Re: [asterisk-users] Want web page to listen to meetme (WebRTC?)
Quick and drity: 1) Meetme has to be configured to record the media stream. 2) You have to install a streaming server. Maybe ffmpeg could do the job: https://trac.ffmpeg.org/wiki/StreamingGuide 3) Then your website should be able to get the stream from the streaming server. You should be able to test this scenario withing some hours. Am 08.12.2014 16:11, schrieb Steve Edwards: I have a web page to do the usual meetme admin stuff -- mute, kick, etc. Now, the client is asking if they can listen to the meetme -- click and audio comes out the computer speakers. How can this be implemented? Is this a use case for WebRTC? -- _ -- 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] Confbridge
Take a look here: http://asteriskfaqs.org/tag/confbridge/page/2 Am 02.12.2014 03:37, schrieb Bryant Zimmerman: I am doing dynamic conference bridges using confbridge in asterisk 11. Is there a way to toggle off an on recording of an ongoing conference call I have figured out how to record a conference if it is turned on when someone enters. Also I have noticed that when setting music_on_hold_class dynamically it does not override what is set on the channel. exten = s,n,Set(CONFBRIDGE(user,music_on_hold_class)=latin) Does anyone have any ideas on how I might fix this as well? -- _ -- 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] Strange Issue: asterisk deleted
Did you take a look at /var/log/syslog? Am 26.11.2014 21:08, schrieb Antoine Megalla: Hi, I looked for asterisk in /usr/sbin using the commands ls and find and whereis and it was not there. I know that the process is killed because when I start asterisk using the command asterisk -c it starts and then it exits and the word killed is wrote on the console. Ever time I copy a new executable to /usr/sbin either using cp command or make install it gets deleted too. Now I used the strace command on asterisk and I can clearly see at the end of the strace the line : killed by SIGKILL This means that something or someone is actually and purposely killing asterisk but I do not know what or who is doing that also I know that I am the only user on the system. Again any indicators to solve this very weird issue are welcomed. Regards, Antoine Megalla Sent from my iPhone On Nov 26, 2014, at 6:12 PM, Thorsten Göllner t...@ovm-group.com mailto:t...@ovm-group.com wrote: Am 26.11.2014 11:37, schrieb Antoine Megalla: Hi, I am struggling with a very strange issue I have been facing for the past week; I have a fresh install of CENTOS 5.11 and I have installed asterisk 1.8.32 form sources. The asterisk installation went fine but as soon as I start asterisk executable it loads everything and then after the Ready line the process gets killed and when I try to run it again i get: /usr/sbin/asterisk : command not found I cleaned the source and re-installed asterisk and again the same thing happened again !!! I downloaded asterisk versions 1.4, 11, 12 and compiled them from sources and installed them (make install) and amazingly, the same thing happened to all of them: I do a make then make install and as soon as I start asterisk the process is killed and the executable removed from /usr/sbin. I tried to look a the asterisk log files but I cannot find a single error in them. Also if it was really deleted how did bash know that asterisk is supposed to be located in /usr/sbin/asterisk ? I tried to copy the executable myself after compilation (everything done as root) to the /usr/sbin and again if it runs then it is deleted. If someone can explain to me this behavior or advise me on what to check to resolve this issue, then I would be grateful. Hi, you write Also if it was really deleted .. - did you looked at it via ls /usr/sbin/asterisk? You compiled asterisk (make / make install) as root I think. Perhaps access rights are not set properly? root is owner but you try to start the daemon as normal user? You write the process is killed. Where do you now? Did you get a message on your terminal? Did you take a look at /var/log/syslog? -- _ -- 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] Strange Issue: asterisk deleted
Am 26.11.2014 11:37, schrieb Antoine Megalla: Hi, I am struggling with a very strange issue I have been facing for the past week; I have a fresh install of CENTOS 5.11 and I have installed asterisk 1.8.32 form sources. The asterisk installation went fine but as soon as I start asterisk executable it loads everything and then after the Ready line the process gets killed and when I try to run it again i get: /usr/sbin/asterisk : command not found I cleaned the source and re-installed asterisk and again the same thing happened again !!! I downloaded asterisk versions 1.4, 11, 12 and compiled them from sources and installed them (make install) and amazingly, the same thing happened to all of them: I do a make then make install and as soon as I start asterisk the process is killed and the executable removed from /usr/sbin. I tried to look a the asterisk log files but I cannot find a single error in them. Also if it was really deleted how did bash know that asterisk is supposed to be located in /usr/sbin/asterisk ? I tried to copy the executable myself after compilation (everything done as root) to the /usr/sbin and again if it runs then it is deleted. If someone can explain to me this behavior or advise me on what to check to resolve this issue, then I would be grateful. Hi, you write Also if it was really deleted .. - did you looked at it via ls /usr/sbin/asterisk? You compiled asterisk (make / make install) as root I think. Perhaps access rights are not set properly? root is owner but you try to start the daemon as normal user? You write the process is killed. Where do you now? Did you get a message on your terminal? Did you take a look at /var/log/syslog? Best regards -Thorsten- -- _ -- 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] Questions on musiconhold.conf custom mode
Am 27.10.2014 08:54, schrieb Olivier: 2014-10-25 19:33 GMT+02:00 Thorsten Göllner t...@ovm-group.com: Am 25.10.2014 00:09, schrieb Olivier: Hello, I need to play some musiconhold content starting at a random duration from the start. Thanks to mode=custom option and either madplay or mpg123 programs, I could successfully get what I was after on a Debian Wheezy system. Now I realized sox version on my target system (Debian Squeeze) cannot convert to MP3 format. So I'm looking after workarounds. 0. I've read many mpg123 or madplay examples. All of them are clutered with option converting MP3 input file into an appropriate format that Asterisk requires for music on hold. What is the name of this appropriate format ? sln ? wav ? 1. Is there a player like mpg123, that can repeat content in appropriate format (see above) to stdout but can read from anything different from MP3 ? 2. Is there an option on Squeeze to convert audio files to MP3 (reverse coversion works OK). 3. Which options could I have for such custom MOH, if I was building on system without g729 transaltion capabilites ans with g729-only SIP trunks or phones ? Is the gsm-format an option for you? So you may convert your moh-File to gsm: sox YouWavFile.wav -r 8000 -c1 MohFile.gsm Hi Thorsten, Yes gsm-format is an option for me but how can you play such gsm file as MOH ? If I'm not mistaken, both madplay or mpg123 would only play MP3 files (I've not tested with other formats, yet). I could successfully play a RAW file with cat but cat has no repeat option, so I still have to find something else anyway. When your musiconhold.conf looks like that ... cut - [general] [default] mode=files directory=moh [your_moh_class] mode=files directory=/your/path/to/your/moh/files cut - ... then you can put any supported file format into the specified directory. GSM is only one option. Asterisk will take the best (meaning cheapest) file format availble in this directory. If you really need mp3 you have to compile sox with mp3-support by yourself OR maybe this is a solution on Debian: http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/ Yes, you're correct. I'll suggest my customer a Wheezy upgrade. -- _ -- 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] make asterisk do something when an outgoing call is picked up
Am 26.10.2014 00:43, schrieb lee:
Hi,
how can I make asterisk do something when an outgoing call is picked up?
