a short summary to accompany the link is great. Tim. ___ -- 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] Is there a way to encrypt passwords stored in the realtime database?
On 20 Aug 2008, at 18:00, Eric Chamberlain wrote: We are exploring using Asterisk for a project and we are looking for a way to encrypt/decrypt the peer passwords stored in the realtime database (postrges). Ideally, we want to use a public key to encrypt the passwords before they go into the database and have Asterisk use a private key to decrypt the password as part of the call out process. Has anyone developed something like this? I haven't done this in asterisk, but we did do a selective encryption layer for a database on a non-voip project. First - understand what you are protecting against: We wanted to be sure that if the backup/sever/tapes/disk were stolen then the personal data in the database would not be accessible without the private key. The way this worked was a bit oracle specific, but the same concepts are available in postgress. Basically you have a base table containing the encrypted fields, this is what is stored on the disk. You then layer on a view (with appropriate triggers/stored procedures) and the application (asterisk realtime in your case) uses this view. The view takes the encrypted fields from the base table and decrypts them before returning the data to the application. The trick is that the key is stored in the user's login session (ie in memory) and is initialized at startup (either by typing or from somewhere that isn't the disk - think of a flash drive superglued to the wall :-) with asterisk I'd be tempted to have it call me and I have to dtmf the key in! ___ -- 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] Question: Soft phone for ACD agents?
On 21 Aug 2008, at 18:44, Jay R. Ashworth wrote: On Thu, Aug 21, 2008 at 09:40:04AM -0700, Michael Collins wrote: To those running call centers I have a question: what kinds of soft phones, if any, do you use? I’m wondering what is out there that has some hooks for custom applications or host system integration, etc. OTOH, do you prefer a desk phone for any reason? If so, why? The experience in the center I manage the network for was that softphones didn't work out that well, and that regular phones on ATAs Interesting - What were the problems with softphones ? Or alternatively , what would you want from an ideal softphone ? Tim. ___ -- 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
