Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)
On 25 March 2010 02:42, James Lamanna jlama...@gmail.com wrote: Hi, I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of D-Channels going down and then coming back up (See below). Read all the discussion about many spans - and I've run 16 E1 spans in one box, and run 8 spans under 200+ concurrent calls. Your 1st gen TE410 card is very old and I'd suggest to contact Digium about a firmware upgrade or a hardware upgrade. As for the spans going down:- 1) Make sure you are syncing your clock to your telco (span=1,1,0,...) 2) Make sure you are using the right IDE driver module for your chipset and not the generic one 3) Avoid long runs (25m?) of unscreened cable on the T1/E1 span Regards, Steve -- _ -- 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 tunnel
Problem is that the port 80 you are talking about is a TCP port. Voip (iax and rtp) use UDP On 2/11/10, mosbah.abdelkader mosbah.abdelka...@gmail.com wrote: Thank you Jamie for your good reply. It is a very good idea to hava the media and control transported over the same port with IAX protocol. The difficulty is in that the port is not well known by the network admins. It is usually blocked. My idea is to use a well know port like port 80 (that is not blocked). Skype for example uses this port. I need recommendations and help. Thanks. *-- Please discover scientific miracles of CORAN http://www.55a.net/* -- Sent from my mobile device -- _ -- 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] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
On 9 February 2010 06:42, Muro, Sam resea...@businesstz.com wrote: Hi Team Can someone advice me on how i can lower the load average on my asterisk server? dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10.1 asterisk-1.4.25.1 2 X TE412P Digium cards on ISDN PRI Im using the system as an IVR without any transcoding or bridging ** top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75 Hi Sam! Are there any side-effects from the high load average? The system doesn't seem to be CPU or disk bound from the look of the CPU stats. System %age is high by way - software echo cancellaton?, and Asterisk is using a lot of cpu which isn't suprising. I'm guessing you are running 8 spans and 200+ calls into your IVR? If the system is actually performing fine then I'd just say that there is something about the Asterisk threads that makes them look runnable and that accounts for the high load average. Is the IVR an agi or fastagi or what? - the code path may have a spinlock logic to it that means that many threads are runnable but when scheduled just go back to sleep. That would account for high load average with lots of spare CPU. If that's what is happening then I wouldn't worry much more about it. Regards, Steve PS: Alex - why the dig about ALL CAPS? The post wasn't in caps? -- _ -- 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] Beginners Guide to setting up a Call Centre
2010/1/12 Jeff LaCoursiere j...@jeff.net That is so not true. FreePBX has hooks in a million places to do custom dialplan stuff - I do it all the time. I also link in custom AGI/AMI applications, custom provisioning, custom LCR, and am even working with one customer that has mastered making FreePBX multi-tenant. If you want to get your hands dirty there is plenty of dirt underneath FreePBX. On the other hand, if you want a simple setup that is easily managed, the GUI is fantastic and saves a LOT of time. And if you are a PHP programmer you can easily modify the operation of any part of it. Preach it brother. We take the same approach and have never had any difficulty integrating our customisations into the FreePBX dialplan. The common structure makes it EASIER for my techies to work on systems that we built and support. On asterisk-users its traditional to be hard core and raw-dialplan and look down on those who have projects to deliver and are happy to have the help. I'm not the insecure - each of you writing your raw dialplans runs some of my code every time you run Asterisk. Regards, Steve -- _ -- 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] Beginners Guide to setting up a Call Centre
2010/1/15 Peter Childs pchi...@bcs.org Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever installed with OpenPBX, Asterix etc by hand) I've got a new server to run Asterix on and want to get it working quickly and yet be configurable in the future with out having to reisntall and start again regally. Currently no VoIP hardware but that will come once I prove the concept. I guess Oh the machine does not have a CD Rom Drive so a network/USB install would be nice.. But I guess I can open the case and plug one in for installation if I must! (Says he who has just installed Ubuntu over the network to check the computer works!) Decide if you are going to be a zealot for your preferred approach - Ubuntu and all that - or if you want a solution that works without tons of extra work. If you wisely decide that you want the latter, then get Elastix and install it. Buy QueueMetrics and install on your Elastix build. Start running your inbound call centre. Steve -- _ -- 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] Beginners Guide to setting up a Call Centre
2010/1/15 Doug Lytle supp...@drdos.info Decide if you are going to be a zealot for your preferred approach That's a little harsh, wouldn't you say? Do whatever your most comfortable with. But, to call me and those like me a zealot, for offering advice that was asked for is a little off, in my opinion. Hi Doug, Maybe I read too much into the original poster's question, and I didn't mean to be harsh. But I used to get called in often here in South Africa to sites where the usual way wasn't good enough for someone so they'd put the whole system together the way they thought it should be done and in the process bumped into all the subtle gotchas that are mostly worked out in the standard builds. Then discovered that its harder than they thought it would be and PBX users are ungrateful b*ggers sometimes and they've walked away. Our efforts to recover these installs are always twice the work because they are tainted by what went before. But we hate to see failed Asterisk projects so we try to get them right. If your objective is to run a simple inbound call centre and get good metrics into the bargain then a FreePBX-based ISO-install (Elastic, AsteriskNow, Trixbox-CE, whathaveyou) plus Queuemetrics will have you up an running in short order. Build from the bare metal using your-own-install-of-your-preferred-distro plus raw Asterisk plus dialplan from scratch plus DIY reportage and you'll be working away after a month and cursing Asterisk. Once you're an expert then you may indeed be able to do a better job for your application than the all-in-one distros. But not first time. So apologies to the poster if I read too much into the question, but this is the sort of situation I thought of. Steve -- _ -- 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] hints through a Local channel
What you are missing is the new state-interface parameter to AddQueueMember.
You can't use functions in a hint exten.
Steve
On 12/14/09, Lenz Emilitri lenz.lo...@gmail.com wrote:
Hello all,
I am trying to set up a dynamic channel to be used as an Agent dialer for a
queue - you know, trying to replace AgentCallBackLogin for an Asterisk 1.6.
I would like to do something like:
[myagents]
exten = XXX,1,Set(realchan=${DB(myagent/${EXTEN})})
exten = XXX,n,Dial(${realchan},tT,60)
This basically fetches the actual channel to be used for dialling and dials
it. What I would like now is to make app_queue aware in advance of the state
of each channel, something like:
exten = 100,hint,SIP/705 (and this works)
But more dynamical, so I would try and look up the actual channel in the
AstDB, like:
exten = XXX,hint,${DB(myagent/${EXTEN})}
This does not seem to be working - is there a way to work around this issue?
(I admit this is the fist time I'm trying to use devoce state and the
related functions, so maybe there is a very simple slution right in front of
my big nose and I'm not seeing it).
Thanks a lot for your help,
l.
--
Loway - home of QueueMetrics - http://queuemetrics.com
--
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[asterisk-users] Suggestions for low level RTP stream generator?
Hi, I need to build a simple, command-line method to generate a legal and perfect RTP stream across a network link, and analyse it on the other side and measure network performance. Want to do this for a number of links and over long periods. I'm trying to characterise performance of various available Internet links locally as input into a design project. Asterisk isn't ideal for me because I just want a one-way stream and in the case of things like Echo, Milliwatt, MusicOnHold etc the generated RTP stream is synchronised to the incoming RTP and so issues in the one direction affect the other. And I'm just looking for a lean-and-mean command line program! rtptools from Columbia seems to be almost what I want, except that for rtpsend I have to specify the desired RTP stream in more detail than I want. (I just want to generate a valid stream containing say silence) I also looked at rat (from mbone days) - it can generate a tone test Can anyone suggest other old-time utilities that might help me? Thanks, Steve ___ -- 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
2009/9/9 Armin Schindler ar...@melware.de 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? Hi, I'm not sure if Steve Kann is still around the project, but if not, I'm familiar with chan_iax2.c and mostly familiar with the iax2 jitter buffer so I might have a go at fixing the problem. Will you open a bug on the bugs.digium.com bug tracker. I did do a test from a SNOM820 (yum) via an IAX trunk with jitter buffer and got the same nasty jerky audio. This is a recent checkout of branch-1.4. Regards, Steve Davies ___ -- 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] Asterisk remote calls with low bandwith and high latency
2009/9/8 James Mutuku listmut...@gmail.com I have 2 sites. One(Site 1) has an asterisk PBx and the Other(site 2) has 2 remote soft phones. The latency btw both sites is btw 500ms-700ms. I know this is a shot in the dark...but are there ways of improving the voice quality for the remote calls(btw site 1 and site 2), Other than increasing bandwidth? Hi, The latency will result in long delay on the call and you can't do anything about that. I suppose its a VSAT link? With respect to the bandwidth all you can do is carefully select a low bandwidth codec. This will depend on the codecs that your soft phone supports. Speex can be tuned to use little bandwidth. G729 can be used with licensed codecs from Digium. Rumour is that G.723 is technically possible with some googling and provided that you have some sort of licence to use the G723 patent. Regards, Steve Davies ___ -- 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] Should digium build a 2FXO / 2FXS 4-port daughterboard?
2009/9/9 Karl Fife karlf...@gmail.com ...of course you need one of these to dial SIP URI's or navigate IVR's from the rotary mechanism. http://www.oldphoneworks.com/rotatone-pulse-to-tone-converter.html On Asterisk I don't think that's true. At least for IVRs on the local Asterisk box, Asterisk will process a rotary dialled 4 just as if you'd pressed 4 on a DTMF phoneset. I guess if you want to operate a remote IVR via the dial phone you'd need the converter. Are you in the US? Hope you use loadzone=us-old. Steve ___ -- 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] Inquiry:Problem with Call Parking
In any event, the real problem is probably that you forgot to 'include = parkedcalls' in your dialplan. Steve On 9/2/09, Lyle Giese l...@lcrcomputer.net wrote: And now that the whole world of Asterisk has your sip user ids and passwords, you should change all of the passwords that are in that file and yes, change the passwords in all your phones. Lyle Giese LCR Computer Services, Inc. hadi motamedi wrote: Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.com mailto:john@compuware.com wrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Tuesday, 1 September 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inquiry:Problem with Call Parking Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- Sent from my mobile device ___ -- 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] I find this incomprehensible ?!
