Re: [asterisk-users] sip register peer (the quest for near 100% availability)
If you have no NAT or dynamic IP in your network, you can just remove the registration process and assign to each peer its IP address. This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. This advice is not valid for android / iphones though. You need the register to be able to have good battery life on those. If you use TCP, the softphone will go to sleep, OS will keep the stream alive. When a SIP packet comes in (INVITE, OPTIONS etc), the OS will wake up the softphone and the softphone will handle the packet. No register means no stream and the softphone will just sleep forever. Z -- _ -- 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] sip register peer (the quest for near 100% availability)
This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. This advice is not valid for android / iphones though. That's absurd. Why would you use a battery-powered smartphone if you are trying to have 100% availability? From what i understood from the original post, Xbrian is looking for a way to work around broken phones that fail to register when they should. I doubt his idea of 100% availability is the same as yours or he would/should be using a different brand/model of phones. + The mobile phone will survive a power outage, because of the register you could be behind NAT as it will open the bindings, you can take it to the bathroom etc. I'm just trying to illustrate the possible advantages of a register before XBbrian redoes his network config. Z -- _ -- 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] Detect Low Quality Calls - Realtime
A few years ago I spoke to a Finnish company that had a commercial solution for automated MOS estimation. So something exists though I have not tested it first-hand. l. -- You need a lot of data to calculate a MOS score, you will need the actual call. The only solution i can think of is that the phones start a fake call as soon as they are in focus and the server calculates some scores based on the fake call. When the client calls, the fake call is terminated and replaced with a real call. About the qualify, i don't know how to get the timing results from within the dialplan, i'm not even sure it's possible without patching. Z. -- _ -- 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] Detect Low Quality Calls - Realtime
On 5.1.2013 г. 03:37 ч., XBrian wrote: I can only detect calls as they hit our server, do the magic and based on latency, bandwidth and MOS (Meaning Opinion Score) - decide whether the call should be let through. I will accept all MOS values of 4.0 You are pretty much limited to measuring the delay and the jitter. The delay you can somewhat estimate prior to the call (with qualify for example). The jitter / packetloss you can only figure out when the call is already up for a while. (e.g. you might have no issues the first minute, but maybe packet loss will come in bursts after a minute). -- _ -- 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] looking for some quality testers for zoiper softphone for android.
On 31.5.2012 г. 20:43 ч., Patrick Lists wrote: Hi zoa, On 31-05-12 17:39, joachim wrote: Ellow, We released zoiper for Android today, available for free here: https://play.google.com/store/apps/details?id=com.zoiper.android.app SIP and IAX is supported, should work quite well, unfortunately it is really hard to test all android and hardware combinations. Any android lovers out there to send us some feedback ? Preferably with packet capture skills ? I am mainly looking for feedback on the audio quality, audio delay and if everything looks ok in the gui. Had a quick look on Google Play. Are g729, AMR-NB and SRTP missing in the description or does it not have those features (yet)? Regards, Patrick Hello, We do not have those features yet, we will add them in the future. Joachim -- _ -- 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
[asterisk-users] looking for some quality testers for zoiper softphone for android.
Ellow, We released zoiper for Android today, available for free here: https://play.google.com/store/apps/details?id=com.zoiper.android.app SIP and IAX is supported, should work quite well, unfortunately it is really hard to test all android and hardware combinations. Any android lovers out there to send us some feedback ? Preferably with packet capture skills ? I am mainly looking for feedback on the audio quality, audio delay and if everything looks ok in the gui. Thank you!!! Joachim (zoa) -- _ -- 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] HELP: Dose G.729 with IPP only worked on Intel CPU?
I know two solutions and the best one is: Buy a digium g729 license. Zoa. Charles Wang wrote: Hi, ALL: I install IPP(l_ipp_ia32_itanium_p_4_1_2.tar) and download the speech codeing (l_ipp-sample-speech-coding_p_4.1.008.tgz) then patch it (g729-041103.diff). My CPU is Centaur VIA Nehemiah with 998.715 MHz processor not INTEL CPU. I choose PIII as its CPU type when I modify Makefile under G729-float. # For PIII OPTIMIZE= -O6 -mcpu=pentium3 -march=pentium3 -ffast-math -fomit-frame-pointer IPPCORE=a6 I got the codec_g729.so and copy it to /usr/lib/asterisk/modules/. Modified /etc/init.d/asterisk and add LD_LIBRARY_PATH and export it. Modified /etc/asterisk/sip.conf and add allow=g729. I worked my asterisk well before add G.729 codec. But after it, my asterisk crashed a few seconds after I run a startup command /etc/init.d/asterisk start. Does anyone have the same problem? signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
From my personal experience, pgsql outperforms mysql when using tables with over 30.000.000 records. For small tables mysql is faster, but also locks up more when 1 thread takes a long time. We used mysql for years, then had to move on to pgsql and never turned back. (we still have some 300 queries / second databases running on mysql for historic reasons.) It all depends on what features you need, most of the time both will do fine, you just need to learn how to optimize your queries and your database config. Oh and SER can also use pgsql so its not needed to stick to mysql, thats not a good reason and pgsql has no problems with 700.000 inserts an hour. Zoa. Giudice, Salvatore wrote: I vote for MySQL. PostgreSQL is fine, but MySQL handles much better under extreme load. MySQL is also usually touted as being generally faster for the average application. However on the flip side, PostgreSQL supports more features than MySQL, such as subqueries, more functionality in stored procedures, cursors, and views. In terms of support, you can get support from MySQL directly, while PostgreSQL means you have to turn to mailing lists. It's really your preference depending on the size of your organization and how skilled your staff is in supporting open source in house. Lastly, be aware that MySQL is distributed under the GNU license with a commercial rider for derivative works and PostgreSQl is a BSD license. Also if you are looking at SER as part of your infrastructure, I would recommend you stick with MySQL. Cheers... SG -Original Message- From: Apollon Koutlides [mailto:[EMAIL PROTECTED] Sent: Thursday, March 10, 2005 2:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Best DB Richard Cook wrote: We use PostgreSQL in house. It performs wonderfully and cross-platform drivers (ODBC, .NET) are way further along than MySQL. We switched from MySQL a couple of months ago and have never been happier. We use Postgres exclusively too (12 databses, several of them with several millions of records, both OLAP and OLTP roles). We switched from informix 4 years ago and we also subscribe to the never been happier point of view. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Important :: Please support the development of a new Jitterbuffer for SIP