The background is that I would like to record incoming and outgoing
phone calls. In order to do this, I need to play an announcement
telling the person calling or being called that the call will be
recorded.
Here's what I'm trying to do:
call comes in:
if(I pick up) {
play announcement to caller;
start recording;
let me talk to the caller;
end recording when call ends;
send recording to my email account;
} else {
record voice mail;
}
call goes out:
if(call is picked up) {
play announcement to callee;
if(callee hangs up) {
end call;
} else {
start recording;
let me talk to callee;
end recording when call ends;
send recording to my email account;
}
} else {
call ends;
offer me to automatically call again later;
}
Please keep in mind that I'm new to asterisk and just got it to work.
Searching for having asterisk do something when an outgoing call is
picked up has been unsuccessful other than that I found out that you can
have it make outgoing calls automatically to play pre-recorded messages:
So asterisk does have a way to detect when a call is picked up and a way
of doing something when that happens.
What I have working so far is incoming and outgoing calls and voicemail
for one phone/user, which is a basic set up I'm trying extend and
improve now.
Maybe this will do a good job for recording all calls:
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
And playing an announcement, when a call is picked, should be done
within your dialplan with this function:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Playback
-Thorsten-
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Re: [asterisk-users] Questions on musiconhold.conf custom mode
Am 25.10.2014 00:09, schrieb Olivier: Hello, I need to play some musiconhold content starting at a random duration from the start. Thanks to mode=custom option and either madplay or mpg123 programs, I could successfully get what I was after on a Debian Wheezy system. Now I realized sox version on my target system (Debian Squeeze) cannot convert to MP3 format. So I'm looking after workarounds. 0. I've read many mpg123 or madplay examples. All of them are clutered with option converting MP3 input file into an appropriate format that Asterisk requires for music on hold. What is the name of this appropriate format ? sln ? wav ? 1. Is there a player like mpg123, that can repeat content in appropriate format (see above) to stdout but can read from anything different from MP3 ? 2. Is there an option on Squeeze to convert audio files to MP3 (reverse coversion works OK). 3. Which options could I have for such custom MOH, if I was building on system without g729 transaltion capabilites ans with g729-only SIP trunks or phones ? Is the gsm-format an option for you? So you may convert your moh-File to gsm: sox YouWavFile.wav -r 8000 -c1 MohFile.gsm If you really need mp3 you have to compile sox with mp3-support by yourself OR maybe this is a solution on Debian: http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/ -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge / internal_sample_rate=auto / warning
Hi there, I am running Asterisk 11.9.0 WANPIPE Release: 7.0.10 DAHDI Version: 2.9.0 Echo Canceller: HWEC libpri version: 1.4.12 When I start the ConfBridge application I get the following warning: [2014-10-24 14:36:21] WARNING[29177][C-6934]: config_options.c:790 uint_handler_fn: Attempted to set internal_sample_rate=auto, but set it to 0 instead due to default) I do not specify a specific user- or bridge-profile so the default profiles are used in confbridge.conf (and there the profiles are empty). But before calling the ConfBridge-App I set the channel var internal_sample_rate=auto. Am I making a mistake or is it an indicator for a wrong configuration? Best regards, -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages
Hi, I have Asterisk 11 with DAHDI (Sangoma E1-Card) running on Ubuntu 12.04 LTS. Asterisk and DAHDI-Drivers are installed from source. When doing an apt-get upgrade the system packages will be update but sometimes Asterisk is broken. Which packages do I have to exclude when I do not have time to recompile Asterisk/Dahdi each time? libc? Kernel-Packages? Thanks so far! -Thorsten- -- _ -- 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] mixmonitor - convert wav to mp3/aac
Am 18.09.2014 11:06, schrieb Marek Cervenka:
hi,
i want convert mixmonitor recorded speech audio from wav to mp3 or aac
can you recommend your settings for speech audio? filters, noise
elimination, compression ratio, ...
i will probably use lame
Give sox with compiled mp3-support a try:
/usr/bin/sox ${src_file} ${dst_file} lowpass 4000 compand 0.02,0.05
-60,-60,-30,-10,-20,-8,-5,-8,-2,-8 -8 -7 0.05
I found it on another website ... but I can't remember. Works fine for me.
