Re: [asterisk-users] audio glitches in conference
It looks to me that u are having clock synchronism problems due to the fact you are using Virtual Machine so u don't have an ISDN card generating clock. Are u using what was called ztdummie as clock source? Can't precise the name of it in chan_dahdi but u have it. What u report isn't new and is well known due to the fact u don't have a precise clock source for meetme.. You need to have chan_dahdi dummie... Hope it helps. Marco Mouta Enviada do dispositivo sem fios BlackBerry® -Original Message- From: Jeff Brower jbro...@signalogic.com Date: Wed, 24 Feb 2010 18:25:07 To: Jonathan Addlemanj...@redowl.ca Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] audio glitches in conference Jonathan- I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording (via eagi) after the conference has been established for half an hour or so, the stream received by the eagi script delayed by about 10 seconds. How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? -Jeff First, the preliminaries: I'm on a debian lenny system, using the 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was running xen, but I've shut down all the domU's to test if they were interfering, so now there's no sharing going on. I've been testing with a simple eagi script to grab the audio from the conference: #!/bin/sh cat /dev/fd/3 /tmp/audio.raw I've been testing it using the following dialplan extensions: [test] exten = testeagi,1,Answer exten = testeagi,n,Wait(3) exten = testeagi,n,EAGI(testeagi.sh) exten = testmeet,1,Answer exten = testmeet,n,MeetMe(testconf,1qd) exten = testsound,1,Answer exten = testsound,n,Playback(testbeep-asterisk) (testbeep is just 30 seconds of sine wave) I've been trying things like this: originate Local/testso...@test extension teste...@test The recorded audio plays back fine - no glitches. (an example is at http://www.vecotourism.org/audio17.wav) originate Local/teste...@test extension testm...@test originate Local/testso...@test extension testm...@test This does have the glitches. (an example is at http://www.vecotourism.org/audio18.wav) What could be causing this? And is there anything else I could be doing to debug it? Thanks. -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] Do i need install Dahdi or libpri ?
2010/2/25 Zhang Shukun bit...@gmail.com:
next ,i want to dial from asterisk to PSTN now. i have see the sample
in the extensions.conf relevent to PSTN as follow:
; If you are freely delivering calls to the PSTN, list them here
;
;exten = _1256428,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
;exten = _1256325,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325
I am pretty sure you need to dial through SIP technology not Dahdi.
Like Dial(SIP/telephonenum...@your-gateway-ip)
but above shows something about DAHDI card.
Wich you don't have.
my question is:
a, Do i need install DAHDI or libpri in my system?
Not to connect your Gateway through sip.
b, how to write in dialplan to realise connection to PSTN.
Thats a quite general question. But I guess the Dial command above
will lead you the right way.
Chris
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[asterisk-users] CDR duration/billsec
Hello list,
I'm having troubles implementing the ${CDR(duration)}
${CDR(billsec)} variables in this scenario:
PEER CALLS OUT -
CALL GOES TO PEER'S DEFAULT OUTGOING CONTEXT -
THE CALL IS SENT TO A MACRO AND GOES IN HANGUP -
THE CALL RETURNS TO EXTENSION h OF PEER'S DEFAULT OUTGOING CONTEXT (here I'm
trying to print the variable)
The problem is I'm always getting '0' from those variables. The
incoming calls aren't passing through a macro and they return the correct
billsec/duration value.
cdr.conf has this enabled: endbeforehexten=yes
asterisk version: 1.6.0.20
Any ideas, clues, suggestions on how may I get this to work?
Thanks in advance,
Alex
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[asterisk-users] Getting: Can't fix up channel from 5 to 7 because 7 is already in use, and pri_dchannel: Answer requested on channel 0/7 not in use on span 1
System have been working great for weeks, using an average 40 of 120 dahdi channels. But today, I suddenly see scary things like this: -- Moving call from channel 5 to channel 7 [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is already in use [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: Ringing requested on channel 0/7 not in use on span 1 -- Moving call from channel 7 to channel 12 [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 pri_fixup_principle: Can't fix up channel from 7 to 12 because 12 is already in use [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: Ringing requested on channel 0/12 not in use on span 1 -- DAHDI/4-1 is ringing [Feb 25 10:18:17] WARNING[17129]: chan_dahdi.c:10608 pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is already in use [Feb 25 10:18:17] WARNING[17129]: chan_dahdi.c:11680 pri_dchannel: Answer requested on channel 0/7 not in use on span 1 [Feb 25 10:18:22] WARNING[17129]: chan_dahdi.c:10624 pri_fixup_principle: Whoa, there's no owner, and we're having to fix up channel 5 to channel 7 [Feb 25 10:21:44] WARNING[17129]: chan_dahdi.c:10661 pri_fixup_principle: Call specified, but not found? What would be the reason for things like this to happen? And are they really just warnings (as it says), or actual errors, where something bad is happening to the actual calls, or calls not acknowledged? Håkon -- _ -- 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] audio glitches in conference
Thanks to Tzafrir for the above mentiong documentation. FYI http://docs.tzafrir.org.il/dahdi-linux/README.html A PBX system should generally have a single clock. If you are connected to a telephony provider via a digital interface (e.g: E1, T1) you should also typically use the provider's clock (as you get through the interface). Hence one important job of Asterisk is to provide timing to the PBX. DAHDI ticks once per millisecond (1000 times per second). On each tick every active DAHDI channel reads and 8 bytes of data. Asterisk also uses this for timing, through a DAHDI pseudo channel it opens. However, not all PBX systems are connected to a telephony provider via a T1 or similar connection. With an analog connection you are not synced to the other party. And some systems don't have DAHDI hardware at all. Even a digital card may be used for other uses or is simply not connected to a provider. DAHDI cards are also capable of providing timing from a clock on card. Cheap x100P clone cards are sometimes used for that purpose. If all the above fail, you can use the module dahdi_dummy to provide timing alone without needing any DAHDI hardware. It will work with most systems and kernels. You can check the DAHDI timing source with dahdi_test, which is a small utility that is included with DAHDI. It runs in cycles. In each such cycle it tries to read 8192 bytes, and sees how long it takes. If DAHDI is not loaded or you don't have the device files, it will fail immediately. If you lack a timing device it will hang forever in the first cycle. Otherwise it will just give you in each cycle the percent of how close it was. Also try running it with the option -v for a verbose output. To check the clock source that is built into dahdi_dummy, you can either look at title of its span in /proc/dahdi file for a source: in the description. Or even run: strings dahdi.ko | grep source: -- Marco Mouta On Thu, Feb 25, 2010 at 8:15 AM, marco.mo...@gmail.com wrote: It looks to me that u are having clock synchronism problems due to the fact you are using Virtual Machine so u don't have an ISDN card generating clock. Are u using what was called ztdummie as clock source? Can't precise the name of it in chan_dahdi but u have it. What u report isn't new and is well known due to the fact u don't have a precise clock source for meetme.. You need to have chan_dahdi dummie... Hope it helps. Marco Mouta Enviada do dispositivo sem fios BlackBerry® -Original Message- From: Jeff Brower jbro...@signalogic.com Date: Wed, 24 Feb 2010 18:25:07 To: Jonathan Addlemanj...@redowl.ca Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] audio glitches in conference Jonathan- I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording (via eagi) after the conference has been established for half an hour or so, the stream received by the eagi script delayed by about 10 seconds. How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? -Jeff First, the preliminaries: I'm on a debian lenny system, using the 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was running xen, but I've shut down all the domU's to test if they were interfering, so now there's no sharing going on. I've been testing with a simple eagi script to grab the audio from the conference: #!/bin/sh cat /dev/fd/3 /tmp/audio.raw I've been testing it using the following dialplan extensions: [test] exten = testeagi,1,Answer exten = testeagi,n,Wait(3) exten = testeagi,n,EAGI(testeagi.sh) exten = testmeet,1,Answer exten = testmeet,n,MeetMe(testconf,1qd) exten = testsound,1,Answer exten = testsound,n,Playback(testbeep-asterisk) (testbeep is just 30 seconds of sine wave) I've been trying things like this: originate Local/testso...@test extension teste...@test The recorded audio plays back fine - no glitches. (an example is at http://www.vecotourism.org/audio17.wav) originate Local/teste...@test extension testm...@test originate Local/testso...@test extension testm...@test This does have the glitches. (an example is at http://www.vecotourism.org/audio18.wav) What could be causing this? And is there anything else I could be doing to debug it? Thanks. -- Jon-o Addleman - http://www.redowl.ca -- _ --
[asterisk-users] Redirect call based on CLI???