Steve-U, This sip jitterbuffer stuff is still for free, no one has to contribute anything, but any help financially, with code or testing is greatly appreciated. Everything is GPL and code is disclaimed to digium. We spent the last 2 months researching + working on a sip jitter buffer, first without using the iax stuff but after long talks with Steve Kann we started all over with his recommendations to have something more generic, in the mean time we are trying to improve the PLC and jitter buffer code as well as testing the stability. The changes made to chan_sip are only minor but the time used to make them and test them are not. Zoa. (Your PLC code rocks so dont take it back please :) Steve Underwood wrote: Olle E. Johansson wrote: Steve Kann has developed a new jitterbuffer for IAX2, that hopefully will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable relase. Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs support in the form of funding in order to take the time to test this out and complete it in time. Please paypal your contribution to [EMAIL PROTECTED] today. Every little dollar is worth quite a lot! I fully trust that Joachim (Zoa) and his team will complete this in a good way and look forward to improved sound quality in the SIP channel. Read more here: http://www.astertest.com/forum/viewtopic.php?t=13 Thank you for your contribution! /Olle The hard work of building the thing was done for free, and now someone brings out the begging bowl for the relatively minor activity or porting into to another home. Frankly, that sucks. Can I have my PLC code back, please? Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mini atx and asterisk (EPIA and the like)
The test is did for my presentation on astricon were partially done on a via samuel 2 See: http://www.astertest.com/forum/viewtopic.php?t=10 Zoa. C. Tomlinson wrote: If you are talking about the epia etc boards, they are mini ITX.. I am running an 800mhz one with 256mb ram as a test server, purely voip, using a couple of SIP and IAX clients. No moh yet. I had to modify the makefile in order for it to work, but once working its fine so far. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: 01 March 2005 14:42 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] mini atx and asterisk (EPIA and the like) Hi, haven't found anything in google's, i wonder if there is a comparative page of what to expect from running * on motherboards like the EPIA and similar ones. Since i have not used *ever* such kind of mini atx form factor boards, I have no clue about their performance. SIP-SIP communications, voicemail SIP-TDM communications, voicemail how may users (SIP hardphones and analog phones via CPE equipment) Thanks, signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP jitter?
Yes, It's untested and unfinished and touches the core of asterisk. (maybe causing massive amounts of deadlocks). zoa. Roy Sigurd Karlsbakk wrote: See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532 There isn't even any code for SIP yet. However the iax integration works wonders for a link with just a bit of packet loss and jitter. Voice conversations are nice and crisp and without the pops associated with lost packets or growth of the jitter buffer. Is there a reason why this isn't in HEAD? roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP jitter?
Actually its not... Its for things supposed to be stable. The jitter buffer is not stable at all, putting this into the cvs-head would mean it would be taken out the day after because all carriers using cvs-head would go down. Its not some addon application you can disable, if this part coredumps, your asterisk coredumps. Btw i am trying the jitter buffer, and as soon as its a little more mature i will start stalking kram to get it into -head, but for now its just too soon. Joachim. Eric Wieling wrote: joachim wrote: Yes, It's untested and unfinished and touches the core of asterisk. (maybe causing massive amounts of deadlocks). So? That's what CVS-HEAD is there for. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP jitter?
Well im using cvs stable in production, but i know several of the carriers out there are using cvs head, because its the only one that has realtime... Anyway, cvs head is not testing. If you want to test the jitter buffer, download the patches compile them and see what happens, then report the results on mantis. If everyone says it doesnt seem to break anything, it will make it to cvs head, if its obviously broken it wont make it to cvs head. I reported it to cause massive deadlocks when using it with several simultaneous calls... This is how the * development people seem to work, and unless someone wants to start a 3rd branch, thats how its going to stay i think I didnt say i wouldnt like the jitter buffer to be in some kind of prepatched asterisk tree, but it should be in a 3rd (all really experimental things should be) I dont think we disagree btw :) Joachim. signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIPP load testing - unexpected message - anyone using sipp sucessfully ?
SIP will get you no RTP, meaning it only works with SIP headers. Asterisks CPU usage is mainly coming from RTP handling. We glued something together that will work for RTP too, you can download it from: http://www.astertest.com/forum/viewtopic.php?t=4 As the moment it only seems to work for non authenticated SIP calls, but it does support RTP. Other options are commercial tools such as WINSIP etc. (more call generators + descriptions can be found in the ppt presentation on www.astertest.com) SIPP works for asterisk testing too, but you need the correct commandline. What did you use ? Joachim Robert Rozman wrote: Hi, I'd like to test Asterisk performance under more concurrent sip calls. I use Sipp, but do get Unexpected message for Call-ID ..., so I wonder if anyone is using sipp succesfully with Asterisk and is willing to share more info about his solution ... Any other convenient way to load test Asterisk ? Is sipp the right tool ? Thanks in advance, regards, Rob. sipp: The following events occured: 2005-02-08 00:23:36: Unexpected message for Call-ID '[EMAIL PROTECTED]': while expecting '100' response, received 'SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.101:5060;received=193.77.90.224;rport=5060 From: sipp sip:[EMAIL PROTECTED]:5060;tag=1 To: sut sip:[EMAIL PROTECTED]:5060;tag=as3e7533a6 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ' . 2005-02-08 00:23:36: Unexpected message for Call-ID '[EMAIL PROTECTED]': while expecting '100' response, received 'SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.101:5060;received=193.77.90.224;rport=5060 From: sipp sip:[EMAIL PROTECTED]:5060;tag=2 To: sut sip:[EMAIL PROTECTED]:5060;tag=as43cce205 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] jitterbuffers - suggested settings
I recommend to deactivate the current jitter buffer and wait till a new one is ready. Joachim. Stuart Elvish wrote: Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know if anybody could suggest the number to put in jitterbuffers= and whether or not they have found this to affect the echo. Any suggestions will be greatly appreciated. Kind Regards Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: iax2-jitter-trunking?