2009/8/30 jonas kellens jonas.kell...@telenet.be I am totally not understanding this : My IAX.conf : register = BOX-YOCAN:pas...@remote_asterisk_ip yocan9...@89.31.97.186 On remote Asterisk : *CLI [Aug 30 20:37:07] -- Registered IAX2 'BOX-YOCAN' (AUTHENTICATED) at ip:4569 So this is normal... Now the following : [remoteasterisk] type=peer host=ip remote asterisk auth=md5 secret=passwd I think you are missing a username=BOX-YOCAN in the [remoteasterik] peer entry. Steve ___ -- 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 step that I do not have
2009/7/21 Jim Dickenson dicken...@cfmc.com How can the first step of the extension be a playback when I do a verbose? Because you have a exten = line that matches *9901 in the empl context or some context that is included into the empl context before (above) dorecord. Steve ___ -- 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] LoadAvg , Codec and Bandwidth Utilisation
Hii, Looking at this, your problem appears to be that you are diskbound. Note the 60% wait time. Use hdparm -t to find out the throughput of each of your disks. If its not 20MB/sec or more then you need to look into the drivers you are using for your disk. If you do have good disk throughput and still stuck with high wait time then you need to look into why you have so much disk IO going on. By the way - there's no big problem with an Asterisk box with a load average of 4. I've seen 10 or so without audible issues. Obviously depends on available CPUs and what exactly is happening on the box. Steve 2009/5/1 David @ULC ucoms2...@gmail.com [root]# top top - 20:19:40 up 53 min, 1 user, load average: 9.54, 7.85, 6.44 Tasks: 224 total, 1 running, 223 sleeping, 0 stopped, 0 zombie Cpu(s): 7.5%us, 3.8%sy, 0.0%ni, 28.6%id, 59.1%wa, 0.2%hi, 0.8%si, 0.0%st PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 2835 root 15 0 41252 14m 4932 S5 0.4 1:08.82 asterisk 2456 mysql 15 0 154m 28m 4480 S2 0.8 0:42.05 mysqld 2995 root 16 0 12380 7220 2740 S1 0.2 0:21.56 AST_update.pl 10700 root 15 0 2304 1088 788 R1 0.0 0:00.03 top 413 root 10 -5 000 D0 0.0 0:02.54 kjournald 428 apache15 0 58168 6460 2952 S0 0.2 0:00.10 httpd 3004 root 15 0 11096 5804 2640 S0 0.2 0:03.12 AST_VDauto_dial 6725 root 15 0 12644 7400 2600 S0 0.2 0:00.90 AST_CRON_audio_ 10428 root 18 0 21536 15m 2600 S0 0.4 0:00.28 AST_CRON_audio_ 13471 apache15 0 58168 6780 3260 S0 0.2 0:00.50 httpd 17253 root 15 0 13248 6696 1456 S0 0.2 0:00.30 FastAGI_log.pl 29074 apache15 0 58168 6692 3172 S0 0.2 0:00.38 httpd 30454 root 16 0 21536 15m 2596 S0 0.4 0:00.65 AST_CRON_audio_ 1 root 15 0 2044 664 572 S0 0.0 0:00.58 init 2 root RT 0 000 S0 0.0 0:00.02 migration/0 3 root 34 19 000 S0 0.0 0:00.00 ksoftirqd/0 4 root RT 0 000 S0 0.0 0:00.00 watchdog/0 5 root RT 0 000 S0 0.0 0:00.01 migration/1 6 root 34 19 000 S0 0.0 0:00.00 ksoftirqd/1 ___ -- 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] SIP trunk with 250 lines
Hi, We have a customer who used a strong quad-core Xeon box to convert up to 800 simultneous calls from SIP to IAX and trunk them to another box. So your requirement doesn't look like a big problem. Steve On 3/24/09, Christian Victor christ...@victormedia.de wrote: Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will be connected to the server through the same SIP trunk as this will be some kind of a hosted pbx. Given he finds a provider wich has this much SIP capacity and IP bandwith and no codec conversion is needed - do you think this is possible with pure asterisk on a decent system? Is there anything I shoudl watch out for? Your help is much appreciated! Chris -- Sent from my mobile device ___ -- 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] Special Information Tones
Hi, Are you sure that Verizon amswers the call? They should play that message as 'early media' without answering, after which they cam clear the call with an appropriate cause code. That would work for you and still give callers the audible ,essage they want. Steve On 3/20/09, drew einhorn drew.einh...@gmail.com wrote: I'm having a problem with Verizon Wireless. I would be extremely surprised if I was the only one having this problem. It seems to me that Verizon Wireless might be able to use one of the Special Information Tones to allow us to solve the problem. But I really do not whether my suggestion is compliant with the ITU-T standards. Perhaps someone can give me an expert opinion on whether I should try to get Verizon to implement my suggestion. First I'll describe the problem. I'm trying to implement Single Number Reach. For example, when a call comes in to one of my DIDs, it simultaneously rings on a couple extensions in my home office and a couple of Verizon Wireless cell phone numbers. Everything works just the way it is supposed to if the cell phones are powered up, and within the range of a cell tower. The problem is if a cellphone is turned off, or out of range and unable to talk to a cell tower, Verizon is unable to find the cellphone on their network, Verizon answers the call and plays a recorded message, instead of allowing the number to continue ringing, and allowing one of the voip extensions, or another cellphone to answer the call. Verizon really wants to get rid of the call as quickly as possible to free up their equipment to handle other calls. Unfortunately we spend a lot of time in rural areas where there is no cell tower to talk to. In that case we really someone else to pick up the call. I'm hoping that if Verizon would precede the voice message with one of the Special Information Tones, we could recognize the fact that the call has not really been answer, and continue to ring on the other lines. Two questions. 1) would the approach be compliant with ITU-T standards? 2) Assuming that it is, and we can convince Verizon to implement this. How difficult would it be to configure asterisk to handle this as I suggest? -- Drew Einhorn ___ -- 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 -- Sent from my mobile device ___ -- 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] Best way to get 60+ analogue extensions.
Hi, Xorcom make what you are looking for. Steve On 3/15/09, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. I can see the Digium AEX2400 with 24 lines, physically they are all very deep, if I had 3 of these in a server it would seem straight forward assuming the motherboard doesn't haven't anything get in the way Equally the Digium TDM2400P supports 24 lines and physically requires similar space The Sangoma A400 provides 24 ports but uses two slots, having 3 of these in a server looks like I need to pick the server carefully. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. I am just interested in any recommendations for server hardware and card combinations that are currently in use. Also if anyone has provided call data out to the RMS system ( http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to hear how it worked. Thanks very much Cheers Duncan ___ -- 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 -- Sent from my mobile device ___ -- 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] Initial silence during call
If there is NAT between the phone and * then that can be responsible. Also, Eyebeam (et al)'s ICE setting causes this. Steve On 3/13/09, Mike Diehl mdi...@diehlnet.com wrote: Hi all, I've got a problem where many times, there is silence at the first 1-2 seconds of a call. Then it clears up and it's crystal clear. I've not put a sniffer on it, yet, but I suspect that the media channel is still being set up. The server shouldn't be too overloaded. Can anyone give me some advise on how to solve/mitigate this problem? Mike. ___ -- 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 -- Sent from my mobile device ___ -- 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] Ast/Hyla/IAX Scalability?
Hi, I know i doesn't make practical difference, but often it is the far end that is atually buggy, not out end. A lot of the work in spandsp to increase success rate is to do with workarounds for issues in the remote machine, Steve On 3/13/09, Marshall Henderson marshall...@gmail.com wrote: On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg dbackeb...@gmail.com wrote: On Fri, Mar 13, 2009 at 11:38 AM, Marshall Henderson marshall...@gmail.com wrote: I recently read the thread entitled Faxing Success Rate on PRI which dealt with Asterisk/HylaFax/IAXmodem. I'm successfully using this 'recipe' in a few instances on systems with only a couple of analog lines all the way up to a full PRI worth of Iaxmodems. Then you have probably seen that YMMD, and that some people claim great success with VoIP fax. Other people claim that the only way to go is a hardware fax solution, like the dedicated multi-modem fax cards. The only way you're going to find a solution that will work for you is to try it, scale it, build your own expertise with your solution, load test it, and watch your error rate. I certainly understand the value of building the solution, testing, patching, and fixing problems as they arise. It was my hope however that others would have large-scale experience with these technologies and could share some pointers. I'm about to perform some bulk testing between two servers to see how the system reacts. I'm more than happy to post my findings here if anyone has interest. The other consideration is your budget and your cost of dropping a fax. The faxmodem cards are not cheap compared to a voip solution. But if the faxes have a high value to the business the hardware cards are probably justified. Again, you'll find people arguing that their voip solution has as low of a failure rate as a hardware solution. I'm jealous. My voip fax solution does not yet have that low of a failure rate, but I'm hopefully getting closer to working out the last bugs. Do you have any specifics to share about the problems you're finding? I've also noticed that IAXmodem is compiled statically against a version of spandsp included with the iaxmodem source. For a large installation, would it be better to compile iaxmodem dynamically to reduce the per-instance size of each iaxmodem? Or, would it be better to simply throw more RAM at it? I'm not sure what difference RAM makes. What breaks a fax on voip is latency and dropped packets. Agreed. I was simply inquiring about the efficiency of IAXmodem at the system resource level. Latency and packet drops will be minimal or nonexistent at all in this environment. You solve both of those problems if you go the hardware solution route with a faxmodem card. I've found hardware fax boards aren't a 100% fix either. Many of the boards are buggy. However, I will have to say that certain manufacturers like Mainpine are near 100%. The in-between solution is using a proprietary telco - fax gateway, like a Cisco box that terminates a PRI and provides FXO ports that you plug into a single-pair faxmodem or a 'real' fax machine. That solution quickly becomes ridiculous when you try to scale it. Are there any concurrency issues when receiving a large number of faxes on a system using IAXmodems? File system contention, but fax files aren't very large, and I would call that a non-issue. Most people don't want a piece of paper; they want a PDF that they can turn into paper once in a while. The purpose of such a system as I'm inquiring about is for digital archival. Very little 'paper' will be in use. Buffering aside, each fax could be written at the speed at which it is received correct? So, if I'm receiving 50 faxes at 14.4kbps each, assuming a direct receive frame--block write, I'd be looking at roughly 90KBps written to disk. Is my logic sound here? Thank you for the response and ideas. Marshall ___ -- 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 -- Sent from my mobile device ___ -- 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] Hints
To get busy state for a sip channel in 1.4 it appears the peer/friend must have a call-limit. Steve On 3/9/09, Cary Fitch ca...@usawide.net wrote: Running an earlier version of Asterisk (1.2), we were using Hints to show busy extensions on other (SNOM) phones. When we went to version 1.4 they stopped working, using the same syntax. (Copied and pasted) Does anyone have any tips or clues? Is the exact location in the file critical? Maybe we put the code in a back alley? Cary -- Sent from my mobile device ___ -- 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] Echo on SIP to SIP calls?