This is a real 'newbie' type question, but I can't get my brain to work today. Is it possible to re-direct an incoming SIP call based on it's CLI? Ideally I would like to check incoming calls against a short whitelist (of say 20 numbers) and redirect to a different extension if there is a match. I would also like to redirect calls that fail to present any CLI (aka WITHHELD/UNAVAILABE) to a different extension At I guess I would suspect I could implement this - can someone give me a starting point? Many thanks Brian -- _ -- 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] SIP Configuration files for Cisco 7905G FW 3-08-12
Hi, Does anyone have sample configuration files for a Cisco 7905G to use with SIP/Asterisk ? I'm on Firmware 3-08-12 - is there a better release to run ? /S -- _ -- 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] OT: Problems with Linksys IP Phone SPA 942
On 2/25/10 6:50 AM, Tilghman Lesher wrote:
DHCP is designed in such a way that you can legitimately have multiple DHCP
servers on the same network. The first DHCP server which replies and meets
the DHCP client's requirements will be the server to which the client
registers. If the Linksys DHCP server is faster (or if you have several
switches and it replies to some hosts faster), then those hosts will likely
use the Linksys as their DHCP server.
You could technically avoid this situation by provisioning some DHCP option
that the Linksys does not and making all of your DHCP clients require that
option, but that takes quite a bit away from the zeroconf usage of DHCP.
Or you could set up a rule on your managed switch such that broadcasts to
UDP port 67 only hit the switch port on which your intended DHCP server is
located.
I've been successfully using the following to catch Linksys phones and
provide DHCP services only to them for the past few years:
class Linksys {
match if ( substring (hardware,1,3) = 00:0e:08 ) or (
substring (option vendor-class-identifier, 0, 7) = LINKSYS );
}
pool {
allow members of Linksys;
deny dynamic bootp clients;
range 10.168.172.100 10.168.172.250;
}
HTH,
Vahan
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Re: [asterisk-users] Redirect call based on CLI???
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf Has example exten = s,1,Answer exten = s/9184238080,2,Set(CALLERID(name)=EVIL BASTARD) exten = s,2,Set(CALLERID(name)=Good Person) exten = s,3,Dial(SIP/goodperson) for white list exten = s/123123123,1,Dial(SIP/phoneA) exten = s/456456456,1,Dial(SIP/phoneA) exten = s,1,Dial(SIP/phoneB) also look around for an example using gotoif, and read the CALLERID(name) before doing setting it On Thu, Feb 25, 2010 at 2:17 AM, Brian brel.astersik100...@copperproductions.co.uk wrote: This is a real 'newbie' type question, but I can't get my brain to work today. Is it possible to re-direct an incoming SIP call based on it's CLI? Ideally I would like to check incoming calls against a short whitelist (of say 20 numbers) and redirect to a different extension if there is a match. I would also like to redirect calls that fail to present any CLI (aka WITHHELD/UNAVAILABE) to a different extension At I guess I would suspect I could implement this - can someone give me a starting point? Many thanks Brian -- _ -- 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] Do i need install Dahdi or libpri ?
On 25 Feb 2010, at 02:16, Zhang Shukun wrote: there is a AudioCodes Mediant 2000 out there. i want to realise ip to PSTN and PSTN to ip connection. Ok. after some configuration on AudioCodes Mediant 2000, PSTN to ip connecttion works. Thats good. a, Do i need install DAHDI or libpri in my system? Depends what else you are doing. I'd always just install it anyway. b, how to write in dialplan to realise connection to PSTN. The Mediant 2000 can be used like any other sip device. Dial(SIP/ whatever/1234567890) -- _ -- 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] curl and ssl certificate
On Thursday 25 February 2010 01:29:37 voipas wrote: Is it possible use asterisk curl function with ssl sertificate? If you're talking about just connecting to an SSL server, that is dependent upon using a version of libcurl with SSL support. If, on the other hand, you're talking about using a client certificate, that is not yet supported, but could be very easily by altering the CURLOPT function in versions which support it (1.6.2 and higher). Look in parse_curlopt_key() in func_curl.c, and it should be fairly obvious how to add custom support for it. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] X-Lite won't register
Beginner to Asterisk, but not beginner to VoIP FreePBX front end running on a dell 1550 and XLite running on a different Woindows XP box Both boxes connected via switch on same subnet. No NAT involved On FreePBX I created a new extension 1001 with a SIP password of 1001 On Xlite, username is 1001, password is 1001, authorization user name is 1001, and domain is IP of Free PBX XLite tries to register then shows 408 error registration timeout Windows box pings Asterisk and firewall is disabled on XP machine What am I missing? Jeff -- _ -- 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] X-Lite won't register
Check your Topology tab - the ICE setting tweaks connectivity if memory serves (I now have McAfee AV so can't use my Xlite to verify this :( ) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Girard, Jeffrey COL MIL USA Sent: Thursday, February 25, 2010 8:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] X-Lite won't register Beginner to Asterisk, but not beginner to VoIP FreePBX front end running on a dell 1550 and XLite running on a different Woindows XP box Both boxes connected via switch on same subnet. No NAT involved On FreePBX I created a new extension 1001 with a SIP password of 1001 On Xlite, username is 1001, password is 1001, authorization user name is 1001, and domain is IP of Free PBX XLite tries to register then shows 408 error registration timeout Windows box pings Asterisk and firewall is disabled on XP machine What am I missing? Jeff -- _ -- 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] Redirect call based on CLI???
On Thu, 2010-02-25 at 03:00 -0800, Kyle Kienapfel wrote: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf Has example exten = s,1,Answer exten = s/9184238080,2,Set(CALLERID(name)=EVIL BASTARD) exten = s,2,Set(CALLERID(name)=Good Person) exten = s,3,Dial(SIP/goodperson) for white list exten = s/123123123,1,Dial(SIP/phoneA) exten = s/456456456,1,Dial(SIP/phoneA) exten = s,1,Dial(SIP/phoneB) Thanks Kyle. I tried the example given but I could not get this to work - basically if I dial it from any phone that does not match 0800800800 (for illustration) it hangs up the channel with an error. exten = 845/0800800800,n,Set(CALLERID(name)=EVIL BASTARD) Auto fallthrough, channel 'SIP/1000-0017' status is 'UNKNOWN' I'm struggling to work out the logic here of a non-match, but this was not caught by i or s in error, so I'm probably missing some brain connection here. However, I've managed to do what I want using gotoif statements matching caller id - but I'd be interested to work out how the above is meant to branch on a non-match. -- _ -- 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] IAX peers one way voice
Hi all, i've 2 asterisk box with dahdi (server A ver. 1.4.29 and server B ver. 1.4.26) connected with IAX channel using gsm codec. - Calling from A to B the call has no problem: ring , answer a speak without problem. - Calling from B to A : B phone always listen ring also when A phone answer. After answer A phone don't listen anything. when A phone hangup the call disconnect. full logger in each server show normally ringing and hangup Any idea will be really appreciated thanks a lot in advance Lorenzo -- Chi vive sperando muore cagando ... Lo Russo isoletta dell'Egeo che non conta un cazzo, 1941 ... sono anche un autore -- _ -- 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] Redirect call based on CLI???
Do a core set verbose 10 and repeat the test. CLI should tell you what you need to handle the exception. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian Sent: Thursday, February 25, 2010 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Redirect call based on CLI??? On Thu, 2010-02-25 at 03:00 -0800, Kyle Kienapfel wrote: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions. conf Has example exten = s,1,Answer exten = s/9184238080,2,Set(CALLERID(name)=EVIL BASTARD) exten = s,2,Set(CALLERID(name)=Good Person) exten = s,3,Dial(SIP/goodperson) for white list exten = s/123123123,1,Dial(SIP/phoneA) exten = s/456456456,1,Dial(SIP/phoneA) exten = s,1,Dial(SIP/phoneB) Thanks Kyle. I tried the example given but I could not get this to work - basically if I dial it from any phone that does not match 0800800800 (for illustration) it hangs up the channel with an error. exten = 845/0800800800,n,Set(CALLERID(name)=EVIL BASTARD) Auto fallthrough, channel 'SIP/1000-0017' status is 'UNKNOWN' I'm struggling to work out the logic here of a non-match, but this was not caught by i or s in error, so I'm probably missing some brain connection here. However, I've managed to do what I want using gotoif statements matching caller id - but I'd be interested to work out how the above is meant to branch on a non-match. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems installing dahdi : kernel sources
Hello list, when installing Dahdi, the following error comes up : You do not appear to have the sources for the 2.6.18-164.11.1.el5xen kernel installed. make[1]: *** [modules] Error 1 The running kernel version : -bash-3.2# uname -a Linux vds.hosting.net 2.6.18-164.11.1.el5xen #1 SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux -bash-3.2# ls /usr/src/kernels/ 2.6.18-164.11.1.el5-x86_64 Isn't the kernel the same as the sources ?? Package kernel-devel-2.6.18-164.11.1.el5.x86_64 already installed and latest version Package kernel-headers-2.6.18-164.11.1.el5.x86_64 already installed and latest version -- _ -- 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] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences?