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[asterisk-users] Voice-Recognition / ASR / with barge in
Hi there, I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine :-) But I am wondering if there is a solution/application which will enable me to implement voice recognition while playing a voice file (barge in). So that the caller hears a voice file and can interrupt it with his voice. Currently (on our platform) the caller has to wait for the end of the voicefie. Then we play a beep. And then we record his voice and realize voice recognition with ispeech (it is an online service). Best regards -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk secure fine tune - stop attack
Am 04.09.2014 16:44, schrieb motty cruz: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306 http://213.136.81.166:9306) to extension '34422' rejected because extension not found in context 'default'. You should not invest time in blocking single IPs. Take a look at fail2ban. http://www.fail2ban.org/wiki/index.php/Asterisk -Thorsten- -- _ -- 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] AGI scripts - delay issue.
Am 02.09.2014 07:09, schrieb Bryant Zimmerman: Hey All We have several AGI scripts that access databases. These work well most of the time. The issue we are having is that on rare occasion our script must fail to a backup database server. When this occurs it may take up to two seconds to do so. The issue is when there is this delay the script loses access to read global channel variable values only after the delay. This is driving me crazy is there some kind of AGI timeout issue or bug that could be causing this. What do you mean with the script loses access to read global channel variable values? What is the asterisk version? What channel tech is used? What type of AGI-Scripts do you use? -- _ -- 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] Copying menuselect options
Am 14.08.2014 17:22, schrieb Mitch Claborn: Is it possible (and advisable) to copy menuselect options from Asterisk 11 to Asterisk 12? If so, is menuselect.makeopts the only file to copy? I am not sure - but I would'nt do that. Make a hardcopy from your console and transcribe the settings to your new installation. It yould take you not more than 10 minutes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Record File / what does randomerror mean? res_agi.c / line 2377
Hi,
I have a question about this here:
Asterisk-Version: 11.10.2
File: res/res_agi.c
Line: 2377
[...]
static int handle_recordfile(struct ast_channel *chan, AGI *agi, int
argc, const char * const argv[])
2304 {
2305 struct ast_filestream *fs;
2306 struct ast_frame *f;
2307 struct timeval start;
2308 long sample_offset = 0;
2309 int res = 0;
2310 int ms;
[...]
2365 /* backward compatibility, if no offset given, arg[6] would
have been
2366 * caught below and taken to be a beep, else if it is a
digit then it is a
2367 * offset */
2368 if ((argc 6) (sscanf(argv[6], %30ld, sample_offset)
!= 1) (!strchr(argv[6], '=')))
2369 res = ast_streamfile(chan, beep,
ast_channel_language(chan));
2370
2371 if ((argc 7) (!strchr(argv[7], '=')))
2372 res = ast_streamfile(chan, beep,
ast_channel_language(chan));
2373
2374 if (!res)
2375 res = ast_waitstream(chan, argv[4]);
2376 if (res) {
2377 ast_agi_send(agi-fd, chan, 200 result=%d
(randomerror) endpos=%ld\n, res, sample_offset);
2378 } else {
2379 fs = ast_writefile(argv[2], argv[3], NULL, O_CREAT
| O_WRONLY | (sample_offset ? O_APPEND : 0), 0, AST_FILE_MODE);
2380 if (!fs) {
2381 res = -1;
2382 ast_agi_send(agi-fd, chan, 200 result=%d
(writefile)\n, res);
2383 if (sildet)
2384 ast_dsp_free(sildet);
2385 return RESULT_FAILURE;
2386 }
In line 2377 I find randomerror. And in fact I get this error
sometimes in my AGI-Scripts but can not reproduce them by my own.
Can anybody tell me please, when this message will be fired? I do not
really understand this source at this point. The message will be sent,
when res is true (or larger 0). res should be set in the lines
before. MAYBE res is 0 in line 2377 when the call hangs up at this point?
Best regards
-Thorsten-
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Re: [asterisk-users] Voicemail message to text
Hi, we implemented ispeech for voice recognition. I works fine. But you have to develop an app of your own to do it. Take a look at http://www.ispeech.org/api (Section 3 Automated Speech Recognition). ispeech let you upload a recorded speex file via http-upload and will return the result at once as http-result. On their website you will find some code also to implement their service in any app. It's simple and you will get a quick result. Best regards -Thorsten- Am 20.05.2014 16:35, schrieb Ishfaq Malik: HI there I was wondering if anyone has implemented voicemail to text and if so, what package is being used to do so? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e:i...@pack-net.co.uk mailto:i...@pack-net.co.uk w:http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] 2 PRI Card - Interrupt Problem
Look for irqbalancer for your distribution: http://www.tutorialspoint.com/unix_commands/irqbalance.htm Am 14.05.2014 09:00, schrieb Chandrakant Solanki: Hello All, I have 2 Digium card configure on Single machine, which can't share interrupt across all CPUs and sometimes asterisk reach 100% CPU usage. Here is system details and /proc/interrupt o/p. OS: CentOS 6.4 Kernel: 2.6.32-431.11.2.el6.x86_64 Dahdi Version: DAHDI Version: 2.7.0.2 Echo Canceller: HWEC Asterisk Version: 1.8.13.0 Output: /proc/interrupts cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 ... 37:1132730 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp 39:1132831 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp ... Thanks. -- Chandrakant Solanki -- _ -- 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] Kernel and DAHDI