It can only be acoustic echo. Asterisk doesn't cancel that - it's the phone's job. Maybe it will fix it to reduce volume of the phones. Steve On 2/27/09, Bruce Komito bru...@bagel.com wrote: I know the subject of echo has been discussed ad nauseum, but I think I have a somewhat unusual problem. I am suddenly experiencing occasional echo on SIP to SIP calls. This is a new development and has never happened in all the years we've been running *. The phones involved are not junk phones (Cisco 7960's and Linksys 942's). I don't recall seeing any settings anywhere than have anything to do with echo cancellation on non-ZAP devices. Anyone have a clue where I should start looking? TIA Bruce Komito WPTI Telecom (775) 236-5815 ___ -- 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 -- Sent from my mobile device ___ -- 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] Strange dialplan matching issue
Hi,
As others have mentioned, the 'n' is a pattern char.
I have a system that uses similar tricks to yours. What I did about
this issue was to change the pattern match chars to be upper case
only. Drop me a line if you want the patch.
Regards,
Steve
On 2/12/09, Chris Bagnall li...@minotaur.cc wrote:
Greetings list,
Wondering if anyone has come across this strange dialplan pattern matching
issue before:
I have a context defined as follows (the plus simply implies it follows on
from an existing context in another #include - which, yes, has been included
first):
[privatedundi](+)
exten = _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
When dialling hilton-202 from another box via IAX2, I get:
NOTICE[3727]: chan_iax2.c:8085 socket_process: Rejected connect attempt from
ip masked, request 'hilton-...@privatedundi' does not exist
Changing the context to read as follows solves the problem immediately:
[privatedundi](+)
exten = hilton-201,1,Goto(hilton,${EXTEN:7},1)
exten = hilton-202,1,Goto(hilton,${EXTEN:7},1)
exten = hilton-203,1,Goto(hilton,${EXTEN:7},1)
Dialling hilton-202 now works every time.
The *really* strange thing is that I have a number of similar pattern
matches, and all the others work fine, it's just this one that doesn't.
The box in question is running 1.4.22, but I have had a similar issue in the
past with a 1.2 box, so it does not appear to be version specific.
Any thoughts?
TIA.
Regards,
Chris
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Re: [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions
Everybody is talking about other products. But yes, the Xorcom will handle all ports active, supports a high density connector at the back, looks just like standard Zap/Dahdi ports to Asterisk, rack mounts nicely and much less $$ than the other solutions. Steve On 2/10/09, Erick Perez eaper...@gmail.com wrote: Hi, I am looking to connect 66 analog phones to an asterisk box. I was thinking of a Xorcom astribank 32port (2 of them and another 8 port). this is because the phones have no near connection to an ip network, so replacing the phones in favor of voip phones+network cabling is kinda out of the question. In your experience, will these units support all the phones talking at the same time with other units on the astribank, as well as to the pbx, pstn, etc? The asterisk pbx will be a server-class Hp Proliant unit (potentially a dl320). i must make sure the astribanks will not die when fully utilized. other hardware suggestions for this task will be nice. thanks, -- Erick ___ -- 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 -- Sent from my mobile device ___ -- 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] Sonicwall potentially causing long ping times to SIP phones
2008/10/23 Kristian Kielhofner [EMAIL PROTECTED] Most of the anything but simple PAT devices I've seen that implement any SIP specific fixups usually end up breaking something along the line. Unless the product is from a company where SIP is their core competency (like Ingate, or /maybe/ Cisco) it's best to stay away and/or disable the SIP specific fixups wherever possible. CISCO PIX's SIP fixup stuff breaks authentication from a SIP device if the SIP device is using an IP address for the proxy and not a DNS name. This is because the PIX rewrites the proxy's IP address where-ever it is seen. And that includes inside the authentication challenge line. (The PIX appears to do a literal search-and-replace in the SIP headers). Which means the authentication fails. We hit this twice with customers. Unfortunately long enough apart that I had to debug it all over again because I forgot about it... The workaround is to use a DNS name to address the proxy. So its definitely not just Sonicwall. Steve ___ -- 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] Astribank loop current adjustment
2008/10/23 Udo Schacht-Wiegand [EMAIL PROTECTED] For a door opener on an Astribank FXS port we need a loop current of 24.5mA . It does not function with the Astribank now, the dialtone becomes quiet immediately after pressing the button on that device. I've seen a limit of 23mA in the zaptel source. Is it possible to change the loop current of the Astribank somehow? You need to ask Xorcom that question. They have the ability to adjust any registers in the FXS port chip so I expect that they can do that. Steve ___ -- 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] Amazing show uptime
2008/9/12 Justin Coffi [EMAIL PROTECTED] Does your box run on the Mr. Fusion power supply? My box is plugged into the national power network. Oh right - I'm in South Africa and the national power supply company is Eskom. I see your point that it couldn't have been up 38 years. For completeness, I'm sure that the best explanation is that the clock was 1970 when Asterisk started. Its running on a little Xorcom XR1000 unit which probably does not have a cmos clock. Steve ___ -- 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] Amazing show uptime
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 ___ -- 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
2008/9/12 randulo [EMAIL PROTECTED] On Fri, Sep 12, 2008 at 11:13 AM, Tim Panton [EMAIL PROTECTED] wrote: I'd guess the battery on your motherboard has died so it is going back to 1970 at boottime. Why do hide the truth, Tim? It's much more likely the motherboard traveled back 38 years in time, is it not? Why don't you guys believe that my Asterisk has just been up for 38 years? asterisk -rx 'show version' Asterisk 0.0.1 built by root @ uunet!olsa99!cstat on a ENIAC running No OS At All on 1968-09-11 16:56:34 UTC Steve ___ -- 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
2008/9/12 Doug Lytle [EMAIL PROTECTED] Stephen Davies wrote: Why don't you guys believe that my Asterisk has just been up for 38 years? Because Mark was born in 1977 and he's 31. Oh dear. Maybe this will help: ;-) :-) Steve ___ -- 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-dev] Locking, coding guidelines addition
2008/7/6 Grey Man [EMAIL PROTECTED]: From what I can gather the suggestion from the FS approach is that each Asterisk channel should be handled after by it's own unique thread and save the need for any locking on the channel data structures in the first place. After a quick grep, there are lots of mutex locks and unlocks in the FS code. As you would expect. I guess Steve Totaro's grunt techs know that, whilst Steve has drunk the koolaid (and is trolling, anyway). Nevertheless - the suggestion as I understand it is that there is less contention for locks in FS due to the design choice that one thread is created that handles one active channel. I guess the theory is that _everything_ done on that channel is done in that thread. By contrast, we have a mix of design styles like the worker threads, network threads etc. But we don't have evidence that contention for mutexes (aka locks) is slowing Asterisk down. So it there is a big performance different it will probably be elsewhere - like the linked lists that are already getting attention. My curiosity is piqued to do a proper comparison of Asterisk and Freeswitch with a realistic workload and compare results (and profile Asterisk if there is a big difference. Steve ___ -- 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] IAX2 trunks unreliable becoming UNREACHABLE aftera time
I have a network of offices using Asterisk that are connected via IAX2 trunks. The trunks work great for a day or two then for no reason at all one end of the trunk will become UNREACHABLE while the other end is still connected. The oving nly way to fix the problem is to shutdown Asterisk completly then start it backup again. The end that dies is not always the same, some times it is server A and some times it is server B. Never have I seen that both ends die, just one. The side that is still connected can make calls to the end that died but not the other way. If you call from the server with the dead IAX2 trunk you here All circuts are busy now. All networks have static IP addresses and their firewalls are setup to allow UDP 4569 to come in to the Asterisk systems. We've got customers who experience this problem. We believe it is a fault with their NAT routers - I've traced traffic and our return packets do not arrive at their Asterisk box. Rebooting their NAT/ADSL routers fixes the problem. (Restarting their Asterisk does too - I assume the traffic from the box causes the NAT router to re-open a mapping. Steve ___ -- 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 scalability
I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Asterisk is highly multi-threaded and definitely takes advantage of multiple cores. There are a few places where concurrency could be further improved, but its really quite good in 1.4. (IAX in 1.4 does handle traffic using a thread pool so will take advantage of multiple cores). By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. Steve ___ -- 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] Calls Being Randomly Bridged
On 20/01/2008, Michael J. Liberatore [EMAIL PROTECTED] wrote: They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. In the SNOM settings there are two options that you should set to No. That is Call Join on Hangup and Xfer on Hangup. (Or names similar to that). Steve ___ -- 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] Calls Being Randomly Bridged
On 21/01/2008, Steve Davies [EMAIL PROTECTED] wrote: b) Attended. Wait for the call to answer, Press transfer, you will be ordered a list of calls in the order 3, 2, 1 - This is 100% wrong. The call you want is LAST in the list. If you have no CID, or have forgotten the CID of the caller, you cannot easily transfer the right call, and might instead connect the wrong caller. Why would you offer an unanswered call over an answered one anyway??? Yes - I completely agree that the SNOM attended-transfer is screwy in the presence of a third call. It causes problems if you have a long-running call and want to leave that on hold whilst handling another call that came in, or if a third call starts to ring in the middle of transferring a pre-existing call. Steve ___ -- 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] Re: [asterisk-dev] CDR changes in Trunk -- Transfers, CDRs, Life, and Everything
Hi Steve, Please look at my asterisk-dev post from a few minutes ago about dcontext and dst where the behaviour changed in a bad way in svn trunk recently. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk MS RTC Library Ethernet Capacity
On 08/06/07, Asterisk [EMAIL PROTECTED] wrote: Would a good 1 gBit switch be enough to handle that (Asterisk box would be connected to that switch with 1 gBit connection, and computers with Microsoft RTC Library would be connected with a 100 mBit connection)? Alex: 30 concurrent calls will be about 2.4 megabits in each direction. 10baseT would probably just about handle it. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
Hi Matthew:
Your environment sounds quite challenging and I'd be interested in the
analysis of what is limiting the throughput.