Hi to all asterisk-users ;) As some of you may know, Kernel 2.6.32 includes a module for the infamous Fritz passive ISDN cards in conjunction with mISDN. I just would like to know if anybody has tried to use a Fritz card as a BRI adapter for Asterisk with the new module. If so, i would be grateful for some hints or howtos. I am willing to try this myself, but i am quite reluctant as i have wasted weeks with Fritz cards over the last decade, most of the time without any success. Any help is appreciated. Regards Daniel Leese -- _ -- 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] Problems installing dahdi : kernel sources
On Thu, Feb 25, 2010 at 9:30 AM, jonas kellens jonas.kell...@telenet.bewrote: Isn't the kernel the same as the sources ?? Package kernel-devel-2.6.18-164.11.1.el5.x86_64 already installed and latest versionPackage kernel-headers-2.6.18-164.11.1.el5.x86_64 already installed and latest version You need the xen-specific kernel headers for your distro. If it's CentOS, you would do it with yum install kernel-xen-header. I'm not sure on Debian-based systems, but I think it's apt-get install kernel-xen-devel. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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] Deadlock while using MGCP on Asterisk
Hello all, I'm running Asterisk 1.2.35 with chan_mgcp activated. The process host around 2,4K users. Along the day I've got some debug reports like : Feb 24 22:25:42 DEBUG[28546] channel.c: Avoiding deadlock for 'MGCP/aaln/1...@028421223635-1' Feb 24 22:29:04 DEBUG[28670] channel.c: Avoiding initial deadlock for 'MGCP/aaln/1...@028421223635-1' Then, at random time (around 10~16 hours after a restart), Asterisk comes into deadlocks : Feb 25 16:28:22 WARNING[8149] channel.c: Avoided deadlock for '0xb713cb60', 9 retries! Feb 25 16:29:07 WARNING[8180] channel.c: Avoided initial deadlock for '0xb713cb60', 9 retries! Feb 25 16:40:21 WARNING[8629] channel.c: Avoided initial deadlock for '0xb713cb60', 9 retries! Avoided seems to correlate that Asterisk is in deadlock status. I put in attached a gdb output during the deadlock if it can helps. How can I correct these errors and avoid the crash not the deadlock J Regards, Adrien .L 0x00671149 in poll () from /lib/libc.so.6 (gdb) (gdb) info thread 21 Thread 0xb7fa4b90 (LWP 2881) 0x00671149 in poll () from /lib/libc.so.6 20 Thread 0xb7f64b90 (LWP 2882) 0x0072c5be in accept () from /lib/libpthread.so.0 19 Thread 0xb7f24b90 (LWP 2883) 0x00729660 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 18 Thread 0xb7ee4b90 (LWP 2885) 0x00673b27 in select () from /lib/libc.so.6 17 Thread 0xb7ea4b90 (LWP 2886) 0x0063a15c in nanosleep () from /lib/libc.so.6 16 Thread 0xb7e64b90 (LWP 2887) 0x00671149 in poll () from /lib/libc.so.6 15 Thread 0xb7e24b90 (LWP 2888) 0x0063a15c in nanosleep () from /lib/libc.so.6 14 Thread 0xb7de4b90 (LWP 2889) 0x0072c5be in accept () from /lib/libpthread.so.0 13 Thread 0xb7da4b90 (LWP 2890) 0x00671149 in poll () from /lib/libc.so.6 12 Thread 0xb7d64b90 (LWP 2891) 0x00673b27 in select () from /lib/libc.so.6 11 Thread 0xb7d24b90 (LWP 2892) 0x00671149 in poll () from /lib/libc.so.6 10 Thread 0xb7ce4b90 (LWP 2895) 0x0063a15c in nanosleep () from /lib/libc.so.6 9 Thread 0xb7c63b90 (LWP 2896) 0x00671149 in poll () from /lib/libc.so.6 8 Thread 0xb62ffb90 (LWP 2897) 0x00671149 in poll () from /lib/libc.so.6 7 Thread 0xb5cbfb90 (LWP 2133) 0x00671149 in poll () from /lib/libc.so.6 6 Thread 0xb627eb90 (LWP 5644) 0x00671149 in poll () from /lib/libc.so.6 5 Thread 0xb613eb90 (LWP 5669) 0x00671149 in poll () from /lib/libc.so.6 4 Thread 0xb617eb90 (LWP 6168) 0x00671149 in poll () from /lib/libc.so.6 3 Thread 0xb5a3fb90 (LWP 7066) 0x00671149 in poll () from /lib/libc.so.6 2 Thread 0xb5fbeb90 (LWP 7646) 0x0072bfd4 in __lll_lock_wait () from /lib/libpthread.so.0 * 1 Thread 0xb7fa5a70 (LWP 2879) 0x00671149 in poll () from /lib/libc.so.6 (gdb) 21 Thread 0xb7fa4b90 (LWP 2881) 0x00671149 in poll () from /lib/libc.so.6 20 Thread 0xb7f64b90 (LWP 2882) 0x0072c5be in accept () from /lib/libpthread.so.0 19 Thread 0xb7f24b90 (LWP 2883) 0x00729660 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 18 Thread 0xb7ee4b90 (LWP 2885) 0x00673b27 in select () from /lib/libc.so.6 17 Thread 0xb7ea4b90 (LWP 2886) 0x0063a15c in nanosleep () from /lib/libc.so.6 16 Thread 0xb7e64b90 (LWP 2887) 0x00671149 in poll () from /lib/libc.so.6 15 Thread 0xb7e24b90 (LWP 2888) 0x0063a15c in nanosleep () from /lib/libc.so.6 14 Thread 0xb7de4b90 (LWP 2889) 0x0072c5be in accept () from /lib/libpthread.so.0 13 Thread 0xb7da4b90 (LWP 2890) 0x00671149 in poll () from /lib/libc.so.6 12 Thread 0xb7d64b90 (LWP 2891) 0x00673b27 in select () from /lib/libc.so.6 11 Thread 0xb7d24b90 (LWP 2892) 0x00671149 in poll () from /lib/libc.so.6 10 Thread 0xb7ce4b90 (LWP 2895) 0x0063a15c in nanosleep () from /lib/libc.so.6 9 Thread 0xb7c63b90 (LWP 2896) 0x00671149 in poll () from /lib/libc.so.6 8 Thread 0xb62ffb90 (LWP 2897) 0x00671149 in poll () from /lib/libc.so.6 7 Thread 0xb5cbfb90 (LWP 2133) 0x00671149 in poll () from /lib/libc.so.6 6 Thread 0xb627eb90 (LWP 5644) 0x00671149 in poll () from /lib/libc.so.6 5 Thread 0xb613eb90 (LWP 5669) 0x00671149 in poll () from /lib/libc.so.6 4 Thread 0xb617eb90 (LWP 6168) 0x00671149 in poll () from /lib/libc.so.6 3 Thread 0xb5a3fb90 (LWP 7066) 0x00671149 in poll () from /lib/libc.so.6 2 Thread 0xb5fbeb90 (LWP 7646) 0x0072bfd4 in __lll_lock_wait () from /lib/libpthread.so.0 * 1 Thread 0xb7fa5a70 (LWP 2879) 0x00671149 in poll () from /lib/libc.so.6 (gdb) thread apply all bt Thread 21 (Thread 0xb7fa4b90 (LWP 2881)): #0 0x00671149 in poll () from /lib/libc.so.6 #1 0x080ba6cd in listener (unused=0x0) at asterisk.c:600 #2 0x0072544a in start_thread () from /lib/libpthread.so.0 #3 0xb7fa4470 in ?? () Thread 20 (Thread 0xb7f64b90 (LWP 2882)): #0 0x0072c5be in accept () from /lib/libpthread.so.0 #1 0x080b0c35 in accept_thread (ignore=0x0) at manager.c:1450 #2 0x0072544a in start_thread () from /lib/libpthread.so.0 #3 0xb7f64470 in ?? () Thread 19 (Thread 0xb7f24b90 (LWP 2883)): #0
Re: [asterisk-users] HFC-S card
On 23 February 2010 13:16, Razza razz...@gmail.com wrote: On 23 February 2010 12:58, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Have you managed to install those zaphfc drivers? Those are basically the same ones from http://code.google.com/p/zaphfc/ Hi Tzafrir. I checkout out that but there were no instructions. Thought I would shamelessly bump my request for simple instructions for patching DAHDi with the ZAPHFC driver on my CentOS 5.4 system. :o) -- _ -- 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] Which H.323 to use in Ast 1.6
Which Avaya system are you running? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Wednesday, February 24, 2010 5:52 PM To: 'Asterisk Users List' Subject: Re: [asterisk-users] Which H.323 to use in Ast 1.6 Could you share your config for the Asterisk and Avaya side too? Thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie A. Stapleton Sent: Wednesday, February 24, 2010 3:37 PM To: Asterisk Users List Subject: Re: [asterisk-users] Which H.323 to use in Ast 1.6 I have always used ooh323 between Avaya and Asterisk. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Tuesday, February 23, 2010 2:24 PM To: 'Asterisk Users List' Subject: [asterisk-users] Which H.323 to use in Ast 1.6 We're doing a project that requires H.323 to an Avaya. Does anyone have experience to share on which H.323 driver to use in asterisk 1.6? Is the diference between h323 and ooh323 still worth the extra effort? (We've only installed h323 under 1.4) If you have setup/config experience with this setup in Asterisk 1.6 please share! Thanks, MD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe() and dahdi_dummy on an embedded system