The codecs dont need to support trunking... Zoa, Kevin P. Fleming wrote: Rich Adamson wrote: Looking at sniffer traces of two simultanous calls, its apparent that two calls use two different udp packets. However, since I'm routing the two calls via simple extensions.conf entries without reference to an iax context, I was wondering if that _might_ be the root cause for this. (Or, could it be that jitterbuffer=yes disables trunking.) For trunking to work, you will need to place the calls via a defined IAX peer on the calling host (that has trunk=yes specified), receive the calls via a defined IAX user on the receiving host (that also has trunk=yes specified) and be using a codec that supports trunking (I don't believe G.711 supports it, which is probably what you are using). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: iax2-jitter-trunking?
Try using friend at both ends for testing. Ive seen the behaviour you describe in the past, when one of the asterisk machines was natted. Joachim Rich Adamson wrote: Looking at sniffer traces of two simultanous calls, its apparent that two calls use two different udp packets. However, since I'm routing the two calls via simple extensions.conf entries without reference to an iax context, I was wondering if that _might_ be the root cause for this. (Or, could it be that jitterbuffer=yes disables trunking.) For trunking to work, you will need to place the calls via a defined IAX peer on the calling host (that has trunk=yes specified), receive the calls via a defined IAX user on the receiving host (that also has trunk=yes specified) and be using a codec that supports trunking (I don't believe G.711 supports it, which is probably what you are using). Been using strictly iax-gsm, no 711. A test conducted with morning with an unattended remote * box indicated that sending calls (via the context method) did in fact trunk two within the same packet, but the responses came back in separate packets. Not sure I understand that piece as the return path should not be context sensitive on that remote box. (Of coarse, maybe testing to a milliwatt generator on the remote box isn't a valid test either. Will have to wait for an on-site person for additional human testing.) Given that two calls from A - B are trunked using the context reference, why would the return path for those two calls not be trunked? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: PGP signature signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: iax2-jitter-trunking?
http://astertest.com/astricon_performance.ppt has some more measurements. Zoa. Mark Eissler wrote: The wiki also shows tests with trunking on/off and using different codecs including g.711. It's not stated anywhere that g.711 doesn't support it: http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 With that said, I'm pretty sure I read it too somewhere that the codecs have to support trunking and that g.711 doesn't. Perhaps it was on the list somewhere? -mark On Feb 7, 2005, at 11:44 AM, Kevin P. Fleming wrote: joachim wrote: The codecs dont need to support trunking... Ahh, I could have sworn I read that on the wiki, but now I can't find it. Must have been Monday morning brain failure :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Proxied SIP
zoa. Todd Lieberman wrote: Chris Tooley wrote: I want to install Asterisk for an organization that wants it to do some call routing for them. They have a SIP provider that is going to provide one termination and one origination account. We are going to have to route a rather large number of calls (50-100,000 concurrent), but can't find any information on how to proxy calls adaquately. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users look to SER but 100,000 calls requires a tremendious amount of bandwidth, make sure you have it! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Enhancing performance and utility of an Asterisk machine
For the brave, you can test it out yourself (if you can get the beta to work without documentation) with the callgenerator on http://www.astertest.com/forum/viewtopic.php?t=4 Its far from finished, but it can be used. zoa. Steven Critchfield wrote: On Wed, 2005-02-02 at 18:51 +0100, Roy Sigurd Karlsbakk wrote: 6b- More than 50 calls VoIP to POTS/T1/E1 will kill an * box due to excesive transcoding?? 6bb- unless using quad machines, plenty of RAM and DSP cards? The max number of calls is something that's not really a hard-and-fast number. There's been numerous discussions regarding *'s upper limit of connections, check the previous postings to this list for more info. Again, it depends on a lot of different variables, like what the * box is doing, how many AGIs it's running, whether or not you're doing extensive DB stuff. Generally speaking, you won't go wrong with a box that's beefy in terms of RAM. Exactly what is it that is RAM demanding in Asterisk? A database server should be separate box anyway Asterisk doesn't use very much RAM itself, but any swapping could destroy call quality. My 2 main asterisk machines that are in production only have 256 megs and perform just fine. All newer machines are getting 512meg or more but it is mainly because of the cost is negligible to do the upgrade now. signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Stress Test
We will put a graphical asterisk load tester online next week. ( i know i said this before, but now its really there :) zoa. signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Stress Test
We will put a graphical asterisk load tester online next week. ( i know i said this before, but now its really there :) zoa. signature.asc Description: PGP signature signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Versatel PRA in Belgium/Netherlands
Michael, could you provide me with contact information for your versatel account manager or dutch versatel PRI tech person i could contact? Joachim. Michael Devenijn wrote: Problem solved : The reason was quite simple ... but annoying : Interrupts !!! damned !!! Thank you -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Florian Overkamp Verzonden: wo 19/01/2005 10:08 Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion' CC: Onderwerp: RE: [Asterisk-Users] Versatel PRA in Belgium/Netherlands Hi, -Original Message- Did somebody already configured a Digium card on the network of Versatel in Belgium or the netherlands, and would like to share his configuration. (zaptel.conf / zapata.conf) We have HDLC errors (timings i presume) Yes, we have such setups. Please contact me off-list with some more info about what card you are using etc. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bugtracker Karma Hall Of Fame
It's also very encouraging for someone to accidently find some bug they dont really care about, post information about it and get -2 for not following up in X hours. Zoa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100 or TE410 card an PRA line