I agree that there's no easy way to distribute and single queue across
multiple boxes.
But here is a scaling idea for you. We've used it successfully to
handle a large inbound call centre. It also provides resilience:
1) Incoming PRIs connect to multiple boxes that we'll call the voice gateways.
Each box can have a proportion of your PRIs connected. Depending on
the box power, up to 8 or so.
2) Agent registrations are spread across these same boxes.
3) Lastly you define two or more additional boxes as your queue servers.
Every queue server has defined on it all the queues you need. But for
each queue one server is regarded as the primary and the other as
secondary. You mix things up so in the normal event about half your
queueing calls are on each server (extend the idea for more than 2
queue servers).
Incoming calls on the voice gateways are sent to the Queue server over IAX:
exten = 1234,1,Dial(IAX2/primary1234/${EXTEN})
exten = 1234,n,Dial(IAX2/sec1234/${EXTEN}) ; if we can't get to the primary
Now when an Agent wants to login, you have their agent gateway log in
to both of the queue servers on their behalf, using an IAX2/.. channel
to get back to the agent's voice gateway.
So on the queue server we have the agents for the queue logged in as
say IAX2/voicegw1/6001, IAX2/voicegw2/6002 etc etc.
The trick is to use transfer=yes aka notransfer=no on the various
boxes. So as soon as the call gets connected to an agent it
disappears off the queue box completely.
The nett result is that the queue servers only have to handle
customers who are still in the queue. As soon as they get connected
to an agent the call is directly from the arriving voice gateway to
the agent's voice gateway and on to the agent. In a proportion of the
time that even turns out to be the same box.
You can scale up the number of voice gateways as required and handle
1000s of calls connected to agents without needing supercomputers.
You still handle all the people queueing on a particular queue all on
the same queueing server. So you can tell them where they are in the
queue and all that. But you can split up your queues across multiple
boxes to help divide and conquer the load.
If you can reach the agent phone directly using IAX (use an IAX
softphone or something) you can make a little optimisation and log
IAX2/agentipaddress into the queue directly. Then the call gets
optimised to go directly from the incoming voice gateway to the
agent's PC.
Resilience? If a queue server is down, new callers will automatically
start to queue on the backup box for the queues affected. The agents
are known on both primary and backup queue boxes so things keep going.
If a voice gateway goes down you lose just some of your PRIs, so you
are still in business. If you need the capacity, use an ISDNguard to
kick the PRIs onto one of the other voice gateways. Agents that were
on the voice gateway that went down will need to reregister to a box
still running. IP address takeover can make that happen.
For me this sort of design is much better than one giant box.
Regards,
Steve Davies
Technical Director
Connection Telecom (Pty) Ltd
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Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes
On 01/06/07, Douglas Garstang [EMAIL PROTECTED] wrote: I previously worked for a company that did some heavy load testing with Asterisk on multiple core Sun systems. We saw that no matter how many cores you threw at Asterisk, it always used ONE core to process calls, even at very high loads. This is definitely not true in the general case. But using IAX2 prior to 1.4 does have a limit like that because all network traffic is handled in a single thread. Take a core dump of a working Asterisk box and count all the threads. There's no general lack of multi-threadedness, that's for sure. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
On 01/06/07, Matthew J. Roth [EMAIL PROTECTED] wrote: Mon Apr 2 12:15:01 EDT 2007 Idle (sar -P ALL 60 14) (60 seconds 14 slices) Linux 2.6.12-1.1376_FC3smp (4core.imminc.com) 04/02/07 12:24:01 CPU %user %nice %system %iowait %idle 12:25:02 all 14.97 0.03 34.25 0.92 49.82 12:25:020 8.83 0.05 33.60 1.28 56.24 12:25:021 17.50 0.02 34.60 0.57 47.32 12:25:022 19.94 0.02 33.52 1.31 45.22 12:25:023 13.62 0.02 35.29 0.52 50.55 Thu May 10 15:30:01 EDT 2007 Idle (sar -P ALL 60 14) (60 seconds 14 slices) Linux 2.6.12-1.1376_FC3smp (8core.imminc.com) 05/10/07 15:38:01 CPU %user %nice %system %iowait %idle 15:39:01 all 2.47 0.01 48.29 0.00 49.23 15:39:010 2.92 0.00 53.17 0.00 43.91 15:39:011 2.98 0.00 48.68 0.02 48.33 15:39:012 2.47 0.02 48.61 0.00 48.91 15:39:013 2.27 0.00 48.35 0.00 49.38 15:39:014 2.38 0.02 47.38 0.00 50.22 15:39:015 2.37 0.02 46.94 0.00 50.67 15:39:016 2.23 0.02 46.63 0.00 51.12 15:39:017 2.17 0.02 46.54 0.00 51.27 Have you got, or could you install oprofile? That will give you a LOT of information as to where your CPUs are spending their time, One guess is that you could be hitting contention in the kernel with all the cores contending for some scarce resource. So your cores can't execute because they are waiting on some kernel mutex for access to some resource. That would account for the increase in system time - oprofile would show where in the kernel they are spending time (where those 50%ishes are going). Steve Uhler at Sun has been studying this on his big multi-core Sparc boxes so he can probably contribute some insight. Hope you don't mind a cc, Steve. We're talking about Asterisk/Linux running out of scaling on an 8 core box. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
Hi, I want to quickly mention that I've had great success with running Asterisk in the under-appreciated Linux-VServer environment. This is not so much a virtualisation environment as a system partioner on steroids. Nothing to do with running windows on Linux and suchlike, but a good way to run lots of Asterisk and other stuff isolated from each other. There is only one kernel, and hardware is not virtualised. A particular guest We run about 10 Asterisk instances, together with web servers, Mysql and more. All on a 1GB RAM Pentium D box. CPU and memory to spare. hildegard steve # vserver-stat CTX PROCVSZRSS userTIME sysTIMEUPTIME NAME 0 82 266.5M 20.6M 21h19m47 8h04m51 26d02h24 root server 8 63 710.6M 1.4G 14m10s98 5m31s37 26d02h20 ctel_web 9 13 1.1G 31.8M 16m28s43 14m02s46 25d05h57 ctel_pbx 10 19 694.8M 172.7M 2h01m43 19m59s65 20d07h58 voipconnect 11 8 927.9M 98.9M 40m43s42 10m09s30 26d02h21 ctel_admin 12 5 210.5M 21.5M 5h13m35 36m41s61 26d02h21 ctel_db 13 8 903.4M 55.2M 3m00s10 1m09s40 26d02h20 ctel_intranet 15 5 213M 1.2M 9m09s35 12m51s00 26d02h19 xconnect 33 29 261.2M 22.7M 0m07s65 0m09s13 3d08h43 testtrunk 56 13 1.1G10M 1m44s45 1m18s94 26d02h21 aaa 57 13 1.1G 13.9M 8m47s52 8m46s10 26d02h21 bbb 58 13 1.1G 16.1M 54m29s99 24m46s22 26d02h14 ccc 60 9 293.4M 31.9M 10h44m42 2h00m55 26d02h20 ddd 61 13 1.1G 6.7M 12m45s11 13m26s42 26d02h19 eee (26 days uptime? Our hosting provider had to do power maintenance. We've never had a crash of the host system). My zttest inside a guest machine: voipconnect zaptel # ./zttest Opened pseudo zap interface, measuring accuracy... 100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00% 99.987793% 100.00% 100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00% 99.987793% 100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00% --- Results after 31 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.995277 Try that with Xen or VMWare. http://www.linux-vserver.org/ (Our host is hardened gentoo with PaX and GRSecurity, plus vserver; guests are gentoo too, though VServer does support guests being different distributions). Hope this helps someone, Steve Davies ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
On 14/05/07, Salvatore Giudice [EMAIL PROTECTED] wrote: Try switching to a Sangoma card. You won't have anymore IRQ issues once you abandon Digium hardware. Its not true, by the way. I've assisted more than one person using a Sangoma who was having issues caused by interrupt stuff. And it was the same sort of things that might affect a Digium board- motherboard raid disabling interrupts, sharing an IRQ with a heavy-interrupting LAN card, etc. Not suprising since its the same underlying problem - excessing interrupt handling latency. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Poor man's High Availability solution
On 29/04/07, Noah Miller [EMAIL PROTECTED] wrote: I've heard of a device that acts as a failover for a PRI line so you can plug a PRI into two different devices and have the PRI failover if one device fails. Unfortunately nothing like this is commercially available today. Sounds like the ISDNguard: http://www.junghanns.net/en/ISDNguard_produkt.html Aha! Thank You! I've wanted something like this for quite some time. A question: Does this require BRIStuff? Yes, but no. Installing BRIstuff gets you the heartbeat stuff. But it took me 5 minutes to extract that from BRIstuff and use it by itself. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [asterisk-users] WIFI SIP- The Best phone
On 09/01/07, Nigel Kendrick [EMAIL PROTECTED] wrote: I've had a play with a Nokia E70 - the 'bar' version of the E61 and gave up! Menu navigation is dire - I went through hoops trying to get SIP working - I know from others it can be done, but I bailed out when I realised that to put these phones in the hands of inexperienced users would be a recipe for a lot of frustration and support calls. Ironically I was going to recommend the E70. It is true that the menus are complex but once configured it does do what it says on the tin - provide a very effective merging of SIP over WIFI and GSM all in one unit. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3k wired to PAP2 for echo testing
On 05/11/06, James Harper [EMAIL PROTECTED] wrote: Even in this configuration, with my impedance settings set to the Australian standard of 220+820||120nf, and the PSTN and PAP2 echo cancellers enabled (or not, and all combinations of) I get local echo as soon as I pick up the handset (I hear my voice bounced back to me). Surely this shouldn't be??? There is no hybrid involved at all! 'course there is. The telephone interface on the one end and the line interface on the other are both 2 wire. Did you have a phone line connected to the other side. Running into an unconnected FXO port is likely to make echo because of the unbalanced impedance. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [asterisk-users] SIP v IAX2
On 26/10/06, Guillermo Salas M. [EMAIL PROTECTED] wrote: What about the bandwidth used for both protocols? Is IAX using less or more bandwidth than SIP? I'll give you an actual measured result. A trunked IAX2 link, carrying 30 simultaneous calls using variable-bit-rate Speex - we saw 7 kilobits / call / second. That's INCLUDING all IAX2, UDP, IP overheads. That's the magic of Speex VBR and trunking. Its much much much less than you can do with SIP. Better even than any of the proprietary boxes with packet-saver technology and the like when using a codec with quality comparable to Speex. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [SPAM??] Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.