That's correct. When you update the kernel package youhave also to recompile dahdi package. Am 12.05.2014 07:05, schrieb Lee, John (Sydney): Hi, I have noticed it for a while but I just thought about confirming this with the Asterisk community. As the compilation of DAHDI will need to reference Kernel-devel, does it mean that after DAHDI is installed, we should not yum update kernel because it will affect the operation of DAHDI? Thanks. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- _ -- 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] Inbound DAHDI Error
Hi, it seems, that the caller hangs up immediatly after calling. Try to reproduce it by yourself. Dial the number (to reach your asterisk server) and hangup after ~ 0.5 sec (or whatever). Best regards, -Thorsten- Am 30.04.2014 01:11, schrieb Bryce Lowe: Hello, I am trying to diagnose an intermittent error when a call comes in over our PRI lines. The problem appears random, however I have feeling it has something to do with the call volume, as the frequency increases with more calls on the system. I am not an expert when it comes to reading the PRI Span Debug statements but here is a call that had a problem and I bolded, italicized, and underlined the part of the debug statement that looks odd (listed under PRI Debug Output (failed call)). Any help is appreciated. Thanks, Bryce *Version(s):* ** Asterisk 11.8.1, installed from the Digium YUM Repositories DAHDI Version: 2.9.0 Digium Card: Wildcard TE235 (VPMOCT064) OS: CentOS 6.5 *My Observations:* ** When I have the problem, the only way I see that Asterisk received a signal on my PRI lines was through the pri debug statements, I don't see anything being hit in the dialplan (for instance the NoOp at the start of my sub-dial-cudatel-extension sub context). Is there another tool I should be using to debug this issue? *PRI Debug Output (failed call):* ** PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=73 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 23832/0x5D18) (Sent from originator) PRI Span: 1 Message Type: SETUP (5) PRI Span: 1 [04 03 80 90 a2] PRI Span: 1 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 1 User information layer 1: u-Law (34) PRI Span: 1 [18 03 a1 83 81] PRI Span: 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0 PRI Span: 1 ChanSel: As indicated in following octets PRI Span: 1 Ext: 1 Coding: 0 Number Specified Channel Type: 3 PRI Span: 1 Ext: 1 Channel: 1 Type: CPE] PRI Span: 1 [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 4f 4d 41 58 20 43 4f 52 50 20 4e 20 47 53 4d] PRI Span: 1 Facility (len=31, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0F, 'source_caller_name' ] PRI Span: 1 [6c 0c 21 83 32 35 33 33 38 30 35 35 39 31] PRI Span: 1 Calling Party Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) PRI Span: 1 Presentation: Presentation allowed, Network provided (3) 'calling_caller_id' ] PRI Span: 1 [70 0b a1 32 35 33 38 37 32 32 33 30 30] PRI Span: 1 Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'dest_number' ] PRI Span: 1 -- Making new call for cref 23832 PRI Span: 1 Received message for call 0x7f7a900012f0 on link 0x1a3cf70 TEI/SAPI 0/0 PRI Span: 1 -- Processing Q.931 Call Setup PRI Span: 1 -- Processing IE 4 (cs0, Bearer Capability) PRI Span: 1 -- Processing IE 24 (cs0, Channel ID) PRI Span: 1 -- Processing IE 28 (cs0, Facility) PRI Span: 1 -- Processing IE 108 (cs0, Calling Party Number) PRI Span: 1 -- Processing IE 112 (cs0, Called Party Number) PRI Span: 1 -- Delayed processing IE 28 (cs0, Facility) PRI Span: 1 ASN.1 dump PRI Span: 1 Context Specific [11 0x0B] 8B Len:1 01 PRI Span: 1 00 - ~ PRI Span: 1 Context Specific/C [1 0x01] A1 Len:23 17 PRI Span: 1 Integer(2 0x02) 02 Len:1 01 PRI Span: 1 01 - ~ PRI Span: 1 Integer(2 0x02) 02 Len:1 01 PRI Span: 1 00 - ~ PRI Span: 1 Context Specific [0 0x00] 80 Len:15 0F PRI Span: 1 4F 4D 41 58 20 43 4F 52-50 20 4E 20 47 53 4D - source_caller_name PRI Span: 1 ASN.1 end PRI Span: 1 interpretation Context Specific [11 0x0B] = 0 0x PRI Span: 1 INVOKE Component Context Specific/C [1 0x01] PRI Span: 1 invokeId Integer(2 0x02) = 1 0x0001 PRI Span: 1 operationValue Integer(2 0x02) = 0 0x PRI Span: 1 operationValue = ROSE_QSIG_CallingName PRI Span: 1 callingName Name PRI Span: 1 namePresentationAllowedSimple Context Specific [0 0x00] = PRI Span: 1 4F 4D 41 58 20 43 4F 52-50 20 4E 20 47 53 4D - source_caller_name PRI Span: 1 q931.c:8646 post_handle_q931_message: Call 23832 enters state 6 (Call Present). Hold state: Idle Span 1: Processing event PRI_EVENT_RING(5) */_PRI Span: 1 q931.c:7135 q931_hangup: Hangup other cref:23832_/* */_PRI Span: 1 q931.c:6892 __q931_hangup: ourstate Call Present, peerstate Call Initiated, hold-state Idle_/* */_PRI Span: 1 q931.c:6081 q931_disconnect: Call 23832 enters state 11 (Disconnect Request). Hold state: Idle_/* PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=73 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 23832/0x5D18) (Sent from originator) PRI Span: 1 Message Type: SETUP (5) PRI Span: 1 [04 03 80 90 a2] PRI Span: 1 Bearer
Re: [asterisk-users] AGI GET DATA behavior
Is your script really so simple? Enable agi debugging (agi set debug on) and take look at it when this happens. -Thorsten- Am 30.04.2014 11:47, schrieb Hoggins!: Hello all, I have a strange problem with a very simple AGI script, using the GET DATA command. When using this command, Asterisk often returns 0 as a result after a GET DATA beep 5000 command, without even waiting for input from the calling party. It is quite random : sometimes Asterisk behaves exactly as documented, and sometimes it gives 200 result=0 without any reason. Do you have an idea of what might be happening ? I'm using version 11.6.0. Hoggins! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI Banner