On 02/25/2010 11:19 AM, Vinícius Fontes wrote: I'm playing around with an ALIX 2D2 board (http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an AMD Geode processor with 256MB of RAM. Also available are two network interfaces, two USB ports and one serial port (no keyboard or VGA). I'm using the Voyage Linux distro, which basically is Debian Lenny optimized for this board. Asterisk 1.6.1.12 runs fine on the system. The only issue I'm having is with MeetMe(). As there's no DAHDI devices attached, I'm running dahdi_dummy. Audio gets all choppy on MeetMe(), but works fine for other applications such as Playback(). SIP calls also work fine. Most probably it's a timing issue. I connected an Astribank unit with 16 FXS in order to provide timing, and after that I get crystal clear audio on MeetMe(). Of course I wouldn't like to have an expensive Astribank attached to the ALIX board just to provide timing. So my question is: is there any way to improve dahdi_dummy's performance, or maybe some other way to get this to work without the need for a physical DAHDI device? What version of DAHDI are you using? As long as the host kernel is able to accurately keep accurate wall time, I'm not aware of any conditions that would prevent dahdi_dummy in dahdi-linux 2.2.1 from working fine, so I'm very curious if this isn't the case. In fact, in the trunk of dahdi-linux dahdi_dummy.ko is off by default and dahdi.ko will be able to keep time regardless of whether there are any physical spans connected or configured. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] MeetMe() and dahdi_dummy on an embedded system
- Shaun Ruffell sruff...@digium.com escreveu: On 02/25/2010 11:19 AM, Vinícius Fontes wrote: I'm playing around with an ALIX 2D2 board (http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an AMD Geode processor with 256MB of RAM. Also available are two network interfaces, two USB ports and one serial port (no keyboard or VGA). I'm using the Voyage Linux distro, which basically is Debian Lenny optimized for this board. Asterisk 1.6.1.12 runs fine on the system. The only issue I'm having is with MeetMe(). As there's no DAHDI devices attached, I'm running dahdi_dummy. Audio gets all choppy on MeetMe(), but works fine for other applications such as Playback(). SIP calls also work fine. Most probably it's a timing issue. I connected an Astribank unit with 16 FXS in order to provide timing, and after that I get crystal clear audio on MeetMe(). Of course I wouldn't like to have an expensive Astribank attached to the ALIX board just to provide timing. So my question is: is there any way to improve dahdi_dummy's performance, or maybe some other way to get this to work without the need for a physical DAHDI device? What version of DAHDI are you using? As long as the host kernel is able to accurately keep accurate wall time, I'm not aware of any conditions that would prevent dahdi_dummy in dahdi-linux 2.2.1 from working fine, so I'm very curious if this isn't the case. In fact, in the trunk of dahdi-linux dahdi_dummy.ko is off by default and dahdi.ko will be able to keep time regardless of whether there are any physical spans connected or configured. Sorry, forgot to include the DAHDI version. voyage:~# dahdi_cfg -tv DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): Configuration == I'm almost sure this board's RTC is not very accurate. Is there any way to measure the RTC's accuracy? -- _ -- 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] Followme broken
Hi, we are running Asterisk 1.6.1.14 and have a issue that when we use followme the call is correctly placed to the mobile phone, the mobile rings, but when answered we do not hear the normal followme introduction message. If we press 1 to accept there is just silence. Has anybody else seen this issue before ? -- Thanks, Phil -- _ -- 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] audio glitches in conference
marco.mo...@gmail.com wrote: It looks to me that u are having clock synchronism problems due to the fact you are using Virtual Machine so u don't have an ISDN card generating clock. Are u using what was called ztdummie as clock source? Can't precise the name of it in chan_dahdi but u have it. zt_dummy is the timing source, yes. I saw the exact same thing happening when I used app_conference though, without zt_dummy loaded at all. I also checked if I could see what source zt_dummy is using, but there's no 'source' line in /proc/zaptel/1 - just one line saying 'Span 1: ZTDUMMY/1 ZTDUMMY/1 1 ' -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] audio glitches in conference
Jeff Brower wrote: How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? I just loaded the file into audacity and measured the gaps by looking at the wave form. I just went through some of the samples I recorded yesterday, and found that all the gaps are multiples of 8 samples - from 8 up to 32. I guess that's because dahdi/zaptel ticks are 8 samples each, so if it misses one, 8 samples get lost. I can't imagine that RTP is involved, since this is happening with purely local channels (just the Playback application and the eagi script) -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] MeetMe() and dahdi_dummy on an embedded system
On Thu, 25 Feb 2010, Vinícius Fontes wrote: - Shaun Ruffell sruff...@digium.com escreveu: On 02/25/2010 11:19 AM, Vinícius Fontes wrote: I'm playing around with an ALIX 2D2 board (http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an AMD Geode processor with 256MB of RAM. Also available are two network interfaces, two USB ports and one serial port (no keyboard or VGA). I'm using the Voyage Linux distro, which basically is Debian Lenny optimized for this board. Asterisk 1.6.1.12 runs fine on the system. The only issue I'm having is with MeetMe(). As there's no DAHDI devices attached, I'm running dahdi_dummy. Audio gets all choppy on MeetMe(), but works fine for other applications such as Playback(). SIP calls also work fine. Most probably it's a timing issue. I connected an Astribank unit with 16 FXS in order to provide timing, and after that I get crystal clear audio on MeetMe(). Of course I wouldn't like to have an expensive Astribank attached to the ALIX board just to provide timing. So my question is: is there any way to improve dahdi_dummy's performance, or maybe some other way to get this to work without the need for a physical DAHDI device? What version of DAHDI are you using? As long as the host kernel is able to accurately keep accurate wall time, I'm not aware of any conditions that would prevent dahdi_dummy in dahdi-linux 2.2.1 from working fine, so I'm very curious if this isn't the case. In fact, in the trunk of dahdi-linux dahdi_dummy.ko is off by default and dahdi.ko will be able to keep time regardless of whether there are any physical spans connected or configured. Sorry, forgot to include the DAHDI version. voyage:~# dahdi_cfg -tv DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): Configuration == I'm almost sure this board's RTC is not very accurate. Is there any way to measure the RTC's accuracy? I use these boards too. Brilliant little things. 