On 02/11/06, Matthew Mackes (Webmail) [EMAIL PROTECTED] wrote: As far as Snow- They look very cool, and I love almost everything Linux based- PDA's, PVR,s, everything- but, I wonder if it will need to be rebooted every once in a while to stay happy- Every phone that is SIP has an OS- so, its hard to say if a Linux phone is more stable then a simpler phone. I translate that as no - haven't looked at the Snom. I think you should. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations
On 15/06/06, Douglas Garstang [EMAIL PROTECTED] wrote: Who said I was a C programmer? Speaking for myself, I just assumed that you understood that the behaviour of an open-source application was the result of contributed code. Your message read to me like something of a demand that someone fixed it. You are probably trying to do something pretty fancy in your dialplan and that probably brings requirements that the original authors didn't foresee. They are scratching their itch. As you said, DUNDi was Mark's initiative to make a open access call routing system, rather than to do with failover. If you can hack Asterisk dialplan code, then I think if you open that file, take a look at other code that sets variables (search for a variable name you know is set, like DIALSTATUS), do some cut and paste and you'll discover that, guess what: you ARE a C programmer. If you can't, well lots of us on the list take contracts for development in the Asterisk code. Post on asterisk-biz with the request. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN call-progress IE in SETUP frames
Hi, I have a strange problem on a single customer's PRI. He can't call certain destinations, receiving an incompatible destination ISDN cause code back from the network. I'm sure that the PRI is misconfigured by the telco; but they (as always) insist there is nothing wrong. Another Asterisk system with identical config works perfectly on a PRI installed at the next-door sister company and going to the same telco switch and all. The problem PRI was reconfigured at the request of a previous vendor who installed a toll-bypass box. Anyway - examing the ISDN traces from the Asterisk box and comparing to what the PBX sends, I see that the old PBX includes a calling equipment is non-ISDN progress IE in its call setup. Asterisk doesn't. Is there any way to collect this value on incoming calls. That is, as we handle an incoming call to query to retrieve the value of this progress IE? And any way to set it when we make outgoing calls? If I can do this then it will be easy to pass this value back and forth and make us more transparent. Matt, Mark, Kevin: there do seem to be some issues with the callprogress IE handling in libpri - for instance, libpri remembers them using a bit-map, but the transmit_call_progress will just send ONE of those remembered. Does anyone want to help me understand as I make adustments... Regards, Steve Davies ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialplan AGI DTMF
On Thu, 27 May 2004, Vladyslav wrote: Good day All. Is there a way to pass DTMF signals to AGI script during conversation ? Actually here what I want to make: Users are usually dial using dialplan and when someone press *4 (during conversation) I want to have agi script to deal with that, but those users should keep talking and even didn't notice that one of them press something. Is there a way to do that or it's complete nonsense? I've been mentally scheming about a way to do this is a generalised way - but right now during conversation the only DTMF that may be detected is a * for disconnect and # to initiate a transfer. Even these are only handled if the right dial options are used. To get what you want you will need to change the source code - in res/res_parking.c, function ast_bridge_call. Steve ___ 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] Downgrading Asterisk
On Tue, 25 May 2004, jo wrote: Sorry, no solution but same problem. Downgrading brings this message on Suse9.0, 2.4.21: [app_txtcidname.so]May 25 23:28:42 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/app_txtcidname.so: undefined symbol: ast_get_txt May 25 23:28:42 WARNING[16384]: loader.c:408 load_modules: Loading module app_txtcidname.so failed! app_txtcidname.so is left over from your test of the new version. Delete it. Better - delete everything in /usr/lib/asterisk/modules and re-make install the version of Asterisk you want to use. Steve ___ 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] fax/sandsp segfaulting asterisk
On Tue, 25 May 2004, Dan Cunningham wrote: Like some others on the list spandsp is segfaulting asterisk when recieving a fax. I'm on debian testing/unstable with freshly checked out asterisk CVS and sandsp. My libtiff version is 3.6.1. You need an older libtiff - v3.5.7. Steve ___ 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] spandsp hylafax asterisk and confusion
On Tue, 25 May 2004, Terry Goodwin wrote: Thanks for offering to help with this. I checked out the procedures and attempted this again without success. Here is the end of the screen output when the compile fails. gcc -02 -g -Include -I ../include -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: 'PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make[1]: *** [app_rxfax.o] error 1 make[1]: leaving directory '/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Ah - I remember this. There may be other fixes, but I resolved this by adding: #ifndef _GNU_SOURCE #define _GNU_SOURCE #endif Just before the #include pthread.h in asterisk/include/asterisk.lock.h Regards, Steve Davies ___ 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] Asterisk Prepaid
On Mon, 24 May 2004, usedcanon wrote: I have a requirement for a setup with prepaid call credits. I am aware of the two applications available (been researching for the past week), app_prepaid and app_rateengine. However neither of the two sound like exactly what I want. However I was wondering that someone who has used it might be able to say if they could be used in my scenario. Basically my scenario is pretty straight forward. Credit will be allocated to the ddi, I dont need any announcements etc (maybe low credit warning during call could be useful thoug). From the users prespective everything will be transparent. However the call should disconnect when the credit runs out. The CDR and the account DB need to be adjusted according to the call made. My guess is that app_prepaid could used with modification, I am assuming here that this is not possible as-is with configuration. Basically in case of the prepaid app, the card number can be replace transparently with the callerID. Hi, I did this to app_prepaid - you can pass a parameter into Prepaid() - its looked up in a table to find an associated card number - if that is found then the card number prompt is skipped and the associated card is used automatically. I can send a patch if you like (will also include a minor change or two to have app_prepaid work against CVS. Steve ___ 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] 2 Sip phones behind un-natted Asterisk
On Mon, 24 May 2004, Chad Brown wrote: 1.The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2.I have verify=yes in the sip.conf for both phones. Both phones constantly go Unreachable. (However, the connection is very fast between * and sip phones) 3.Sometimes but not always when I try to call phone1 phone2 rings. Is this Nat messing with me or something else? In any case...Any advice out there? Yes - I think your NAT firewall is messing with you. I suspect that if you configure the two phones in different ports - IE move one away from 5060, then you'll probably unconfuse your firewall. Steve ___ 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] Caller ID with BT CD50
On Sun, 23 May 2004, gARetH baBB wrote: On Sun, 23 May 2004, Karl Dyson wrote: Of course, although my wife is happy with the Cisco 7905s that have sprung up around the house, she still likes the cordless DECT units we have, and so they're plugged into an ATA186. Problem is, they no longer display caller id due to the ATA186 not poking it out in BT format I guess. If I were to buy some US cordless handsets would they do the ATA186 firmware 3.0+ supports more formats. Certainly with 3.0 my BT DECT 3010 (rebadged Siemens) base copes fine. I also found my Philips DECT phone in the UK had a few CID modes of which one worked with the ATA. Only downside was CID only displayed after the first ring-ring. Steve ___ 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] uClibc patch?