Both CRC4 off and CRC4 on will work fine with those cards. Since its a belgian carrier, i probably already set it up in the past. so if needed i could do it again :) zoa. Theodoros Georgiou wrote: Eric Wieling wrote: Michael Devenijn wrote: We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? 120 ohm is an RJ45 connection. YES for the CRC that should be standard for EuroISDN Do not have a clue about the last Theo 1) Neither. Digium cards require an RJ-45 connection. Search the mailing list for info on this. I seem to remember seeing talk of many coax (what your telco wants to provide) to RJ-45 converters available. 2) It doesn't matter 3) I have no idea. I assume you always want 2-way. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internet bandwidth (comparing overhead)
For real life bandwidth tests : check the ppt on www.astertest.com Zoa. alexandre::aldeia digital wrote: Hi, I like to know why iLBC and GSM generate a 40-50kbps bandwidth Is very high, if compared with your calculations for other codecs(G723.1 / 17kbps and G729 / 24 Kbps). Alexandre Kanuri, Seshu (Company IT) wrote: /SNIP/ Some corrections are needed: 6.3kbps of G723.1 will become around 17kbps on the IP level without silence suppression because of the additional overhead imposed by protocols like RTP, IP, etc . If you add the Ethernet (or WAN protocol overhead) this will increase even more (although slightly). Similarly, a voice stream of G729 at 8kbps will become around 24kbps on the IP level, and slightly more on the Ethernet or ppp level (around 25 kbps). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware selection
You might want to take a look at the ppt on www.astertest.com Zoa. Jon Radon wrote: I think the wiki has most of this covered. Just requires a little reading and investigation. http://www.voip-info.org/tiki-index.php?page=Asterisk%20dimensioning I really think it's going to be impossible to account for every variable in asterisk. There's just too many. Okay so we document XX function, but with YY codec or ZZ codec? What happens when it's in turn used with AA function? What CPU is required then? There's no end in sight. On Wed, 17 Nov 2004 17:09:55 +0800, Ronald Wiplinger [EMAIL PROTECTED] wrote: Minimum P-300, PCI 2.2 is the recommendation, but how does the real world works? How fast should be the CPU if I have xx functions ??? How much RAM should I use for xx functions ??? How much hard disk should I reserver for xx functions ??? I did not write the functions, but can we make a list of how much horse power we need for basic plus if this function, and that function? You will not get a CPU below 1G, a hard disk below 80 G, RAM below 2x128 M anyway. What is the recommendation for the the power? bye Ronald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Top posting
Let's keep discussing on the posting format, i'm sure all your asterisk problems will go away by doing so. Now, just to keep things going, if i delete all previous posts in my mail, would this be A) a top post B) a bottom post C) all of the above D) None of the above E) you don't really care as you just opened another useless message, all you want to do now is have your spam filter discard all messages containing the word bottom, or top as all you really wanted to do is get your asterisk up and running. F) The answer for D starts with a capital letter while the other messages don't and this is completely unacceptable, so just to be politically correct you stopped reading at that line. Congratulations, you just wasted at least 30 seconds on another useless message, thats about the same time to setup and dial the telemarketeer torture script, good for hours of priceless entertainment and timewasting. (and you might actually learn something while doing so) My 0.2 cents, (lets collect them to pay the poor bastard.digium will have to hire one day to moderate this mailinglist.) Zoa. Anyone replying to this post should be either damn funny or accept any no-more-support-from-me consequences. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 for *
Roger, Could you send me contact details of the author ? I tried to send you a private email, but i got rejected by your spam filter for not having a reverse dns configured. Zoa. Roger Schreiter wrote: Hi, it is now 3 months ago, that I told here, I were beta testing SS7 for asterisk. I promised to give a result afterwords - here it is: There are still some minor problems (maybe more a zaptel or hardware problem that a SS7 one), but in general it is running very stable. I assume the author will soon present some kind of licencing model. Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Over 10,000 lines. Will asterisk manage?
I'm confident asterisk can manage such a setup, but you will need a damn good consultant to set it up. :) (You cannot buy just a huge asterisk machine, you will need some kind of cluster to do this). Joachim (zoa) jafar mohammed wrote: Hi all, I am to come up with a proposal to setup a network of over 15,000 lines. I would like to scale down the costs by using Asterisk as the main switching equipment. Let me give u the full scenario. 1. Fiber optic cables are to run from the central exchange to over 2 kilometer radius at selected distribution points. 2. Every subscriber will have a CAT5 cable terminating at his residence/office. This will provide both Internet/Voice and maybe video to the subscriber. 3. SIP phones will be used by the clients, codec U-Law. Bandwidth is no problem since the fiber network will provide over 10Gbit. 4. Fiber will run to the main Telecommunication provider(PSTN) and 2 mobile providers. Questions are which media protocol should I use? How many asterisk servers will I need? Are SIP phones/IAX phones reliable for this kind of project and are they available in such quantities? How many simultaneous calls can I achieve if no transcoding is being done? Keep in mind that their is no need for T1/PRI or any other type of external lines. Asterisk is to switch the voice data only. I believe asterisk will be able to handle this without a problem and its the way forward for a country which is ages back in telecommunications. The client has been approached to buy a switching equipment that can handle the stated amount of lines for a figure of $500,000. Asterisk can definately beat that. Jafar __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem compiling ZPAHFC with Suse 9.1, Kernel 2.6.5