Am 28.03.2014 10:32, schrieb Haider Khalil: Hello Experts, I want to know if there is any way to modify welcome banner on asterisk console when I connect using asterisk -r Hi, did you compile asterisk from source? Take a look at main/asterisk.c (line 174 in asterisk v 11.5.1). I think you have to change it there manually and recompile it. -Thorsten- -- _ -- 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] Text to Speech Engine
Take a look at http://www.ispeech.org/ I implemented Speech-Recognition. The API is well documented and easy. Am 10.01.2014 21:16, schrieb Jai Rangi: Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -- _ -- 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] High load on asterisk servers
Do you see no hint in the atserisk console (log)? Am 20.12.2013 09:01, schrieb Henrik Andresen: Hi Stefan, I use own dns-servers on local subnet so I don't think it's the problem :( Also I have hosts in local hosts-files. /Henrik On 19/12/13 14:47, Stefan Schmidt wrote: Maybe this happens if you have a short delay to your dns servers. This could increase the load very fast and after some seconds it might be over again. I have installed a dns recurser with own caching on all of my asterisk servers and now everything runs much more smoothly. best regards stefan Am 19.12.2013 11:56, schrieb Henrik Andresen: All calls are sip--sip On 19/12/13 11:32, Thorsten Göllner wrote: Am 19.12.2013 10:37, schrieb Henrik Andresen: I have a problem with asterisk. I got ~15 asterisk servers on new hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1 and 10. No disk activity, no ram or swap problem. But asterisk main process is using up to 300-500% cpu. This happens both with 30 channels in use and 100+ channels in use. I'm not doing transcoding or anything. any clue ? One server with 300 channels load on 5 One server with 600 channels load on 0.02 After 5 minutes it might be ok... some times its ok after 1 hour. I do no recording, no transcoding just g711a Two servers does not have sip-registrations as they are gateways to our sip-propvider. The other servers got around 1000-1200 sip registrations. Running asterisk 11.5.x, 11.6.0 and now trying 11.7.0 All servers HP with centos 6.5 (has been 6.3 and 6.4 as well) Any clue ? /Henrik What calls cause these problems? SIP or E1/T1-Calls? -- _ -- 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] High load on asterisk servers
What about the load, when only 1 or 2 calls are on this machine? Am 20.12.2013 09:01, schrieb Henrik Andresen: Hi Stefan, I use own dns-servers on local subnet so I don't think it's the problem :( Also I have hosts in local hosts-files. /Henrik On 19/12/13 14:47, Stefan Schmidt wrote: Maybe this happens if you have a short delay to your dns servers. This could increase the load very fast and after some seconds it might be over again. I have installed a dns recurser with own caching on all of my asterisk servers and now everything runs much more smoothly. best regards stefan Am 19.12.2013 11:56, schrieb Henrik Andresen: All calls are sip--sip On 19/12/13 11:32, Thorsten Göllner wrote: Am 19.12.2013 10:37, schrieb Henrik Andresen: I have a problem with asterisk. I got ~15 asterisk servers on new hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1 and 10. No disk activity, no ram or swap problem. But asterisk main process is using up to 300-500% cpu. This happens both with 30 channels in use and 100+ channels in use. I'm not doing transcoding or anything. any clue ? One server with 300 channels load on 5 One server with 600 channels load on 0.02 After 5 minutes it might be ok... some times its ok after 1 hour. I do no recording, no transcoding just g711a Two servers does not have sip-registrations as they are gateways to our sip-propvider. The other servers got around 1000-1200 sip registrations. Running asterisk 11.5.x, 11.6.0 and now trying 11.7.0 All servers HP with centos 6.5 (has been 6.3 and 6.4 as well) Any clue ? /Henrik What calls cause these problems? SIP or E1/T1-Calls? -- _ -- 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] High load on asterisk servers
Am 19.12.2013 10:37, schrieb Henrik Andresen: I have a problem with asterisk. I got ~15 asterisk servers on new hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1 and 10. No disk activity, no ram or swap problem. But asterisk main process is using up to 300-500% cpu. This happens both with 30 channels in use and 100+ channels in use. I'm not doing transcoding or anything. any clue ? One server with 300 channels load on 5 One server with 600 channels load on 0.02 After 5 minutes it might be ok... some times its ok after 1 hour. I do no recording, no transcoding just g711a Two servers does not have sip-registrations as they are gateways to our sip-propvider. The other servers got around 1000-1200 sip registrations. Running asterisk 11.5.x, 11.6.0 and now trying 11.7.0 All servers HP with centos 6.5 (has been 6.3 and 6.4 as well) Any clue ? /Henrik What calls cause these problems? SIP or E1/T1-Calls? -- _ -- 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] [NEWBIE] Right dect to buy to use with asterisk
Hi, I made good experienes with Siemens Gigaset C610 IP. This model is about 90 Euro. Configuration via web interface. But encryption (SIPS/SRTP) is *not* possible with this phones. -Thorsten- Am 11.12.2013 11:30, schrieb Mario Giammarco: Hello, I need to setup this configuration: - asterisk as IVR; - dect phones. So basically I need a standard set of features: - each dect phone has its extension so I can call it directly; - handover of a call with R key; - if a call is not replied by someone ring all phones. I have little budget. I can choose to buy a fritz!box or a gigasect dect/ip base station. Which one should I buy? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk / SIP-Call / AGI-Script / SIGHUP after Answer