On Wed, 21 Apr 2004, Jeremy Jones wrote: I've been searching on an error I'm getting trying to compile against uClibc, related to enum support. I found reference in an earlier thread (http://lists.digium.com/pipermail/asterisk-users/2003-June/014176.html) to a patch adding an Makefile option to remove enum support. Anyone have that diff file lying around? Its in bugs.digium.com Steve ___ 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] ** WANTED: FreeBSD or OpenBSD programmer
On Wed, 21 Apr 2004, Tom wrote: It doesn't look very hard. FreeBSD supports recursive mutexes. It is just a matter of getting the appropriate defines. I'm going to look at this. On my Gentoo system I had to add #define _GNU_SOURCE to lock.h just before it #includes pthread.h. That enabled recursive mutexes. Steve ___ 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] X100P and NTL (ex Cable + Wireless)
On Tue, 13 Apr 2004, Alex Brett wrote: Has anybody got any experience using an X100P on an NTL phone line in the UK (I'm in an ex Cable Wireless area if that makes any difference). The problem I'm having (and judging by the number of references to it I've found searching it is a common one) is getting * to detect when the line has been hung up. It doesn't matter if it comes through to a person directly as they can just hang that phone up, but when it hits voicemail, and it sits there for two minutes recording an empty message, and then emails it to the person it can be a bit annoying! Hi Alex, Indeed the call end termination doesn't work on an NTL line. I'm not so sure it works too well on other lines either. I did some work a while back to add detection of the UK busy/hangup signal on the line, but I never got it working well enough to depend on it. The problem is that it is a single frequency tone. (The US one is dual-tone). Women's voices used to sometimes trigger my detector - causing hangups. The main practical issue is with voicemail, as you say. My final solution was to switch to ISDN. Steve ___ 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] X100P and NTL (ex Cable + Wireless)
On Tue, 13 Apr 2004, Vic Cross wrote: On Tue, 13 Apr 2004, Stephen Davies wrote: I did some work a while back to add detection of the UK busy/hangup signal on the line, but I never got it working well enough to depend on it. The problem is that it is a single frequency tone. (The US one is dual-tone). Women's voices used to sometimes trigger my detector - causing hangups. I'm looking at the same thing now, for AU busy tone. If there's some work-in-progress that you wouldn't mind releasing, I'd be keen to have a look. I think the problem with the current code (for us!) is the short length of time over which it tests for busy. Extending this might help prevent voice-off. It will be a balancing act though, as down here the ringing indication is the same frequency tone (and I'd rather not have my outgoing calls detected as busy when they are actually ringing). Hi, I did have my code test for the hangup tone over a longer period. This is the tough one as * has to listen all the time to the call to watch out for it. In the UK ringing and busy are different, which does make a difference. Anyway - I've sent my patch to you separately. It may not apply to current Asterisk, but hopefully will be useful anyway. Steve ___ 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] SoftFAX/spandsp: installing and results on Gentoo
Hi, I've installed spandsp library an RxFAX app on my Gentoo based * server - here's a little report on the process. Firstly - my Gentoo box had libtiff 3.6.0 installed, but did not have /usr/include/tif_dir.h and tiffiop.h. First thing I did was to copy the two missing headers from ftp.opencall.org. The result of that was seg faults when rxfax started trying to write the tiff file. I presume the headers on ftp.opencall.org don't match the 3.6.0 release of libtiff. The second thing I tried was to update my gentoo box to libtiff 3.6.1 - I emerged that, then also downloaded the source and from that installed the extra headers. That got rid of the seg fault, but the generated tiff files were now almost completely black with just a couple of stripes instead of the faxed image. I then emerge unmerged tiff, and downloaded and built tiff-3.5.7 from source manually. Rebuilt spandsp and asterisk and bingo rxfax now works. My conclusion - spandsp seems to have a compatibility issue with tiff-3.6.1. (Unless my install approach was broken). And - you may want to make the two missing header files available for more than just 3.5.7 of libtiff. I'm really impressed with the end-result - awesome to do that all in software! Regards, Steve Davies ___ 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] Packet8
On Sat, 20 Mar 2004, Zac Amsler wrote: I know this issue has been address before, but I can not find someone who has the answer. I am trying to get my * server to authenticate directly to packet8. I was very close to them actually giving me the information and possibly using them for my SIP - PSTN termination, but that fell through. They didn't think they had enough bandwidth. (LOL) There are a few questions that I would like to know answers to. - Does anyone currently have a working implementation in which asterisk authenticates to pakcet8? (Making and receiving calls via packet8) If so, could you please share? Hi, I used to use * with Packet8 - it took some fixes to the * SIP implementation but those are in the CVS long time ago. But then Packet8 started sending emails complaining about my foreign UA software and threatening disconnection. I suppose this was to do with stopping people pumping millions of minutes through one flat-rate account. Ironically, I was on the per-minute rate. Anyway - I disconnected and concluded that Packet8 didn't want to deal with us. No loss to us - providers like Nufone and Magrathea and others are there to take our business. Steve ___ 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] PCI front mount chassis?
On Fri, 12 Mar 2004, Brian Capouch wrote: I too am running 6 cards in my system, although not in a high traffic capacity load environment. So far my (limited) high-load simulations have shown no problems. So - is it apocryphal that the Digium cards (drivers) won't share interrupts? If there is a real issue with sharing interrupts then it seems to me to be a bug that needs fixing. PCI bus supports shared interrupts, why doesn't the hardware/driver? Yours curiously, Steve ___ 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] Outbound Transfer and the # key
On Wed, 10 Mar 2004, Fran Boon wrote: Patch failed - this is what this output is showing. As Matt said the patch needs modifying to patch cleanly against the current version of the code... You didn't read his mail properly. Steve ___ 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] ENUM when your country's ITU representative is uncooperative
Hi, I thought it would be neat to put my SIP/IAX reachable systems into the ENUM system. But reading about it I see that its rather centrally controlled within the ITU. My country code (+27) is not delegated. My country has a monopoly telco whose only interest in VOIP is to keep it all to themselves and not permit any other usage. So - what to do? If I approach the administrators for e164.arpa ([EMAIL PROTECTED], apparently) will they delegate 7.2.e164.arpa to me? I guess that they won't. (It would be fun if they would, for some definition of fun (I once administered .mu and the Mauritius telco thought THEY should administer it)). Considering that they probably won't delegate, how about Asterisk supporting a second parallel ENUM tree under a domain that we can control ourselves? Thanks for any comments, Steve Davies ___ 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 the registration traffic
On Tue, 17 Feb 2004, Rich Adamson wrote: I have an full-time Internet connection with a limited amount of traffic per month included in the subscription. What can I do to reduce the registration traffic with IAXTEL which makes about 10MB/h? There is any way to keep the registration active ( I have a static IP address) without the need to register all the time? Dan, Sounds like something is not set up correctly or something else is happening. I just used a sniffer to qualify my connection, and over a 24 minute period there was a total of 13,379 bytes of traffic (no calls). Analyzing the actual packets indicates five packets occur every 49.7 seconds, and those packets (with headers) contained a total of 426 bytes. If I did the math correctly, that would suggest 10 megabytes of traffic every 13.5 days (not per hour). Rich Hi, Perhaps its that Dan's box is trying to register with IAX1? All the attempts cause quite a flurry if the other side doesn't want to know. On my system I used noload = chan_iax.so in my /etc/asterisk/modules.conf to get rid of iax1. Steve ___ 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] I finally did IT!!!! Internal dial tone
On Tue, 10 Feb 2004, Alex Lopez wrote:
[outsidedialtone]
exten = s,1,Playtones(350+440) ; US standard dialtone from indications.conf
exten = _X,1,SetVar(FIRSTNUM=${EXTEN}) ; Had to get the first digit dialed
and hold on to it!!
exten = _X,2,StopPlaytones()
exten = _X,3,Goto(outgoingdial,s,1)
Hi,
Interesting work-around - but you could instead use the
PlayInterruptableTones command that I sent in as a patch a while back
- check the bugs.digium.com.
Regards,
Steve
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Re: [Asterisk-Users] iax2 jitter buffer help
On Thu, 5 Feb 2004 [EMAIL PROTECTED] wrote: Hi I wonder if anyone has a fix or any advice for the IAX2 jitter buffer. My internet connection here in South Africa has an international ping time of 550ms +- 50 ms. According to the scientific approach I would like to add a 100ms jitter buffer. (nevermind the latency)! I have tried playing with maxjitterbuffer and maxexcessjitterbuffer settings, I also tried from the CLI IAX2 set jitter 700 with all kinds of parameters. Hi Clive, Are you on a Telkom ADSL line? I've found it unusable for VOIP over the last two weeks - simply not enough throughput. Its only a few prioritised ports (eg port 80 - web, 21 - ftp) that have any decent throughput. Steve ___ 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:[OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help
On Thu, 5 Feb 2004, Chris Lee wrote: On the subject of South Africa What are the laws regarding using the Internet to carry telephone traffic? What are the laws regarding connecting digium kit to Telkom equipment? As I recall they are quite restrictive, have they been eased up a bit? The law is still very restrictive. Equipment should be ICASA approved for connection to the network. Digium equipment isn't. VOIP may be used on private networks. However such use is for office-to-office calls, and may not be used to bypass Telkom. This is generally understood to mean connecting in from the PSTN and then breaking back out again. Even VOIP on private networks is supposed to be dependent on getting a private telecommunications licence. In SA a private network means a network built out of Telkom data circuits. No actual private commmunications links are allowed. VPN-type networks are not included. Value Added network providers - including ISPs and suchlike are not supposed to allow the use of their service for transporting VOIP, and certainly may not market services like that. Of course they don't know and I'd guess they don't ask. Technically I guess using services like Vonage or whatever from SA is questionable too. Of course South African's have developed a certain attitude to the law, and enforcement is difficult, especially for small-scale private use. For example type-approval of equipment seems to be pretty much overlooked - see no evil, hear no evil. I'm no lawyer and perhaps Telkom/ICASA/Dept of Communications' interpretations of the law are wrong - I don't think they've really been tested in the courts. I also may have got some of the subtleties slightly wrong. You might ask why a country which could benefit so much from communication innovation has such restrictive law. It's a sad story of money, power and influence. You can read an interesting article on the SAT3 undersea cable and communications in Africa at: http://www.myadsl.co.za/forum/topic.asp?TOPIC_ID=1635 Regards, Steve Davies PS: 512k down / 256k up ADSL, capped at 3GB total inbound+outbound traffic, brutal traffic shaping which (coincidentally?) often breaks VOIP: +/- US$120 per month to you, sir. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] determining legal VoIP service