Hi, I've some problems compiling/installing the ZAPHFC-Driver. I've download the actuell version bristuff-0.1.0-RC4a from junghanns.net. I use SUSE 9.1 with kernel 2.6.5.-111. I've made the symbolic link to Linux-2.6 and test the link successfully. I've done make oldconig, make menuconfig and make in the linux-directory. When I start ./compile.sh in the bri-stuff-directory (./download.sh alraedy done before), zaptel and libpri will be compiled without problems. But compiling of qozap and zaphfc will end wit error: zt_register, -_unregister, -_transmit, -_receive and -_chunk are not defined It's possible to install the ZAPHFS-driver with make loadNT but it reports 0 channels configured. I've alraedy googled this problem but find only users with the same problem, no resolution. Have anyone one? Joachim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Load test IAX
.call files are through the manager. An simple app exists, and should make it online very soon on www.astertest.com (just cleaning up the code to make it a bit more user friendly atm). At 20:26 21/10/2004, you wrote: Is there a way to load test IAX? I know I can setup long duration calls via manager. Just wondering if there is an app that will spawn sessions easily. Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iLBC/PCM16 Huge Cost
I have seen similar things in the past, but only during startup. When started, do a show translation and look again, if that value is ok, you can ignore the one on startup. Zoa. At 12:06 23/10/2004, you wrote: Hello, During asterisk bootup, I've been having a fun time with a random delay which can be quite long, from what seems to be the codec_ilbc.so file. I notice in verbose mode the cost is rather high, and was hoping someone will have some insight on what's going on here. Prior to a harddrive dying, I was running * on this same hardware flawlessly. The only difference now is a new RAID card (no IRQ conflicts), and a pair of harddrives instead of jsut one. This seems to happen on both CVS and stable 1.0.1. [codec_ilbc.so] = (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ILBC to SLINR, cost 215 == Registered translator 'lintoilbc' from format SLINR to ILBC, cost 629693 TIA, Trevor Peirce ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trabas Radius
If you look very hard, you can find two versions on trabas on the web, an old one, not working and a new one not complete and not installing. (the SQL files are incomplete for example) If you combine both, and you are extremely patient you might be able to get it to actually display something in your browser, by the time you get there, you might understand that its maybe not a very good idea to even want to try to use it. Many people tried, none survived. Joachim. I just saved you a week of complete misery, send me some beer on [EMAIL PROTECTED] :p At 02:04 23/10/2004, you wrote: Any tips, tricks or treats out there? I'm building a new system and would like to move away from my SQL based call rating solution... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iLBC/PCM16 Huge Cost
Could you give us more information on: Distro, kernel version, compiler, makefile flags, version of asterisk, and hardware on your machine, + loaded modules ? GSM to LPC10 is also way tooo slow. - *CLI show uptime System uptime: 27 minutes, 2 seconds *CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - - - - - - - - - - - GSM - - 2 2 4 2 1 1238 - - 529695 ULAW - 5 - 1 4 2 1 1238 - - 529695 ALAW - 5 1 - 4 2 1 1238 - - 529695 G726 - 7 4 4 - 4 3 1240 - - 529697 ADPCM - 5 2 2 4 - 1 1238 - - 529695 SLINR - 4 1 1 3 1 - 1237 - - 529694 LPC10 - 196 193 193 195 193 192 - - - 529886 G729A - - - - - - - - - - - SPEEX - - - - - - - - - - - ILBC - 219 216 216 218 216 215 1452 - - - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iLBC/PCM16 Huge Cost
Could you tell us what RAID card you are using + what drivers you are using for it. Could you try to run it without the raid card ? Zoa. At 12:35 23/10/2004, you wrote: Trevor Peirce wrote: Sure. Running Fedora Core 2 with latest updates (kernel 2.6.8-1.521) on a Celeron 1.70 GHz chip. Half a gig DDR ram, one generic X100P card with it's very own IRQ. Asterisk is the latest CVS. It's about time for bed.. spent too many hours trying to figure out other things that I'm starting to lose it! I'll be back in a few hours to fill in any other details that might help to diagnose this problem. Thanks, Trevor Peirce ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about ISDN reason codes
Did someone actually got this to work ? The examples on voip-info are not correct i think. It will never go to priority 2, since the call already got hungup at that point, using dial|g doesnt seem to work either. Joachim At 04:17 22/10/2004, you wrote: Neill Wilkinson wrote: All, newbie to Asterisk and just trying to get a load of bits together including PSTN interface using Digium Quad E1 interfaces using EuroISDN. Question can I/how do I get access to the ISDN reason codes for call disconnect? /path/to/asterisk/docs/README.variables Pay special attention to DIALSTATUS and to HANGUPCAUSE. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about ISDN reason codes
Thnx Eric, Could you also post your extensions.conf ? (or just the extension calling that macro ?) Joachim At 04:48 22/10/2004, you wrote: Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay special attention to the [macro-dial-result] joachim wrote: Did someone actually got this to work ? The examples on voip-info are not correct i think. It will never go to priority 2, since the call already got hungup at that point, using dial|g doesnt seem to work either. Joachim At 04:17 22/10/2004, you wrote: Neill Wilkinson wrote: All, newbie to Asterisk and just trying to get a load of bits together including PSTN interface using Digium Quad E1 interfaces using EuroISDN. Question can I/how do I get access to the ISDN reason codes for call disconnect? /path/to/asterisk/docs/README.variables Pay special attention to DIALSTATUS and to HANGUPCAUSE. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about ISDN reason codes
Are you sure this works ? (and does it work whatever end hung up ?)
If it works, its not expected behaviour. (at least i dont think it is, it
should never go to the next priority when the call got hungup).
zoa.
At 05:06 22/10/2004, you wrote:
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1})
exten = _91NXXNXX,2,Macro(dial-result)
joachim wrote:
Thnx Eric,
Could you also post your extensions.conf ? (or just the extension calling
that macro ?)
Joachim
At 04:48 22/10/2004, you wrote:
Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay
special attention to the [macro-dial-result]
joachim wrote:
Did someone actually got this to work ?
The examples on voip-info are not correct i think.
It will never go to priority 2, since the call already got hungup at
that point, using dial|g doesnt seem to work either.
Joachim
At 04:17 22/10/2004, you wrote:
Neill Wilkinson wrote:
All,
newbie to Asterisk and just trying to get a load of bits together
including PSTN interface using Digium Quad E1 interfaces using EuroISDN.
Question can I/how do I get access to the ISDN reason codes for call
disconnect?