5 watts and 80+ concurrent calls handling media before they fall over! Not much good for transcoding though... http://unicorn.drogon.net/cutie.jpg However I have a slightly different approach in that I have my own semi-custom Linux for them which runs entirely in RAM. I also custom-compile the kernel for the architecture, and have compiled up dhadi and asterisk specifically for that CPU too. (Read my earlier whinges about it some time back!) dsx$ df -h FilesystemSize Used Avail Use% Mounted on /dev/ram0 136M 79M 58M 58% / /dev/hda3 189M 95M 95M 50% /data dahdi_dummy on my systems use the high resolution timer and not RTC: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.0-rc2 dahdi_dummy: Trying to load High Resolution Timer dahdi_dummy: Initialized High Resolution Timer dahdi_dummy: Starting High Resolution Timer dahdi_dummy: High Resolution Timer started, good to go Other than a few quick tests, I've not really run many meetmes on one, but I have a few dozen of these out in the world with clients so I don't really know what they're doing with them... However they all use IAX trunking which I understand requires timing too. There's a few tweaks you can do to the system even running the distro you're using - make sure no extra services are running - make sure logging is minimal and not using fsync on every write (see your syslog config file), mount partitions with the noatime and nodiratime flags - use as fast a CF card as you can get, and so on. (Google for tuning hints for systems like the Acer Aspire One and other similar laptops with SSDs and so on) Output of ps ax: PID TTY STAT TIME COMMAND 1 ?Ss 0:01 init [2] 2 ?S 0:00 [kthreadd] 3 ?S 0:00 [ksoftirqd/0] 4 ?S 0:00 [events/0] 5 ?S 0:00 [khelper] 60 ?S 0:00 [kblockd/0] 67 ?S 0:00 [khubd] 106 ?S 0:00 [pdflush] 108 ?S 0:00 [kswapd0] 153 ?S 0:00 [aio/0] 737 ?S 0:00 [pdflush] 1006 ?Ss 0:00 /sbin/syslogd 1010 ?Ss 0:00 /sbin/klogd -x 1021 ?Ss 0:00 /usr/sbin/sshd 1028 ?Ss 0:00 /usr/local/apache2/bin/httpd -k start 1042 ?Ss 0:00 /usr/sbin/ntpd -p /var/run/ntpd.pid -g 1047 ?Ssl0:00 /usr/sbin/asterisk -p 1086 ?Ss 0:00 /usr/sbin/cron 1099 ttyS0Ss 0:00 /bin/login -- 1323 ?S 0:00 /usr/local/apache2/bin/httpd -k start 2089 ttyS0S+ 0:00 -bash 2094 ?Ss 0:00 sshd: d...@pts/0 2098 pts/0Ss 0:00 login -h yakko.drogon.net -p -f 2099 pts/0R 0:00 -bash 2109 pts/0R+ 0:00 ps ax Not sure I can get it any more minimal than that! Gordon-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
Re: [asterisk-users] Deadlock while using MGCP on Asterisk
Thank you guys for your feedback. I consider the upgrading to 1.4.29.1. Does it can definitively prevent me from this kind of freeze ? Regards, Adrien .L De : Miguel Molina [mailto:mmol...@millenium.com.co] Envoyé : jeudi 25 février 2010 18:21 À : alemo...@legos.fr; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Deadlock while using MGCP on Asterisk Adrien Lemoine escribió: Hello all, Im running Asterisk 1.2.35 with chan_mgcp activated. The process host around 2,4K users. Along the day Ive got some debug reports like : Feb 24 22:25:42 DEBUG[28546] channel.c: Avoiding deadlock for 'MGCP/aaln/1...@028421223635-1' Feb 24 22:29:04 DEBUG[28670] channel.c: Avoiding initial deadlock for 'MGCP/aaln/1...@028421223635-1' Then, at random time (around 10~16 hours after a restart), Asterisk comes into deadlocks : Feb 25 16:28:22 WARNING[8149] channel.c: Avoided deadlock for '0xb713cb60', 9 retries! Feb 25 16:29:07 WARNING[8180] channel.c: Avoided initial deadlock for '0xb713cb60', 9 retries! Feb 25 16:40:21 WARNING[8629] channel.c: Avoided initial deadlock for '0xb713cb60', 9 retries! Avoided seems to correlate that Asterisk is in deadlock status. I put in attached a gdb output during the deadlock if it can helps. How can I correct these errors and avoid the crash not the deadlock J Regards, Adrien .L That kind of Avoided deadlock... messages, typical for early 1.2 systems have gone on recent versions on 1.4.X and higher. Did you consider upgrading? Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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] Asterisk Crashs due to some Sip messages
Hello Asterisk community, Today my asterisk server stop working and i had to reboot the server in order to make it work again, take a look at the error messages in the CLI at the time of the crash: [Feb 25 12:44:20] WARNING[6965] chan_sip.c: sip_xmit of 0x920ae80 (len 545) to 10.4.2.3:5060 returned -2: Network is unreachable [Feb 25 12:44:20] WARNING[6965] chan_sip.c: Network error on retransmit in dialog 4bf327cc199699d00ddb74911742a...@10.4.1.6 [Feb 25 12:44:20] WARNING[6965] chan_sip.c: Transmit error :: Cancelling transmission of transaction in call id 4bf327cc199699d00ddb74911742a...@10.4.1.6 Can you please help me with this? what could be the problem here? Thanks in advance -- _ -- 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] Morse Code
Hi Chris, Morse code is mainly used for HAM radio activity with Asterisk, connecting radio repeaters area through Internet. It's often mandatory (local regulations side) to transmit from time to time and/or at end of traffic period a Morse sequence including the repeater (or repeater's owner) HAM radio call sign and position (and some extra info as time, temperature and so...). You can read more about Asterisk and HAM radio here : http://www.zapatatelephony.org/ and here : http://app-rpt.qrvc.com/ Best 73's from F6HQZ Francois (France) Le 25/02/2010 19:45, Chris Kairalla a écrit : This is just curiosity, but I'm wondering why the Morsecode app has remained part of the trunk for all of these years. Is there any practical use for this or is it just an homage to the ghosts of telecommunications past? Does anybody use the Morsecode app for anything interesting? I'm strangely fascinated by this core piece of Asterisk functionality. -Chris -- _ -- 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] Morse Code
This is just curiosity, but I'm wondering why the Morsecode app has remained part of the trunk for all of these years. Is there any practical use for this or is it just an homage to the ghosts of telecommunications past? Does anybody use the Morsecode app for anything interesting? I'm strangely fascinated by this core piece of Asterisk functionality. -Chris -- _ -- 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] Morse Code
snip Does anybody use the Morsecode app for anything interesting? I'm strangely fascinated by this core piece of Asterisk functionality. /snip Duh! How are we going to spread the word about how to take those alien bastards down if we don't keep morse code around!?!??! http://www.imdb.com/title/tt0116629/ [quote] 02:17:03 We know how to take 'em out, General. Spread the word. 02:17:10 Get on the wire to every squadron around the world. 02:17:14 Tell them how to bring those sons of bitches down. [/quote] :) -Dave -- _ -- 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] MeetMe() and dahdi_dummy on an embedded system
- Gordon Henderson gordon+aster...@drogon.net escreveu: On Thu, 25 Feb 2010, Vinícius Fontes wrote: - Shaun Ruffell sruff...@digium.com escreveu: On 02/25/2010 11:19 AM, Vinícius Fontes wrote: I'm playing around with an ALIX 2D2 board (http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an AMD Geode processor with 256MB of RAM. Also available are two network interfaces, two USB ports and one serial port (no keyboard or VGA). I'm using the Voyage Linux distro, which basically is Debian Lenny optimized for this board. Asterisk 1.6.1.12 runs fine on the system. The only issue I'm having is with MeetMe(). As there's no DAHDI devices attached, I'm running dahdi_dummy. Audio gets all choppy on MeetMe(), but works fine for other applications such as Playback(). SIP calls also work fine. Most probably it's a timing issue. I connected an Astribank unit with 16 FXS in order to provide timing, and after that I get crystal clear audio on MeetMe(). Of course I wouldn't like to have an expensive Astribank attached to the ALIX board just to provide timing. So my question is: is there any way to improve dahdi_dummy's performance, or maybe some other way to get this to work without the need for a physical DAHDI device? What version of DAHDI are you using? As long as the host kernel is able to accurately keep accurate wall time, I'm not aware of any conditions that would prevent dahdi_dummy in dahdi-linux 2.2.1 from working fine, so I'm very curious if this isn't the case. In fact, in the trunk of dahdi-linux dahdi_dummy.ko is off by default and dahdi.ko will be able to keep time regardless of whether there are any physical spans connected or configured. Sorry, forgot to include the DAHDI version. voyage:~# dahdi_cfg -tv DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): Configuration == I'm almost sure this board's RTC is not very accurate. Is there any way to measure the RTC's accuracy? I use these boards too. Brilliant little things. 