On Fri, 30 Jan 2004, Dustin Goodwin wrote: Actually I believe this is one of the few things that can be done without worrying about the state(s) PUC coming down on your head. Since your users are in another country the state PUC cannot consider you providing a telephone service in their jurisdiction. On the other hand, this is quite likely not allowed on the Nigeria and Ghana ends. That's the way it is in my country - South Africa - anyway. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Internal Lines Dialing Out
On Fri, 30 Jan 2004, Steve Rodgers wrote: Oops! I forgot the leading underscore. Use this version below. Steve. exten =_ NXX,1,Dial(Zap/1/$EXTEN) exten = _1NXXNXX,1,Dial(Zap/1/$EXTEN) And reaching us wot is in the rest of the world...? ;-) Steve ___ 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] running asterisk under root
On Thu, 29 Jan 2004, Dmitry Mishchenko wrote: All example of installing Asterisk shows running it under root user. Why is that? Can it be run under regular non-privileged user account. Sure - with the right permission tweaking. I made a group telephony. Had to fiddle with permissons and set group to telephony for stuff like /etc/asterisk /var/log/asterisk. And in my case /dev/capi20 etc. Steve ___ 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] Asterisk Indications
On Sun, 25 Jan 2004, Christopher Lee wrote: I've had a closer listen to 400*17 through the handpiece rather than just on speaker phone, and I get the feeling that the Australian ringing tone must have been tweaked slightly, perhaps with the introduction of the newer Ericsson AXE exchanges? 400*17 sounds familiar, perhaps the older exchanges (cross-bar?) used that format? That said, the 400+420 isn't exactly how my current exchange sounds, but sounds good to me anyway :-) I'm looking at tweaking the sounds somewhat more and moving away from the exchange sounds... I'd actually like to get it sounding more like a Nortel Meridian system, but I don't yet have any example rings to work off to try and get it similar sounding. Try the 383+417 and see how that sounds for you. Regards, the other Steve Davies ___ 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] Asterisk Indications
On Sun, 25 Jan 2004, Christopher Lee wrote: The original indications has 400+17/400, but I find that sounds more like two beeps (which could possibly be confused with the Australian congestion/busy tones). Shouldn't it be 400*17? Steve ___ 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] T1 Sync clarification
On Wed, 14 Jan 2004, Steve Underwood wrote: That must have been an FSK modem. Most advanced modems completely loose sync on the first sample slip. The sample slip causes a jump in phase, and phase is critical to the correct operation of most modems. It was V.22. No error correction or anything new-fangled like that. (Not auto dial either). Steve ___ 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] T1 Sync clarification
On Wed, 14 Jan 2004, TC wrote: What are the practical effects with in-correct clock sync -like to you hear odd buzzing, or dropped voice or gaps of audio ?? Old-fart anecdote about this - in the early 80s we had some 1200bps modems that we used to connect to client sites. When our phone company went digital we suddenly started getting a } character at a regular interval of 10 or 15 seconds. This turned out to be clock slips in the new digital trunk between the two exchanges. So there is one effect of clock slips. Steve ___ 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] More words for Allison
On Sun, 11 Jan 2004, John Todd wrote: zed Thanks! knots per hour Pretty sure the measure is just knots. IE 40 knot wind, or the wind will be 40 knots. Steve ___ 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] OT: Calculating Bandwith
On Wed, 7 Jan 2004, calvis wrote: I am trying to calculate bandwidth needs. Is 1 T1 Line able to provide 488.5 Gigabytes of traffic for 1 month based on a 30 day week? I did my calculation as follows: 1.544 mps * number of seconds in a minute(60) * number of minutes in a hour (60) * number of hours in a day(24) * number of days in a month(30) = 4002048 megabits / 1024 = (3908.25 gigabits) / 8 = (488.53125 gigabytes) of bandwidth for a T1 Line. Please correct me if I am wrong. Thanks, Charles Theoretically your calculation is fine - in fact the line will do that much in each direction, so twice as much in total. In practice for almost all real-world uses, you will find that user experience is intolerably bad before you get anythere near that theoretical capacity. This is because of bursty use of the line, and because of TCP's behaviour under contention. In my experience you should expect to get say 25-40% of the theoretical capacity per month with acceptable performance. Obviously it depends on how you are going to use the line. Regards, Steve Davies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unexpected ISDN hangup on outbound call
On Wed, 7 Jan 2004, Sjur Eivind Usken wrote: We have setup an asterisk box to let everybody call into the university internal network, but I get unexpected hangups when doing an outbound call from SIP to the ISDN interface, and it happens from 20 seconds to some minutes into the call. --the dial and the problem--- -- Executing Dial(SIP/57966-a19d, Modem/g1:96121||rt|) in new stack -- Executing Dial(SIP/57966-a19d, Modem/g1:96121||rt|) in new stack -- Called g1:96121 -- Called g1:96121 -- Modem[i4l]/ttyI1 answered SIP/57966-a19d -- Modem[i4l]/ttyI1 answered SIP/57966-a19d voipgk*CLI WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) -- Hungup 'Modem[i4l]/ttyI1' -- Hungup 'Modem[i4l]/ttyI1' - As far as Asterisk is concerned, it never finished establishing the SIP connection from the source phone. I suggest enabling SIP debug and have a look at the messages. Perhaps you have a NAT issue. You may like to dial say the Echo application on your Asterisk server to satisfy yourself that this isn't an ISDN issue. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] manipulating with numbers - StripMSD, Prefix
On Wed, 7 Jan 2004, Tilghman Lesher wrote:
On Wednesday 07 January 2004 06:06, Dawid Mielnik wrote:
Hello,
I can not seem to be able to get StripMSD and Prefix to work for me
in extensions.conf. This is an example of what I have:
exten = _050.,1,StripMSD,1
exten = _50.,Prefix,01051
exten = _001051.,1,Dial(${TRUNK2}/${EXTEN})
exten = _001051.,2,Busy
exten = _001051.,102,Busy
What I want to achieve is to call 001051501657887 on TRUNK2 when
dialing 0501657887.
Here's an idea - don't use StripMSD and Prefix anymore, as there are
better options now:
and don't leave out the priority in the second exten line.
Steve
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Re: [Asterisk-Users] FWD problems
On Wed, 24 Dec 2003, denon wrote:
I've been having issues getting FWD to work. I posted this same Q to the
FWD forum (no responses yet), but I was hoping someone here had some insight:
My setup is like this:
sip.conf:
register = 21542:[EMAIL PROTECTED]/6002 ; Free World Dialup
[fwd.pulver.com]
type=peer
host=fwd.pulver.com
fromuser=21542
fromdomain=fwd.pulver.com
username=21542
secret=password
In extensions.conf:
; Free World Dialup
[fwd]
exten = _10113.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _10113.,2,Congestion
(I use a 10113 prefix for FWD numbers).
We're chatting to friends in the UK right now so seems to work for me.
Steve
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Re: [Asterisk-Users] FWD problems
On Wed, 24 Dec 2003, Iain Stevenson wrote: I have exactly this problem and posted a bug report to FWD about a week ago - no response yet. It's bizarre that FWD recognises you to dial another user but not to call outside their network. Sounds more like a FWD problem than a * problem to me. Suspect your INVITE into FWD isn't authenticated so FWD thinks of you as a foreigner. Perhaps a sip debug will help see what is happening. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing dead SIP peers give misleading (BUSY) voicemail result ...
On Sun, 21 Dec 2003, Darren Nickerson wrote: In the case of a physically-disconnected ZAP extension, the Dial application succeeds, moving on to the next step in the dialplan. That is much more in line with my expectation. With an X100P card, diconnecting the card from the line results in attempts to dial out on it also giving BUSY. This gives similar issues to yours. Again, I think it should continue at the next priority rather then branching to the +101 one. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog phone not ringing
On Sat, 19 Jul 2003, Darren Poulson wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I've got my developers kit from telappliant and got a machine up and running to become the house phone system. Most things are working now, such as incoming calls, call transfer, call parking, voicemail, etc. The one thing I can't do is make my analog phone ring! I can see the call coming in on the asterisk console and can then pick up the analog phone, but no ringing! Incoming calls come in on the X100P (channel 1), the analog phone is on the only other channel from a TDM400P The one thing that I think it could be is the connector to convert from RJ45 to BT phone socket. I'm using a mod tap that I had lying around. Not sure what the wiring is like inside it. In the UK, phones use a three-wire connection with a separate line for the ringer. This is supposed to stop phone bells tinkling when another phone is dialled. Anyway - you need the RJ11 to BT adapter that includes the ringing capacitor. Maplin sells the adapters both with and without the cap. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P and PSTN caller id
On Thu, 26 Jun 2003, Dan wrote: There is nobody with an X100P in Europe having this issue related to the PSTN Caller ID? Please help! Well - my X100P doesn't pick up Callerid from my UK line. But I always assumed that it was just not compatible with UK-style Callerid. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] compile in uclibc enviroment
On Thu, 19 Jun 2003, Holger von Ameln wrote:
Hi,
Stephen Davis offered to send me a patch that leaves out enum support.
That would at least solve the undefined references to res_ninit,
res_nsearch and res_nclose in enum.c.
Cheers,
Holger
Hi,
Here it is, attached. Adds a setting in the Makefile where enum
support can be turned off.
There will probably be some offset when patching due to other changes
in my sources.
Steve
Index: Makefile
===
RCS file: /usr/cvsroot/asterisk/Makefile,v
retrieving revision 1.17
diff -u -r1.17 Makefile
--- Makefile17 Jun 2003 22:30:25 - 1.17
+++ Makefile19 Jun 2003 10:50:00 -
@@ -51,6 +51,9 @@
#
MALLOC_DEBUG = #-include $(PWD)/include/asterisk/astmm.h
+# Do you want ENUM support?