Hi, I am facing a (for me) strange problem. When placing a SIP-Call I normally get connected and the dialplan is executed. The Call-Flow is controlled by a PHP-Agi-Script. The script answers the call (via AGI-Command) and a simple voicefile is played. SOMETIMES(!) I get disconnected immediately after the Answer - without any reason. This happens about all fifth call. Later on you will find my SIP-Debug-Output. I can see a BYE-Message. But why? AGI-Debug-Messages: (yes - I can the result is -1 but why? Normally it is 0) -- snip -- SIP/thorsten-01f8AGI Rx Answer SIP/thorsten-01f8AGI Tx 200 result=-1 -- snip -- SIP-Debug-Messages: -- snip -- --- SIP read from UDP:217.92.105.86:51861 --- INVITE sip:3...@myhost.org SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:51861;rport;branch=z9hG4bKPj47b1a62ac3744acd996426618d90388f Max-Forwards: 70 From: Thorsten (myhost) sip:thors...@myhost.org;tag=4313e82f4af9423bab056113e5e05713 To: sip:3...@myhost.org Contact: sip:03794281@192.168.1.2:51861 Call-ID: a19e81e8a2d74f718e1263ab3fd3b328 CSeq: 28484 INVITE Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: 100rel, replaces, norefersub, gruu User-Agent: Blink 0.5.0 (Windows) Content-Type: application/sdp Content-Length: 386 v=0 o=- 3589198761 3589198761 IN IP4 192.168.1.2 s=Blink 0.5.0 (Windows) c=IN IP4 192.168.1.2 t=0 0 m=audio 10054 RTP/AVP 108 99 98 9 0 8 96 c=IN IP4 192.168.1.2 a=rtcp:10055 a=rtpmap:108 opus/48000 a=rtpmap:99 speex/32000 a=rtpmap:98 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv - --- (13 headers 17 lines) --- Sending to 217.92.105.86:51861 (no NAT) Sending to 217.92.105.86:51861 (no NAT) Using INVITE request as basis request - a19e81e8a2d74f718e1263ab3fd3b328 Found peer 'thorsten' for 'thorsten' from 217.92.105.86:51861 --- Reliably Transmitting (NAT) to 217.92.105.86:51861 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:51861;branch=z9hG4bKPj47b1a62ac3744acd996426618d90388f;received=217.92.105.86;rport=51861 From: Thorsten (myhost) sip:thors...@myhost.org;tag=4313e82f4af9423bab056113e5e05713 To: sip:3...@myhost.org;tag=as7b1fc32b Call-ID: a19e81e8a2d74f718e1263ab3fd3b328 CSeq: 28484 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=myhost, nonce=0d688867 Content-Length: 0 Scheduling destruction of SIP dialog 'a19e81e8a2d74f718e1263ab3fd3b328' in 32000 ms (Method: INVITE) --- SIP read from UDP:217.92.105.86:51861 --- ACK sip:3...@myhost.org SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:51861;rport;branch=z9hG4bKPj47b1a62ac3744acd996426618d90388f Max-Forwards: 70 From: Thorsten (myhost) sip:thors...@myhost.org;tag=4313e82f4af9423bab056113e5e05713 To: sip:3...@myhost.org;tag=as7b1fc32b Call-ID: a19e81e8a2d74f718e1263ab3fd3b328 CSeq: 28484 ACK User-Agent: Blink 0.5.0 (Windows) Content-Length: 0 - --- (9 headers 0 lines) --- --- SIP read from UDP:217.92.105.86:51861 --- INVITE sip:3...@myhost.org SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:51861;rport;branch=z9hG4bKPj71edb9caa0e84a52b14777e7d949bc2a Max-Forwards: 70 From: Thorsten (myhost) sip:thors...@myhost.org;tag=4313e82f4af9423bab056113e5e05713 To: sip:3...@myhost.org Contact: sip:03794281@192.168.1.2:51861 Call-ID: a19e81e8a2d74f718e1263ab3fd3b328 CSeq: 28485 INVITE Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: 100rel, replaces, norefersub, gruu User-Agent: Blink 0.5.0 (Windows) Authorization: Digest username=thorsten, realm=myhost, nonce=0d688867, uri=sip:3...@myhost.org, response=c1a2ab209d255b4ee805edd4de48380a, algorithm=MD5 Content-Type: application/sdp Content-Length: 386 v=0 o=- 3589198761 3589198761 IN IP4 192.168.1.2 s=Blink 0.5.0 (Windows) c=IN IP4 192.168.1.2 t=0 0 m=audio 10054 RTP/AVP 108 99 98 9 0 8 96 c=IN IP4 192.168.1.2 a=rtcp:10055 a=rtpmap:108 opus/48000 a=rtpmap:99 speex/32000 a=rtpmap:98 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv - --- (14 headers 17 lines) --- Sending to 217.92.105.86:51861 (NAT) Using INVITE request as basis request - a19e81e8a2d74f718e1263ab3fd3b328 Found peer 'thorsten' for 'thorsten' from 217.92.105.86:51861 == Using SIP RTP CoS mark 5 Found RTP audio format 108 Found RTP audio format 99 Found RTP audio format 98 Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 96 Found unknown media description format opus for ID 108 Found audio description format speex for ID 99 Found audio description format speex for ID 98 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format
Re: [asterisk-users] RTP port ranges
Maybe this could help you: http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf Am 13.09.2013 11:49, schrieb Jonas Kellens: Hello, and when I define 11500 - 11954 it should use a random port in this range. Where is it stated that you MUST use 1-2 ??? Someone else please ? Jonas. On 09/13/2013 11:46 AM, Andrew Colin wrote: Because normally it will use a random port between them On 9/13/2013 11:43 AM, Jonas Kellens wrote: On 09/13/2013 11:41 AM, Andrew Colin wrote: Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Jonas. -- _ -- 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] dahdi configuration issue
Did you open a ticket at Sangoma-Site? What wanpipe driver version do you use? Is it a production machine? Or can you test it in that way, that you crossover lines from one card to the other? Am 04.09.2013 10:48, schrieb DHAVAL INDRODIYA: Hello List, I have configure 2 sangoma card each with 8 PRI lines with dahdi 2.6 the problem is i can see all channels configured in dahdi_cfg 480 channels configured but when I see /dev/dahdi i can only see 240 channels. what could be problem I am using it wanrouter and when I put PRI in new card i only got calls on new line that means one of the card is inactive at same time all the lines and alarms are okay only suspected thing is /dev/dahdi. is there nany setting in linux or kernel level which need to be set for solve this issue. any help appreciated. Thanking You --Dhaval -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip-Client / type=peer / Why can this client place calls?