5 watts and 80+ concurrent calls handling media before they fall over! Not much good for transcoding though... http://unicorn.drogon.net/cutie.jpg However I have a slightly different approach in that I have my own semi-custom Linux for them which runs entirely in RAM. I also custom-compile the kernel for the architecture, and have compiled up dhadi and asterisk specifically for that CPU too. (Read my earlier whinges about it some time back!) dsx$ df -h FilesystemSize Used Avail Use% Mounted on /dev/ram0 136M 79M 58M 58% / /dev/hda3 189M 95M 95M 50% /data dahdi_dummy on my systems use the high resolution timer and not RTC: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.0-rc2 dahdi_dummy: Trying to load High Resolution Timer dahdi_dummy: Initialized High Resolution Timer dahdi_dummy: Starting High Resolution Timer dahdi_dummy: High Resolution Timer started, good to go Other than a few quick tests, I've not really run many meetmes on one, but I have a few dozen of these out in the world with clients so I don't really know what they're doing with them... However they all use IAX trunking which I understand requires timing too. There's a few tweaks you can do to the system even running the distro you're using - make sure no extra services are running - make sure logging is minimal and not using fsync on every write (see your syslog config file), mount partitions with the noatime and nodiratime flags - use as fast a CF card as you can get, and so on. (Google for tuning hints for systems like the Acer Aspire One and other similar laptops with SSDs and so on) Output of ps ax: PID TTY STAT TIME COMMAND 1 ?Ss 0:01 init [2] 2 ?S 0:00 [kthreadd] 3 ?S 0:00 [ksoftirqd/0] 4 ?S 0:00 [events/0] 5 ?S 0:00 [khelper] 60 ?S 0:00 [kblockd/0] 67 ?S 0:00 [khubd] 106 ?S 0:00 [pdflush] 108 ?S 0:00 [kswapd0] 153 ?S 0:00 [aio/0] 737 ?S 0:00 [pdflush] 1006 ?Ss 0:00 /sbin/syslogd 1010 ?Ss 0:00 /sbin/klogd -x 1021 ?Ss 0:00 /usr/sbin/sshd 1028 ?Ss 0:00 /usr/local/apache2/bin/httpd -k start 1042 ?Ss 0:00 /usr/sbin/ntpd -p /var/run/ntpd.pid -g 1047 ?Ssl0:00 /usr/sbin/asterisk -p 1086 ?Ss 0:00 /usr/sbin/cron 1099 ttyS0Ss 0:00 /bin/login -- 1323 ?S 0:00 /usr/local/apache2/bin/httpd -k start 2089 ttyS0S+ 0:00 -bash 2094 ?Ss 0:00 sshd: d...@pts/0 2098 pts/0Ss 0:00 login -h yakko.drogon.net -p -f 2099 pts/0R
Re: [asterisk-users] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences?
Hi! As some of you may know, Kernel 2.6.32 includes a module for the infamous Fritz passive ISDN cards in conjunction with mISDN. Haven't tried/seen that, but mISDN ... I avoid it if I can. I am willing to try this myself, but i am quite reluctant as i have wasted weeks with Fritz cards over the last decade, most of the time without any success. Any help is appreciated. I have excellent success with the tiny fcpci and chan_capi, which is also working great with capi4hylafax. See net-dialup/fcpci-0.1-r1 in gentoo (should not be difficult to use this on other distros, but I have never done so). Do not confuse this with the fritzcapi! See: http://opensuse.foehr-it.de/ (initially for ... SUSE :-) Maintainer appears to be: Stefan Briesenick sbrie...@gentoo.org http://www.chan-capi.org/ (Armin Schindler is occasionally active on this list here) HTH, Philipp P.S.: I am moving away from PCI cards towards external gateways, because those are nothing to worry about when you put a new kernel into place. I am tired of that dance. -- _ -- 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] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid
Hi,
I have two asterisk servers with the same version of 1.4.29.1.
The first server named it as MYE1. MYE1 is an incoming server that can
accept incoming calls from PSTN(ZAP E1).
The second server is a pbx functions server and named it as MYPBX(SIP).
The sip.conf of MYE1 likes below:
[MYPBX]
type=peer
host=mypbx.abc.com
nat=no
disallow=all
allow=g729
canreinvite=yes
qualify=no
context=default
insecure=port,invite
The sip.conf of MYPBX likes below:
[MYE1]
type=peer
host=mye1.abc.com
nat=no
disallow=all
allow=g729
canreinvite=yes
qualify=no
context=did
insecure=port,invite
The call flow is
1. Mobile with disable callerid(+886-912-345678) make a call to DIDs on the
E1 (for example: +886-922-66 and enters MYE1 system. But my telecomm
provider helps me to solve the callerid and make it enable. So that, I can
find callerid of Mobile from MYE1.
2. MYE1 accept this call and dial it to MYPBX. In this moment, I can find
the fllowing message on the CLI of MYE1.
In Another word, the Caller ID is correct here.
-- Accepting call from '912345678' to '092266' on channel 0/22, span
4
-- Executing [0922666...@default:1] Set(DAHDI/94-1,
CDR(userfield)=0922E1) in new stack
-- Executing [0922666...@default:2] Set(DAHDI/94-1,
CALLERID(num)=912345678) in new stack
-- Executing [0922666...@default:3] Set(DAHDI/94-1,
CALLERID(num)=912345678) in new stack
-- Executing [0922666...@default:4] NoOp(DAHDI/94-1, CID num:
[986230883]) in new stack
-- Executing [0922666...@default:5] Dial(DAHDI/94-1, SIP/
mypbx.abc.com/092266) in new stack
-- Called mypbx.abc.com/092266
-- SIP/mypbx.abc.com-2551 is ringing
extensions.conf
exten = 092266,1,Set(CDR(userfield)=0922E1)
exten = 092266,n,NoOp(CID num: [${CALLERID(num)}])
exten = 092266,n,Set(CALLERID(num)=${CALLERID(num)})
exten = 092266,n,NoOp(CID num: [${CALLERID(num)}])
exten = 092266,n,Dial(SIP/mypbx.abc.com/${EXTEN})
exten = 092266,n,Hangup
3. But the strange thing is MYPBX. I use the function NoOp to find the
callerid that call from MYE1.
-- Executing [0922666...@did:1] NoOp(SIP/MYE1-0185, CID Num:
Anonymous) in new stack
-- Executing [0922666...@did:2] Hangup
extensions.conf
exten = _X.,1,NoOp(CID Num: ${CALLERID(number)})
exten = _X.,1,Hangup
4. I got the ngrep message from MYPBX.
U 210.200.XXX.XX:5060 - 61.65.XX.XX:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
61.65.XX.XX:5060;branch=z9hG4bK276d72eb;received=61.65.XX.XX;rport=5060.
From: Anonymous sip:anonym...@anonymous.invalid;tag=as2b63fbb6.
To: sip:0922666...@mypbx.abc.com sip%3a0922666...@mypbx.abc.com.
Call-ID: 1a9bc32e05eeb8b27bfadfba35c09...@61.65.xx.xx.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO.
Supported: replaces.
Contact: sip:0922666...@210.200.xxx.xx.
Content-Length: 0.
.
U 210.200.XXX.XX:5060 - 61.65.XX.XX:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP
61.65.XX.XX:5060;branch=z9hG4bK276d72eb;received=61.65.XX.XX;rport=5060.
From: Anonymous sip:anonym...@anonymous.invalid;tag=as2b63fbb6.
To: sip:0922666...@xm1.gvlink.net sip%3a0922666...@xm1.gvlink.net
;tag=as66351139.
Call-ID: 1a9bc32e05eeb8b27bfadfba35c09...@61.65.xx.xx.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO.
Supported: replaces.
5. My questions are:
A. Why can't I receive the CALLERID from MYPBX(the secondary server)? I
am sure I use Set(CALLERID(num) for it.
B. Why does the CALLERID that sends from MYE1 become as Anonymous? How
can I fix it with the correct orginal callerid(912345678)?
C. Why does my FROM message become as Anonymous
sip:anonym...@anonymous.invalid instead of 912345...@mye1.abc.com ?
If you have any suggestions, please let me know. Thank you very much.