/path/to/asterisk/docs/README.variables
Pay special attention to DIALSTATUS and to HANGUPCAUSE.
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Re: [Asterisk-Users] Question about ISDN reason codes
Aha, oke :)
I was thinking of the answered statuses. That g was not working for me last
time i checked.
But so at least its working when a call did not get answered, thats already
good news for me.
Thanks a lot...
Joachim
At 05:23 22/10/2004, you wrote:
Yes it works. It will go to priority 2 if the call was NOT ANSWERED for
any reason (busy, number not in service, etc). You may need to add ,,g on
the Dial line to get Asterisk to go to priority two if the CALLEE hangs up.
I do not do post call processing if the CALLER hangs up.
joachim wrote:
Are you sure this works ? (and does it work whatever end hung up ?)
If it works, its not expected behaviour. (at least i dont think it is, it
should never go to the next priority when the call got hungup).
zoa.
At 05:06 22/10/2004, you wrote:
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1})
exten = _91NXXNXX,2,Macro(dial-result)
joachim wrote:
Thnx Eric,
Could you also post your extensions.conf ? (or just the extension
calling that macro ?)
Joachim
At 04:48 22/10/2004, you wrote:
Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay
special attention to the [macro-dial-result]
joachim wrote:
Did someone actually got this to work ?
The examples on voip-info are not correct i think.
It will never go to priority 2, since the call already got hungup at
that point, using dial|g doesnt seem to work either.
Joachim
At 04:17 22/10/2004, you wrote:
Neill Wilkinson wrote:
All,
newbie to Asterisk and just trying to get a load of bits together
including PSTN interface using Digium Quad E1 interfaces using EuroISDN.
Question can I/how do I get access to the ISDN reason codes for
call disconnect?
/path/to/asterisk/docs/README.variables
Pay special attention to DIALSTATUS and to HANGUPCAUSE.
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Re: [Asterisk-Users] Performance with ASTCC.
go check out www.astertest.com It has some of the info you are looking for, more will follow soon. Zoa. At 05:37 21/10/2004, you wrote: Hi everyone, I am looking for information about performance on Astrrisk especialy using astcc. Could anyone send me a table or something like that with columns CPU usage, memory usage and calls supported with this platform. How many calls can it support simultaneously? How many calls per second it should manage? THANK YO VERY MUCH!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 190: Good or crappy
could you tell me how you changed the headset volumes ? does that option also work on snom 200s ? Joachim At 02:03 14/10/2004, you wrote: Hi Sudhir, I purchased couple of SNOM 190 phones last week. Connected them to the Asterisk server, and they seemed to work fine. However, after sometime they seem to lose registration with Asterisk as I can make calls but cannot receive calls. We do not have these problems with the snom 190. It is probably a combination of your settings in sip.conf (maxexpirey defaultexpirey) and the 'Proposed Expiry' of the snom (defaults to 1 hour, user_expiry1: 3600) The headset (has Lucent's logo on them but look like Plantronics') would not work properly either. There is no audio on the other side whereas the same headset works great with GS HandyTone-Analog Phone combination. Finally, I tried putting an amplifier in the middle and now the headset works ok. Do you have the original snom headset? Our snom headsets do not have any logo on it. One of the tech people from Snom told me they have a built-in amplifier for the microphone. Did you connect the headset to the special RJ connector? We also changed the volumes: headset_device: headset_rj vol_headset_mic!: 8 vol_headset!: 15 In the beginning we had big problems with the headset because of an annoying buzzing sound. With help of Snom we figured out that a shielded UTP cable, connected to a grounded switch (or PC) solved this problem. The handset keeps falling off the cradle. Very poor design IMO. Yes, could be better, but not really annoying. Regards, Joris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
Those wild times especially occur before any audio is sent. (e.g. while ringing or pre ringing). At 17:10 29/08/2004, you wrote: On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote: If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. I notice that the timing measurements are still showing wild values at times - here is a partial grab of an iax2 show channels: Lag Jitter JitBuf Format 00020ms 6291456ms ms ALAW 00012ms 6291440ms ms ALAW 00017ms 0004ms ms ALAW 00012ms 286523393ms ms ALAW 00012ms 0025ms ms ALAW -978714621ms 6293280ms ms ALAW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SMP Performance
Another question, will you be needing 25 extensions of 25 simultaneous lines ? What compression will you be using ? g729 ? If its 25 extensions and max 3 simultaneous lines or so and you are not using g729, i guess any p3 or higher will do the job just fine. If you need 25 simultaneous g729 encodings, a p4 would be most appropriate. But, a quad xeon will work just fine, i'm using several hyperthreaded dual xeons for the job. (looks like 4 processors to asterisk). Cheers, Joachim. At 15:43 25/08/2004, you wrote: 25 should be the max ever. This machine used to be my testbed server. I may end up swapping it out later for a 1U IBM, but I just wanted to make sure that in the meantime it'd be able to handle what we are doing with it. We bought it refurbished for $600 about a year ago. I was just wondering about the SMP part, I've been told that it doesn't work well with SMP, and then I've been told it works fine. I just wanted a 2nd or 3rd opinion before I went ahead and implemented this. Another dumb question, I've gotten the idea that the best phones out there are the Cisco 7960s, any other good phones out there that are decently priced? Nortel? 3Com? -Tim -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 8:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SMP Performance There is nothing wrong with running Asterisk on SMP. It runs quite well actually. I'm assuming you just have the Quad Xeon 450mhz sitting around because you can't buy them new anymore, so it probably isn't costing you anything to use it. In which case it isn't a waste. If you are paying more than $800 for it, save it and just buy a new P4 for less. A $200 machine may not be able to handle 25 concurrent conversations, and may have some used or sub-standard parts in it, so that may not be the best choice. You should be able to have upto 25 channels running on this machine no problem, How many maximum conversations do you forsee running concurrently at one time on this system? MATT--- -Original Message- From: Matt Schulte [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SMP Performance Meaning Asterisk won't/can't take advantage of the four CPU's? Or it's overkill for this scenario? -Original Message- From: joachim [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 12:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SMP Performance Send me the quad and i'll send you a 200$ pc to do this job. The quad is heavily overpowered. Joachim. At 22:00 24/08/2004, you wrote: content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary=_=_NextPart_001_01C48A15.130BF232 We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astricon - call for help