Hi,
I am using Asterisk 11.5.1. As far as I understood, the following
configuration allows a sip client only to receive calls (type=peer) but
not to place calls
(http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I place
calls though with this config?
sip.conf
...
[thorsten]
type=peer
host=dynamic
context=my_context
nat=force_rport,comedia
secret=...
dtmfmode=rfc2833
disallow=all
allow=g722
allow=g729
allow=g729
...
extensions.conf
...
[my_context]
exten = _X.,1,Dial(DAHDI/g1/${EXTEN},60)
...
Of course: when removing a valid context the client can not place the
call. But I thought this behaviour can be controlled via type=peer?!
Thanks in advance
-Thorsten-
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[asterisk-users] Asterisk 11.5.1 / TLS and Media Encryption / Blink as Client / no audio
Hi, I use Asterisk 11.5.1 and it works fine. :) Now I want to use TLS and media encryption. I followed this guide: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial When I place a call via Blink-Client (0.5.0) I get connected and Blink shows 2 locks. The blue lock shows Signaling is encrypted using TLS and the orange lock shows Media is encrypted using sRTP. BUT i hear no audio. After ~60 seconds I get the following message: NOTICE[21005]: chan_sip.c:28800 check_rtp_timeout: Disconnecting call 'SIP/tgoellner-002c' for lack of RTP activity in 62 seconds sip show peers shows me, that my Blink-Client is registered on port 60071. All other SIP-Clients (no TLS an no media encryption) are registered at port 5060. I tried to open the tcp and udp port range from 1 to 61000 (in iptables). But with no success. I am not sure, but I think it's a firewall/NAT problem?! (Yes, my client is behind a router NAT) Any idea? -Thorsten- -- _ -- 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-Client / type=peer / Why can this client place calls?
Thanks a lot. Seems to be a good hidden page, isn't it? ;-)
Am 03.09.2013 14:30, schrieb Steve Totaro:
On Tue, Sep 3, 2013 at 8:11 AM, Thorsten Göllner t...@ovm-group.com
mailto:t...@ovm-group.com wrote:
Hi,
I am using Asterisk 11.5.1. As far as I understood, the following
configuration allows a sip client only to receive calls
(type=peer) but not to place calls
(http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I
place calls though with this config?
sip.conf
...
[thorsten]
type=peer
host=dynamic
context=my_context
nat=force_rport,comedia
secret=...
dtmfmode=rfc2833
disallow=all
allow=g722
allow=g729
allow=g729
...
extensions.conf
...
[my_context]
exten = _X.,1,Dial(DAHDI/g1/${EXTEN},60)
...
Of course: when removing a valid context the client can not place
the call. But I thought this behaviour can be controlled via
type=peer?!
Thanks in advance
-Thorsten-
See if this is helpful.
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
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Re: [asterisk-users] Installing asterisk and dahdi on ubuntu
Permissions: take a look at /etc/udev/rules.d/dahdi.rules. Last line. OWNER and GROUP should be the same as the user running the asterisk process (root or asterisk?). Am 29.08.2013 11:47, schrieb bilal ghayyad: Hello; I am installing asterisk and dahdi on ubuntu and I used my username bghayad to login for ubuntu and do the installation, actually I feel my problem is related to the username and permission but I am not able how to fix it, I am facing now mainly the following two problems: The first one, asterisk is not starting automatically although I did sudo make config (for asterisk and dahdi) and the asterisk and dahdi scripts have been created under /etc/init.d/ The second problem, I started asterisk using asterisk -cvvv and from the CLI, I tried dahdi show version and dahdi show status, I am getting the following results: *CLI dahdi show status No DAHDI found. Unable to open /dev/dahdi/ctl: Permission denied Command 'dahdi show status ' failed. *CLI dahdi show version Failed to open control file to get version. Below is my ubuntu information: bghayad@Bilal:/usr/sbin$ lsb_release -a No LSB modules are available. Distributor ID: Ubuntu Description:Ubuntu 12.04.1 LTS Release:12.04 Codename: precise -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk ip authentication
You should take a look at this options: type=friend context=my_context host=ip_address Am 26.07.2013 16:52, schrieb jin jan: Hi all, I've tried to sen calls to asterisk from different soft switch. I want to define ip authentication(not register) to an extension for make call through asterisk. Is there any way to make call from asterisk without register. Only ip authentication. I tried too many different configurations but it hasn't worked. This is my sip.conf --sip.conf [] host=x.x.x.x qualify=yes type=peer insecure=port,invite context=from-internal disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm But gives SIP/2.0 401 Unauthorized error. Kind Regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk ip authentication
Additionally you shoudl take a look at sip set debug on (in cli) and then place a call. Am 26.07.2013 17:14, schrieb Thorsten Göllner: You should take a look at this options: type=friend context=my_context host=ip_address Am 26.07.2013 16:52, schrieb jin jan: Hi all, I've tried to sen calls to asterisk from different soft switch. I want to define ip authentication(not register) to an extension for make call through asterisk. Is there any way to make call from asterisk without register. Only ip authentication. I tried too many different configurations but it hasn't worked. This is my sip.conf --sip.conf [] host=x.x.x.x qualify=yes type=peer insecure=port,invite context=from-internal disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm But gives SIP/2.0 401 Unauthorized error. Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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] limitation on number of contexts in extensions.conf
Enter CLI via /usr/sbin/asterisk -r and execute dialplan reload. Any errors? BTW: you should think about upgrading to 1.8 (for example). Am 25.07.2013 08:49, schrieb Kamlesh Kumar: Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include filename) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. Regards, Kamlesh -- _ -- 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] Mysql Support int Asterik-11
Why not use ODBC? Am 24.07.2013 13:41, schrieb Prashant Abhang: Hi, I was having question about mysql driver support ( not odbc). Do we still need the asterisk-add-on to be installed for mysql support. If yes, Which version should be used and from where I should get it? Thanks in adavance. Thanks Regards, PrashantAbhang -- _ -- 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] Which is the stable version to use?