+ENUM_SUPPORT = #-DENUM_SUPPORT
+
# Where to install asterisk after compiling
# Default - leave empty
INSTALL_PREFIX=
@@ -85,12 +88,14 @@
INCLUDE=-Iinclude -I../include
CFLAGS=-pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations
$(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE #-DMAKE_VALGRIND_HAPPY
CFLAGS+=$(OPTIMIZE)
+CFLAGS+=$(ENUM_SUPPORT)
CFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null /dev/null
21; then echo -march=$(PROC); fi)
CFLAGS+=$(shell if uname -m | grep -q ppc; then echo -fsigned-char; fi)
ifeq (${OSARCH},OpenBSD)
CFLAGS+=-pthread
endif
+#CFLAGS+=-DSLD
#CFLAGS+=$(shell if [ -f /usr/include/linux/zaptel.h ]; then echo
-DZAPTEL_OPTIMIZATIONS; fi)
LIBEDIT=editline/libedit.a
@@ -125,7 +130,8 @@
ulaw.o alaw.o callerid.o fskmodem.o image.o app.o \
cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o \
dsp.o chanvars.o indications.o autoservice.o db.o privacy.o \
- astmm.o enum.o srv.o
+ astmm.o
+OBJS+=enum.o srv.o
CC=gcc
INSTALL=install
Index: asterisk.c
===
RCS file: /usr/cvsroot/asterisk/asterisk.c,v
retrieving revision 1.11
diff -u -r1.11 asterisk.c
--- asterisk.c 22 May 2003 14:24:06 - 1.11
+++ asterisk.c 19 Jun 2003 10:50:03 -
@@ -1339,10 +1339,12 @@
printf(term_quit());
exit(1);
}
+#ifdef ENUM_SUPPORT
if (ast_enum_init()) {
printf(term_quit());
exit(1);
}
+#endif
/* We might have the option of showing a console, but for now just
do nothing... */
if (option_console !option_verbose)
Index: enum.c
===
RCS file: /usr/cvsroot/asterisk/enum.c,v
retrieving revision 1.5
diff -u -r1.5 enum.c
--- enum.c 12 Jun 2003 12:48:57 - 1.5
+++ enum.c 19 Jun 2003 10:50:08 -
@@ -11,6 +11,8 @@
*
*/
+#ifdef ENUM_SUPPORT
+
#include string.h
#include fcntl.h
#include unistd.h
@@ -382,3 +384,5 @@
{
return ast_enum_init();
}
+
+#endif /* -DENUM_SUPPORT */
Index: loader.c
===
RCS file: /usr/cvsroot/asterisk/loader.c,v
retrieving revision 1.5
diff -u -r1.5 loader.c
--- loader.c16 May 2003 02:50:46 - 1.5
+++ loader.c19 Jun 2003 10:50:10 -
@@ -146,7 +146,9 @@
/* We'll do the logger and manager the favor of calling its reload here first
*/
reload_manager();
+#ifdef ENUM_SUPPORT
ast_enum_reload();
+#endif
ast_rtp_reload();
time(ast_lastreloadtime);
Index: srv.c
===
RCS file: /usr/cvsroot/asterisk/srv.c,v
retrieving revision 1.1
diff -u -r1.1 srv.c
--- srv.c 12 Jun 2003 22:14:03 - 1.1
+++ srv.c 19 Jun 2003 10:50:23 -
@@ -11,6 +11,8 @@
*
*/
+#ifdef ENUM_SUPPORT
+
#include string.h
#include fcntl.h
#include unistd.h
@@ -297,3 +299,5 @@
res_nclose(srvstate);
return ret;
}
+
+#endif /* ifdef ENUM_SUPPORT */
Index: channels/chan_sip.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.125
diff -u -r1.125 chan_sip.c
--- channels/chan_sip.c 18 Jun 2003 22:34:55 - 1.125
+++ channels/chan_sip.c 19 Jun 2003 10:50:50 -
@@ -664,6 +668,7 @@
portno = atoi(port);
else
portno = DEFAULT_SIP_PORT;
+#ifdef ENUM_SUPPORT
if (srvlookup) {
char service[256];
int tportno;
@@ -675,6 +680,7 @@
portno = tportno;
}
}
+#endif
hp = gethostbyname(hostn);
if (hp) {
strncpy(r-tohost, peer, sizeof(r-tohost) - 1);
RE: [Asterisk-Users] E1 in South Africa
On Wed, 18 Jun 2003, Bradley Greep wrote: Hello Tielman Koekemoer E1 is used in the world except for North America and one or two other places. It consists of 30 speech or data channels and 2 signalling (1 for framed signalling, and one for channel signalling) E1 is superior to the North American T1 system over here it's called E1 to differentiate from T1. South Africa uses E1. which according to a colleague is known as pcm30. (if he remembers correctly.) The E-100 Card should work but I doubt if it's telkom certified. (That's your problem) Here's Telkom's E1 product: http://www.telkom.co.za/isdn/isdn30.jsp Also, http://www.telkom.co.za/isdn/technical_protocols.jsp gives technical details. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug with SIP and indications?
Hi, I'm having trouble using Ringing with a SIP client. I'm trying to give the caller the impression that the line hasn't been answered, whilst listening for various extensions to be dialled. Here's is the extension: exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,DigitTimeout,3 exten = s,4,ResponseTimeout,5 exten = s,5,Ringing There are various extensions that can be dialled. For example, I have one that calls another SIP target. I've got two problems: The first is cosmetic:- is there any way to get the Ringing to work like Background, and turn off the Ringing indication once digits start being dialled? The second is a show-stopper, though: As soon as Asterisk tries to connect to another SIP destination, I see in the log: WARNING[27662]: File chan_sip.c, Line 957 (sip_write): Asked to transmit frame type 64, while native formats is 4 (read/write = 8/4) And Asterisk drops the connection. Any advice for me? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a few questions about sip implementation
1. 8.2.6.1 Sending a Provisional Response says that UASs SHOULD NOT issue a provisional response to non-INVITE requests. From my message yesterday * appears to be sending a SIP/2.0 100 Trying to X-Lite's REGISTER request before sending the SIP/2.0 200 OK message. Is this correct? Yes, that is what it is doing and and while it may not adherent to the exact reading of the RFC, I have seen several other proxies doing the same thing (examples: FWD's SIP proxy (Cisco?) does send 100 Trying but SER does not) so I will assume it's an awkward industry standard, though perhaps not exactly compliant to the RFC paragraph that you describe. But note that SHOULD NOT != MUST NOT. I assume this has been added as a clarification since the old RFC. They couldn't make it MUST NOT because of existing implementations that already did it. What I'm trying to say is that Asterisk doesn't fail compliance on that point (though no doubt there are places where it does!) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy message with call waiting?
Why not have dial just dial, then have applications like WaitForAnswer, WaitForDisconnect etc...? This would give more granularity to the call flow control and allow someone to get brave and write a WaitForHuman or whatever. Hmm... I can't think of too many instances where the functionality of the existing Dial application would need to be extended. In rather the same vein as your comments - I'd like to be able to announce info about the call to the CALLED party and give them options to accept, fwd to vmail, etc etc. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Linux traffic shaping to prioritise SIP/IAX traffic?
Hi, Has anyone done anything with the Linux advanced routing stuff to give SIP and IAX traffic priority? What I have in mind is a high-pri queue for voip traffic, all the rest in another queue that gives way to the VOIP stuff. Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Linux traffic shaping to prioritiseSIP/IAX traffic?
On 10 Jun 2003, Emanuele Pucciarelli wrote: Il mar, 2003-06-10 alle 17:19, Stephen Davies ha scritto: Has anyone done anything with the Linux advanced routing stuff to give SIP and IAX traffic priority? What I have in mind is a high-pri queue for voip traffic, all the rest in another queue that gives way to the VOIP stuff. When the tos option is set correctly (to nodelay), the default queueing in recent kernels already does that, because the pfifo_fast queue is used (if I recall correctly). But there is never any queue on my Linux box. It all storms out of the ethernet interface and gets queued up in my cable modem which doesn't know anything about tos settings. I did find the wondershaper script on www.lartc.org which looks like it will do what I need. Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Linux traffic shaping to prioritiseSIP/IAX traffic?
On 10 Jun 2003, Emanuele Pucciarelli wrote: That is not entirely correct. There is an output queue, and pfifo_fast is the default (see the LARTC Howto, 9.2.1.1). But you are right when you say you need something to slow down the data;the simplest choice should be addingthe Token Bucket Filter (9.2.2.2). OK - thanks for the pointers! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip channel driver causes asterisk to crash when talking to quintum A800
On Sat, 7 Jun 2003, Daryl Jones wrote:
I experienced the exact same symptoms but didn't have the confidence
to post my experience to this list because of my lack of experience with
Asterisk. I restored the June 1 version from CVS and the problem went away.
There's definitely a problem in code since June 1.
Well,
Here's a simple patch to fix.
Steve
Index: channels/chan_sip.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.119
diff -u -r1.119 chan_sip.c
--- channels/chan_sip.c 6 Jun 2003 00:06:52 - 1.119
+++ channels/chan_sip.c 8 Jun 2003 13:21:38 -
@@ -3322,11 +3327,12 @@
}
return 0;
}
+
static char *get_calleridname(char *input,char *output)
{
char *end = strchr(input,'');
char *tmp = strchr(input,'\');
- if (!end) return NULL;
+ if (!end || (end==input)) return NULL;
/* move away from */
end--;
/* we found name */
Re: [Asterisk-Users] busydetect and X100P hangups
On Sun, 8 Jun 2003, Brian Capouch wrote: FYI to anyone else who may be experiencing random hangups; I removed the busydetect=yes lines from the conf files on my asterisk servers, and haven't had a hangup since. I had done that once before and it didn't seem to have much of an effect, so I'm not breaking out the champagne yet. But so far over dozens of calls both made and received since I took that line out, I haven't had a single hangup. Yeah - I turned on busydetect on my X100P and went out for the day. I was not popular when I got home. The busydetect works by listening for busy-tone-like pattern of some sound and silence. As you can imagine, that's not too reliable. callprogress=yes works by listening for the actual busy tone. That works well for me - or at least its turned on here and people aren't complaining about hangups. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip channel driver causes asterisk to crashwhen talking to quintum A800
On Sat, 7 Jun 2003, shido wrote: This is the sip debug when the call went through Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Content-Length: 157 Content-Type: application/sdp CSeq: 1 INVITE From: sip:[EMAIL PROTECTED];tag=402ada92-5 To: sip:[EMAIL PROTECTED] User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP 64.42.218.146;branch=z9hG4bK-tenor-64.42.218.146-5 Quintum: 0c01030b0239380501 v=0 o=Quintum 4 4 IN IP4 64.42.218.146 s=VoipCall c=IN IP4 64.42.218.146 t=0 0 m=audio 10240 RTP/AVP 0 c=IN IP4 64.42.218.146 a=rtpmap:0 pcmu/8000/1 11 headers, 8 lines Using latest request as basis request Sending to 64.42.218.146 : 5060 (non-NAT) Capabilities: us - 4, them - 4, combined - 4 Non-codec capabilities: us - 1, them - 0, combined - 0 Funnily enough I've been looking at the same problem. Will get a chance to look a bit more tomorrow. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone know about callerid format used by NTL in Cambridge?
Hi, Does anyone know anything about the callerid format that NTL uses here in Cambridge, UK. This is the former Cambridge Cable, who is sometimes different from the rest of NTL. They did say that only some equipment works with their switch. I hoped that they might use US-style CID, which would work with the X100P, but it doesn't seem to come through. I do notice that my DECT phone bought here in the UK understands the callerid as put out by the ATA186. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Net2Phone SIP
On Mon, 2 Jun 2003, Mark Thompson wrote: I can use an ata186 to connected directly to n2p through sip.net2phone.com without any special settings. I can connect from * to iconnecthere, but, whatever I try from * to n2p produces SIP/2.0 401 Unauthorized (Can forward the full * sip log and ata186 log if it would help) It is normal for n2p to send back an Unauthorised if you send an unauthenticated INVITE. Asterisk should re-send the INVITE but this time authenticated. For that to work, the entry in sip.conf needs a username and secret. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dinosaur *
On Mon, 2 Jun 2003, Tilghman Lesher wrote: First, they're going to have to be MMX. The 133 might be, but the 75 is definitely not. Normally you don't want to go much below a 200MHz processor for a base-level system; you could certainly try something slower, but not without MMX instructions. Well - I was able to test on a Cyrix 133MHz. No MMX I'm sure. It worked OK for simple testing. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