Hi all, I'm trying to make a paper/presentation for astricon with a lot of graphs and performance statistics for asterisk. I'll try to handle: - differences between the 2.6 kernel and the 2.4 series - differences in hardware, ranging from slow embedded pc's to SMP setups. - difference between opteron and xeon - difference between icc and gcc (if i find a way to compile it). + maybe different optimization flags. - differences between codecs. - differences between iax/sip/h323 But my major problem is, how should i do the benchmarks ? I need more suggestions. My tests showed that the server load cannot be used to determine the quality of the calls. (i could make perfect sounding calls with a load of 96 - no its not a typo!) In general, i think that overall cpu usage should not be over 50% otherwise small clicks start to appear. So far i think of trying to measure the maximum amount of calls (without audio) i can make to a server untill it refuses to accept new ones, by using a wait in the extensions.conf. One way codec decoding i'd test by sending no audio to the server, but connecting using alaw and decode a prerecorded (gsm etc) file on the server. Codec encoding i could maybe do by sending alaw to the server and monitoring to /dev/null. What else can you guys think of ? I'd like to split up calls into smaller chunks. (decoding/ encoding / sip / zap / applications etc) to make it easier for people to say, hey i'll be using sip(x) to zap(y) with g729(z), what server do i need ooh its x + y + z Feed me... :) Zoa. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMP Performance
Send me the quad and i'll send you a 200$ pc to do this job. The quad is heavily overpowered. Joachim. At 22:00 24/08/2004, you wrote: content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary=_=_NextPart_001_01C48A15.130BF232 We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] redhat 9 and oh323
Could you define 'problem'? At 01:31 7/08/2004, you wrote: Hi ALL: I was compiled oh323 successfully on redhat 8 with gcc 3.2-7. But the same asterisk-oh323 on another machine with redhat 9 has problem.i shall say redhat 9 uses gcc 3.2.2 as its default compiler. any suggestion? mohammad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
I think the reason is because the telephony equipment of your telco is still analog. (In belgium it was the same, until they started replacing all the old stuff with fancy digital things.). At 17:30 29/08/2004, you wrote: Hi, in Spain that process is correct. If you setup a communication between a caller and a called, if called phone hangs, in caller side hear a silence, but is a correct process. It's is due to in the called side you can hangup a phone and pickup other phone without lost communication. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Walter Klomp Enviado el: jueves, 29 de julio de 2004 16:44 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything seems to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... Here is my Zaptel config and my Zapata config. I presume the extensions config etc are OK as my call-flow never changed and things were working fine with my AS5300. Am I missing something ? How do I debug the Zap channels ? Cheers, Walter Klomp /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 # This is the line in question... span=2,1,0,ccs,hdb3,crc4 # not used yet span=3,0,0,ccs,hdb3,crc4 # not used yet span=4,0,0,ccs,hdb3,crc4 # not used yet # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 alaw=1-124 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 signalling=pri_cpe channel = 63-77 channel = 79-93 ; Span 4 group=4 signalling=pri_cpe channel = 94-108 channel = 110-124 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiSIP and Zyxel Prestige 2000W
I'm also using a ZyXEL, and sound quality is very very bad when using WEP :/ Any solutions to this problem ? (or download links to newer/other firmware ?) Joachim. At 23:26 13/07/2004, you wrote: The ZyXEL firmware is still quite buggy and has some serious usability issues. When it works, it works quite well, sound quality is pretty ok with G.729. I tested the Pulver firmware, but did not notice any substantial difference (e.g. same bugs as with ZyXEL) I am still waiting for a 1.0 release. Go for it if you can live with early adopter pains. There is one bug which I really have a problem with: The phone does not properly communicate a SIP cancel to Asterisk. Some people claim that it works for them... Dominique Steve wrote: Hi, Anyone have any experience with either of these, I 'd appreciate some feedback? Plus it seems pretty easy to steal a connection with this. Zyxel Prestige 2000W WiSIP thanks, - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -- dominique kull taridium.communications ltd t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] illegal instruction -via c5
Gentoo has a page with all the allowed compile flags for different versions of the via processors and gcc versions. (google is your friend :) Zoa. At 10:32 7/06/2004, you wrote: brian k. west wrote: Its called searching the mailing list... Check the Makefile it does have some indications of what to do on a VIA chip. # Pentium VIA processors optimize #PROC=i586 also, depending on your version of GCC, there may be a bug that allows emitting some instructions it shouldn't ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP CBQ BandLimit HELP!!
Daniel, Do you have a working firewall ruleset for HTB, optimized for voip ? Joachim. (Zoa) At 10:55 1/06/2004, you wrote: Hi Carlos, Try HTB. It is better than CBQ, requires less CPU and have a better help: http://luxik.cdi.cz/~devik/qos/htb/ Daniel Carlos Arnt wrote: Hi all, Reading about CBQ on internet i can say I dont understand well ;) So anyone that has a good background can help me out with this simple question ? I just want priorize my UDP packets to always has 90% of my link when use a VOIP connection with asterisk. My asterisk run in the same machine then my firewall. How then can i : 1 - Mark the packets with iptables then i will know TCP and UDP packets then come in and out 2 - Use CBQ to put a prio=1 in the UDP Packets then i will always know that when a VOIP conn start will always have the best rate of my link. I think i know how mark the packets with the Iptables. iptables -t mangle -A PREROUTING -p tcp -j MARK --set-mark 9000 iptables -t mangle -A PREROUTING -p udp -j MARK --set-mark 9002 and iptables -t mangle -A OUTPUT -p tcp -j MARK --set-mark 9001 iptables -t mangle -A OUTPUT -p udp -j MARK --set-mark 9003 I think that i mark all UDP and TCP packets. So i just need use a CBQ RUle (Now it's the worst) Honestly i dont know .. So let's see. DEVICE=eth0,10Mbit,1Mbit RATE=112Kbit WEIGHT=1Kbit MARK=9000 etc etc I use an 256kbits(Down) - 128Kbits(Up) ADSL connection Then i have PPP0 and my eth1 for my internet net. Just need put the best priority to all UDP Packets forcing the rest of services like SMTP/POP3./HTTP etc that use TCP in the low priority Can anyone help me ? Because i think my Voip has a poor quality because this (Heavy use of mail and http services). Thanks alot for helping out. Carlos I ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP CBQ BandLimit HELP!!