--
Best Regards
Charles
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Re: [asterisk-users] audio glitches in conference
Jonathan- How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? I just loaded the file into audacity and measured the gaps by looking at the wave form. I just went through some of the samples I recorded yesterday, and found that all the gaps are multiples of 8 samples - from 8 up to 32. I guess that's because dahdi/zaptel ticks are 8 samples each, so if it misses one, 8 samples get lost. I can't imagine that RTP is involved, since this is happening with purely local channels (just the Playback application and the eagi script) Did you measure the distance between gaps? If those distances are multiples of RTP payload length, then possibly a network latency issue is involved, but otherwise I agree with other posters, it sounds more like a timing and/or sampling synchronization problem. Are you handling TDM data, for example T1/E1 or ISDN? If so, then normally the TDM card inputs the external T1 line clock and then provides it as a master for software handling TDM data. If somehow the TDM clock is not configured this way, or the software (Asterisk) is using another clock, then effectively you have two (2) different TDM clocks and they are drifting with respect to each other. In this way you could miss bursts of samples (or get incorrect samples) as the clocks drift to be 180 out. Then you'd be Ok again for a while until it happens again, and so on... -Jeff -- _ -- 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] MeetMe() and dahdi_dummy on an embedded system
On Thu, 25 Feb 2010, Vinícius Fontes wrote: Just checked and I'm using the high res timer as well: Feb 25 17:42:32 voyage vmunix: [ 27.028798] dahdi_dummy: Trying to load High Resolution Timer Feb 25 17:42:32 voyage vmunix: [ 27.028816] dahdi_dummy: Initialized High Resolution Timer Feb 25 17:42:32 voyage vmunix: [ 27.028831] dahdi_dummy: Starting High Resolution Timer Feb 25 17:42:32 voyage vmunix: [ 27.028849] dahdi_dummy: High Resolution Timer started, good to go Feb 25 17:42:32 voyage vmunix: [ 27.055253] dahdi: Registered tone zone 20 (Brazil) Ok. Looks good. I also compiled DAHDI and Asterisk from the sources. Took about 2 hours but it finally compiled and is running okay. :) Still not sure what's happening, since even with 2 users on the meetme room I still get the choppy audio. My best guess would be something kernel related. Thinking about recompiling the kernel, but I'm not sure what I could set to maybe solve these issues. That's why I cross compile - it just takes too long! Would you mind sharing the kernel version number you're running on your boxes, and if I'm not asking too much, the .config file you used? Thanks a lot in advance. http://unicorn.drogon.net/configs/config.2.6.30.1.geode Drop that into .config in a stock 2.6.30.1 kernel off www.kernel.org and off you go. That will produce a kernel with no modules in it. You'll need to re-make dahdi. Good luck! Gordon-- _ -- 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] AST-2010-003: Invalid parsing of ACL rules can compromise security
Asterisk Project Security Advisory - AST-2010-003 ++ | Product | Asterisk | |+---| | Summary | Invalid parsing of ACL rules can compromise | || security | |+---| | Nature of Advisory | Unauthorized access to system | |+---| | Susceptibility | Remote Unauthenticated Sessions | |+---| | Severity | Moderate | |+---| | Exploits Known | No| |+---| |Reported On | Feb 24, 2010 | |+---| |Reported By | Mark Michelson| |+---| | Posted On | Feb 25, 2010 | |+---| | Last Updated On | February 25, 2010 | |+---| | Advisory Contact | Mark Michelson mmichelson AT digium DOT com| |+---| | CVE Name | | ++ ++ | Description | Host access rules using permit= and deny=| | | configurations behave unpredictably if the CIDR notation | | | /0 is used. Depending on the system's behavior, this | | | may act as desired, but in other cases it might not, | | | thereby allowing access from hosts that should be| | | denied. | | | | | | Note that even if an unauthorized host is allowed access | | | due to this exploit, authentication measures still in| | | place would prevent further unauthorized access. | | | | | | Note also that there is a workaround for this problem, | | | which is to use the dotted-decimal format /0.0.0.0 | | | instead of CIDR notation. The bug does not exist when| | | using this format. In addition, this format is what is | | | used in Asterisk's sample configuration files. | ++ ++ | Resolution | Code has been corrected to behave consistently on all | || systems when /0 is used.| ++ ++ | Affected Versions| || | Product | Release | | || Series | | |+-+-| |Asterisk Open Source| 1.2.x | Unaffected | |+-+-| |Asterisk Open Source| 1.4.x | Unaffected | |+-+-| |Asterisk Open Source| 1.6.x | All 1.6.0, 1.6.1 and 1.6.2 | || | releases| |+-+-| | Asterisk Addons | 1.2.x | Unaffected | |+-+-| | Asterisk Addons
[asterisk-users] DTMF timing - first # keypress not registering
Hi Everyone, I set up my Asterisk 1.4.24 system and everything works well except when I dial into another service (like conference calling with GoToMeeting) where I must enter my pin followed by a pound sign. When I do this, it does not register - BUT if I press the pound sign a second time (after waiting 1 sec after the first time I pressed #) it seems to work. I was able to resolve this on my Grandstream GPX2000 by changing the DTMF Send method to in audio (the other two options: 1. via RTP(rfc2833) 2. via SIP INFO did not work.) My Aastra 9480i phones do not have this option. It has 1. RTP, 2. SIP INFO, 3. BOTH - which I tried all three to no avail. I am currently set at SIP INFO. My Xlite softphones are also not passing the first keypress of # - but if I wait a second or two and press # again, it works. Thank you very much for any pointers on resolving this. -- _ -- 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] Asterisk 1.6.0.25, 1.6.1.17, and 1.6.2.5 Now Available
The Asterisk Development Team has announced security releases for the following versions of Asterisk: * 1.6.0.25 * 1.6.1.17 * 1.6.2.5 These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The releases of Asterisk 1.6.0.25, 1.6.1.17, and 1.6.2.5 resolve an issue with invalid parsing of ACL (Access Control List) rules leading to a possible compromise in security. The issue and resolution are described in the AST-2010-003 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2010-003, which was released at the same time as this announcement. It should also be noted that release candidates for the 1.6.x series of Asterisk have been skipped (1.6.0.23-rc2, 1.6.1.15-rc2, and 1.6.2.3-rc2). New release candidates will be released as 1.6.0.26-rc1, 1.6.1.18-rc1, and 1.6.2.6-rc1 pending another security release. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.25 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.17 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.5 Security advisory AST-2010-003 is available at: http://downloads.asterisk.org/pub/security/AST-2010-003.pdf Thank you for your continued support of Asterisk! -- _ -- 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] Asterisk 1.6.0.17 PBX with two interfaces does not routes RTP packets - SIP Conf Problem likely
Hi, I am try to configure Asterisk as PBX system with two interfaces as shown below. One interface pointing to the local subnet with a SIP phone and another interface pointing to the external ISP SIP Sever. SJPhone(X.X.141.32)-(Y.Y.47.149)local-intf-|Asterisk|external- intf(Z.Z.247.106)(w.w.158.26)ISP-SIP-ServerOutsideWorld I am able to setup a call from the Phone to the outside world and I have the audio (RTP packets) coming from the outside world being routed to my phone but the audio from my Phone IP(X.X) is not going out to the SIP-Server. In fact I think it is not even reaching the Asterisk server because the SDP in the 183 going to the phone has the IP address of the external-inf(Z.Z.247.106) of the Asterisk PBX when it should actually (Y.Y.47.149) --- Transmitting (NAT) to X.X.141.32:5060 --- SIP/2.0 183 Session Progress^M Via: SIP/2.0/UDP X.X.141.32;branch=z9hG4bK87468d2002f44b86a0046f2b0166;receiv ed=X.X.141.32;rport=5060^M From: Irfan Lateef sip:2...@y.y.47.149;tag=327f290e2e7^M To: sip:99084611...@y.y.47.149;tag=as24228e21^M Call-ID: 876BAA6B36F644F7B4EF7BE5D4B7E8BD0x87468d20^M CSeq: 2 INVITE^M User-Agent: Asterisk PBX 1.6.0.17^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M Supported: replaces, timer^M Require: timer^M Session-Expires: -1;refresher=uas^M Contact: sip:99084611...@z.z.247.106^M Content-Type: application/sdp^M Content-Length: 315^M ^M v=0^M o=root 1021147583 1021147583 IN IP4 Z.Z.247.106^M s=Asterisk PBX 1.6.0.17^M c=IN IP4 Z.Z.247.106^M t=0 0^M m=audio 18702 RTP/AVP 0 8 3 101^M I have the following in the sip_nat.conf localnet=Y.Y.47.149/255.255.0.0 externhost=Z.Z.247.106 externrefresh=10 fromdomain=att.com nat=yes qualify=yes canreinvite=no I think the SDP should have give the Y.Y.47.149 IP on the local net side to the phone. But I am unable to figure how make it do that. The Asterisk log shows this. [Feb 25 11:06:30] VERBOSE[1449] logger.c: -- Executing [...@macro-dialout-trunk:19] ^[[1;36;40mDial^[[0;37;40m(^[[1;35;40mSIP/2005-19dc0db8^[[0;37;40m, ^[[1;35;40mSIP/ATT-alpi016-IPFlex1/19084611234,300,^[[0;37;40m) in new stack [Feb 25 11:06:30] VERBOSE[1449] logger.c: == Using SIP RTP TOS bits 184 [Feb 25 11:06:30] VERBOSE[1449] logger.c: == Using SIP RTP CoS mark 5 [Feb 25 11:06:30] VERBOSE[1449] logger.c: -- Called ATT-alpi016-IPFlex1/19084611234 [Feb 25 11:06:32] VERBOSE[1449] logger.c: -- SIP/ATT-alpi016-IPFlex1-19dda0f8 is making progress passing it to SIP/2005-19dc0db8 [Feb 25 11:06:32] VERBOSE[1449] logger.c: Audio is at Z.Z.247.106 port 18702 [Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding codec 0x4 (ulaw) to SDP [Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding codec 0x8 (alaw) to SDP [Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding codec 0x2 (gsm) to SDP [Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 25 11:06:32] VERBOSE[1449] logger.c: Any help is greatly appreciated. Thanks and Regards, Irfan Lateef -- _ -- 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] How can we pickup a call that is not going to a real extension?