My tests with all shapers on adsl so far give me too much jitter to use without jitter buffer as soon as i do an upload. Zoa. At 13:38 1/06/2004, you wrote: On Tuesday 01 June 2004 06:33, Daniel Bichara wrote: joachim wrote: Do you have a working firewall ruleset for HTB, optimized for voip ? No but you can build your own following htb tutorial. The tutorials frankly suck ass. I am no newbie to Linux or firewalling and it's thorougly confusing. The examples change multiple things at a time and don't sufficiently explain what's going on in each change. I'll post my working HTB script when I get in to work. I'm using a Cisco 2610 with service classes to regulate what's going on on the far end of my link (a Pairgain MegaBit Modem 300S), and Linux HTB on my end. The Pairgains are great little SDSL ethernet bridges but they will buffer an unbelievable amount if you let them. :-( -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hyperthreading?
I also had some dual xeon machines not able to use ht with 2.4 kernels 2.4.22 It all depends on the hardware... Joachim (zoa) At 15:16 1/06/2004, you wrote: I'm running asterisk on a 2.8Ghz w/HT and 2.4.25 kernel. I wasn't aware that I needed to disable HT, but all seems to be running ok for now. The 2.4.x kernel seems to be completely ignorant of hyper threading, which IMO, is quite frustrating. HTT has been around for years now, and 2.4 kernels still can't use it. They can't? HT is detected in /proc/cpuinfo (flags) and I see two processors with 2.4.25 SMP kernels... What exactly isn't it using? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * on Opteron
I have a dual opteron system running opteron on 64 bit. I would not recommend it, its possible to get it to work, but i think a lot of work + you cant use g729 etc... (its not that fast either, and if you want it without msi mainboard its more expensive than a dual xeon.) Zoa. At 19:49 1/06/2004, you wrote: On Mon, 2004-05-31 at 12:07, Greg Boehnlein wrote: On Mon, 31 May 2004, usedcanon wrote: I have used with Athlon 64, but noth opteron. Can imagine it being much different though. I'll let you know in a couple of weeks when my Dual Opteron workstation is finished. We are currently using 2 Opteron servers for our PBX systems (1 backup) and they work very well. We are NOT running these with 64-bit linux, only in 32-bit mode. We are using the Tyan motherboards which come with 2 GigE nics and PCI-X slots =) -- Erik Barker ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom and multiple lines
i think oej is working on something like that... Zoa. At 00:27 1/06/2004, you wrote: At 4:05 PM -0600 on 5/31/04, Rich Adamson wrote: So, first, why do the lights stay on, and secondly, can they light when anyone is using that extension? snip Not only do we need the secretay/boss key system arrangement, but a traveling technician would like to be able to add his SIP extension to someone else's phone when he is working at their station. snip How do I get the lights to work correctly on a SNOM 200 when I configure it for more than one line? The lights stay on solid, although the buttons work correctly for making calls. Thanks in advance. I'm using a snom 200 v2.03o with two extns defined, and the lights work as expected. (They didn't on some earlier version though.) Make sure to define the two (or more) buttons in web interface under Key Mappings (P1 = Line = Number sip:[EMAIL PROTECTED], P2 = ...), matching Settings, SIP, Lines registered Account numbers. If I press extn button #2 and place a call, the callerid properly indicates the correct extension. If I call the extn number assigned to button #2, the LED correctly flashes indicating an incoming call. When the call is complete, all LEDs are off. Regarding your key system question, I've never heard of anyone with a configuration that would actual light button #2's LED if some remote sip phone happened to be on the extension number assigned to that key. If you could dream up a way to do that, it would be dependent on the exact sip phone that you're using. There are no sip standards for turning on/off LED's like that other then the MWI. [snip] Rich Actually, there do exist standards that would be able to provide the functions you're talking about with LED lighting based on who was on what extension. The SIP SUBSCRIBE/NOTIFY tools were written to some degree with that type of feature in mind. In fact, the rumor is that the limited SUBSCRIBE/NOTIFY support in Asterisk is specifically for the Snom phones, but I don't know (and doubt) if it does exactly this. I _do_ know that Snom has touted that they are SUBSCRIBE/NOTIFY compliant, but I don't know the exact methodology of how they light up lights, put things on the LCD, or whatever. If someone wants to send me a Snom 220, I'll be happy to figure out what's required. :-) The Polycoms are also rumored to support this type of feature, but again I don't know the exact mechanics to make it work. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No ringing sound on GS phones
Make sure to use CVS-head and you'll get ringing. At 23:51 28/05/2004, you wrote: On Fri, 2004-05-28 at 16:54, Stefan de Konink wrote: The same problems occurs at our Red Hat system after the upgrade from 0.7.2 to 0.9.0. I didn't tryed the Grandstream phones, but our SIP enabled Cisco 79xx's. I doubt its the grandstream phones. We have a testbed here like this: GS102 -- Asterisk 0.7.2 -- (IAX2) -- Asterisk 0.9.0 -- GS102 Calling from 0.9.0 - 0.7.2 we get ringback. Calling the other way we get no ringback. Before upgrading to 0.9.0 we got ringback in both directions. -- Robert Withrow, [EMAIL PROTECTED], +1 978 288 8256, ESN 248 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