Hello,
We have a situation where a call comes in, users are notified via an external
process (curl request to web service), and we can't answer the call until a
callee can call in and pickup the call.
How can we implement this functionality?
We tried using :
[caller-inbound-leg]
; code to send the CALL_UUID information to users.
exten =
_[+0-9a-zA-Z*#_].,n(r203),Dial(LOCAL/${call_uu...@inbound-wait-loop,,r)
; wait for call pickup from callee's inbound leg
[inbound-wait-loop]
exten = _[+0-9a-zA-Z*#_].,1,Wait(30)
[callee-inbound-leg]
; code to figure out the CALL_UUID used for the callee-leg
exten = _[+0-9a-zA-Z*#_].,n,Pickup(${call_uu...@inbound-wait-loop)
We thought Pickup() would work, but it only seems to work if the call is in the
Dial state.
The logs have results like:
-- Executing [5552...@caller-inbound-leg:8]
Pickup(SIP/20678350-5cd1-11de-bcf8-123139006632-004e,
14f0dff4-9a34-11dd-93fd-0015588ab...@inbound-wait-loop) in new stack
[2010-02-25 19:20:28.936] NOTICE[18759]: app_directed_pickup.c:294 pickup_exec:
No target channel found for 14f0dff4-9a34-11dd-93fd-0015588ab9f3.
Is there a way for a callee to pickup a call in the Wait state?
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Eric Chamberlain
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[asterisk-users] How to tell if asterisk is handling media or not?
I'm trying to get my asterisk server to reinvite. I have two asterisk servers with public IP's. My users (behind NAT) register on one server (I'll call it server 1), and for some calls they are transfered over to the other server (server 2), because that server has the E1's. I want server 1 to be in the signaling path for billing reasons, but handling the media stream is killing my capacity, and it should not be necessary as server 2 also has a public IP address. I have tried playing around with the canreinvite options in sip.conf but the problem is I cannot tell if asterisk is reinviting the call or not. How can I figure out where the media stream is going? thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to tell if asterisk is handling media or not?
In 1.2 you can use rtp debug in the CLI On Thu, Feb 25, 2010 at 8:27 PM, Alejandro Recarey alexreca...@gmail.com wrote: I'm trying to get my asterisk server to reinvite. I have two asterisk servers with public IP's. My users (behind NAT) register on one server (I'll call it server 1), and for some calls they are transfered over to the other server (server 2), because that server has the E1's. I want server 1 to be in the signaling path for billing reasons, but handling the media stream is killing my capacity, and it should not be necessary as server 2 also has a public IP address. I have tried playing around with the canreinvite options in sip.conf but the problem is I cannot tell if asterisk is reinviting the call or not. How can I figure out where the media stream is going? thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid
Charles Wang wrote: The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com http://mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default insecure=port,invite Add sendrpid=yes here. The sip.conf of MYPBX likes below: [MYE1] type=peer host=mye1.abc.com http://mye1.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=did insecure=port,invite Add trustrpid=yes here. A. Why can't I receive the CALLERID from MYPBX(the secondary server)? I am sure I use Set(CALLERID(num) for it. B. Why does the CALLERID that sends from MYE1 become as Anonymous? How can I fix it with the correct orginal callerid(912345678)? C. Why does my FROM message become as Anonymous sip:anonym...@anonymous.invalid instead of 912345...@mye1.abc.com mailto:912345...@mye1.abc.com ? You see this because, even though the number has been made available to you, it's marked as a blocked call. Your server is honoring this and blocking the number when it dials the next server. By using Remote Party ID, you'll be able to carry this information forward to your next server. Regards, -- Trevor Peirce Digital Conceptions Canada http://www.digitalcon.ca 1-888-606-3030 / 250-391-7822 -- _ -- 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] Do i need install Dahdi or libpri ?
Thank you! it's very helpful 2010/2/25 Steve Howes steve-li...@geekinter.net: On 25 Feb 2010, at 02:16, Zhang Shukun wrote: there is a AudioCodes Mediant 2000 out there. i want to realise ip to PSTN and PSTN to ip connection. Ok. after some configuration on AudioCodes Mediant 2000, PSTN to ip connecttion works. Thats good. a, Do i need install DAHDI or libpri in my system? Depends what else you are doing. I'd always just install it anyway. b, how to write in dialplan to realise connection to PSTN. The Mediant 2000 can be used like any other sip device. Dial(SIP/ whatever/1234567890) -- _ -- 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 -- Best regards, Sucan -- _ -- 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] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available
hi, all after my installation of asterisk and adds-on . when start astrisk, error accours as follow: [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available what's wrong with me ? Thanks. -- Best regards, Sucan -- _ -- 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] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available
On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun bit...@gmail.com wrote: hi, all after my installation of asterisk and adds-on . when start astrisk, error accours as follow: [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available-- Is MySQL running and all the proper values set in the appropriate files? Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available
yes. mysql run ok the configuration is ok too. i think is this error shows asterisk can't find mysql database? 2010/2/26 Warren Selby wcse...@selbytech.com: On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun bit...@gmail.com wrote: hi, all after my installation of asterisk and adds-on . when start astrisk, error accours as follow: [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available-- Is MySQL running and all the proper values set in the appropriate files? Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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 -- Best regards, Sucan -- _ -- 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] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available
On Friday 26 February 2010 00:09:55 Warren Selby wrote: On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun bit...@gmail.com wrote: [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available-- Is MySQL running and all the proper values set in the appropriate files? Does the config name in extconfig.conf right after the word mysql exist as a section in res_mysql.conf? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available
2010/2/26 Tilghman Lesher tles...@digium.com: On Friday 26 February 2010 00:09:55 Warren Selby wrote: On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun bit...@gmail.com wrote: [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available-- Is MySQL running and all the proper values set in the appropriate files? Does the config name in extconfig.conf right after the word mysql exist as a section in res_mysql.conf? the section in extconfig.conf is : sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies and section in res_mysql.conf is : [asterisk] ;dbhost = 127.0.0.1 dbname = asterisk dbuser = root dbpass = net263 dbport = 3306 ;dbsock = /tmp/mysql.sock ;dbsock = /var/run/mysqld/mysqld.sock requirements=createclose ; or createclose or createchar -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 -- Best regards, Sucan -- _ -- 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] Morse Code
On Thu, Feb 25, 2010 at 8:00 PM, David Gibbons d...@videon-central.com wrote: Duh! How are we going to spread the word about how to take those alien bastards down if we don't keep morse code around!?!??! And what about if you're trapped in ship that sinks? What if the 3g coverage isn't good? Or you have no more battery? /r -- _ -- 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