Depends on used kernel and perhaps on other hardware you are using. Am 23.07.2013 00:09, schrieb bilal ghayyad: Hello I need to deploy asterisk on production and same thing for DAHDI, which version is recommended for this? Regards Bilal -- _ -- 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] How to increase the calls per second limit ?
Hi,
where did you change the ulimit? The following command should show you,
if your setting is correct:
asterisk -rx ulimit descriptors
In my installation I edited the limits here:
vi /etc/security/limits.conf
[...]
asterisksoftnofile 8192
asteriskhardnofile 32768
#EOF
This assumes, that asterisk runs under the user asterisk (first column
is the user). I am not sure if a reboot is neccessary after changing
this file. Give it a try and check the asterisk process with the above
given command.
-Thorsten-
Am 21.06.2013 23:23, schrieb Olivier:
Hello,
As an exercice, I installed sipp on the same box as a Asterisk 11.4
instance (to keep network equipements out of the equation).
I'm focusing on the maximum number of new calls this Asterisk instance
can deal with.
Here is the dialplan (AEL) I'm playing with:
_X. = {
Verbose(0,Incoming call from ${CALLERID(num)} to ${EXTEN} in
${CONTEXT} - case A);
Answer();
MusicOnHold(default,20);
HangUp();
};
For now, I'm repeatedly hitting a 35 cps limit (with a small 4%
failure rate, all of them occuring at the end of a 200 calls wave).
When this occurred for the first time, I could read a Too many open
file error while sipp calls failed. At that time, ulimit and stack
wre respectively set to 2048 and 8192.
Then I increased those settings to 32768 (from 2048) and 2048 (unchanged).
Now I'm still hitting the same 35 cps limit but Asterisk displays :
res_musiconhold.c:343 ast_moh_files_next: Unable to open file
'/var/lib/asterisk/moh/reno_project-system': No such file or directory
As you may guess, the above file exists so I suppose I'm hitting
another limit but I can't find it.
Suggestions ?
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Re: [asterisk-users] Asterisk / PHP-AGI / pthreads
Hi Satish :) You reminded me of my teacher of old school days. Very well explained. I have somewhat similar requirement where I need to play some announcements to entertain a caller while passing/processing some data through webservice call (). do you want to use C or PHP? -Thorsten- -- _ -- 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] MOH don't work after update
Take a look here: http://blog.our-files.com/2012/07/format_mp3-so-building-for-asterisk-1-8-11-using-packages-asterisk-org/ Am 16.06.2013 09:43, schrieb Olivier CALVANO: Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c, Fermeture) in new stack [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for Fermeture -- Executing [334xx@Accueil_Phibee_HNO:4] Hangup(SIP/SIP05-000c, ) in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP05-000c' I understand that he search the file in .ulaw, but why i don't use the mp3 ? musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/moh [Horaires] mode=quietmp3 directory=/var/lib/asterisk/moh/Horaires ps fax: 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 find /var/lib/asterisk/moh/ /var/lib/asterisk/moh/Horaires/Fermeture.mp3 ll -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010 /var/lib/asterisk/moh/Horaires/Fermeture.mp3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk / PHP-AGI / pthreads
Hi there, does anyone have experience with Asterisk-AGI-Scripts in PHP while using pthreads in PHP? Are there any limitations or problems known? Best regards -Thorsten- -- _ -- 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] MOH don't work after update
Is the subdir Horaires readable/executable for User Asterisk/Asterisk? Did you try to convert it to wav? Am 17.06.2013 09:47, schrieb Thorsten Göllner: Take a look here: http://blog.our-files.com/2012/07/format_mp3-so-building-for-asterisk-1-8-11-using-packages-asterisk-org/ Am 16.06.2013 09:43, schrieb Olivier CALVANO: Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c, Fermeture) in new stack [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for Fermeture -- Executing [334xx@Accueil_Phibee_HNO:4] Hangup(SIP/SIP05-000c, ) in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP05-000c' I understand that he search the file in .ulaw, but why i don't use the mp3 ? musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/moh [Horaires] mode=quietmp3 directory=/var/lib/asterisk/moh/Horaires ps fax: 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 find /var/lib/asterisk/moh/ /var/lib/asterisk/moh/Horaires/Fermeture.mp3 ll -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010 /var/lib/asterisk/moh/Horaires/Fermeture.mp3 -- _ -- 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] Light-weight voice recognition for IVR
Hi, some month ago we installed a VoiceRec-Module from Vestec (https://www.vestec.com/) on Asterisk 11.x. It works so far and you will find examples for your dialplan. It should be ok for your needs. -Thorsten- Am 13.06.2013 23:19, schrieb asterisk users: Hello list, 'Just wondering if anyone can point to a very light-weight and easy to incorporate into Asterisk (v. 11.x) to handle a minimal set of responses, like: 0 - 9 yes no (maybe * and # for some people) The idea is that within an IVR menu, the caller could respond by speaking to the typical IVR options, like: For Archie, press or say 1 now For Veronica, press or say 2 now For Jughead, press or say 3 now (etc.) You have selected option 2 for Veronica, press 1 or say yes if this is correct. If a voice response was received (not a DTMF key press) indeterminate, some status would be useful (beyond just a timeout). It would be great if this was simple to code into the dialplan, much like like the current background/wait model for keypresses. Low cost or free would be nice too! Thanks for any suggestions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming DAHDI Channel explained
Hi, I use a Sangoma A104d-Card (with 4 x germany E1). I process some calls via an AGI-Script. When parsing the AGI-Variables I can see one that look like that: [agi_channel] = DAHDI/i3/211123456-89c What hat do the values mean in detail, please? DAHDI : this is clear i3 : does it mean, that the call comes in via E1-Port 3? 211123456 : Incoming-Call Caller-ID -89c : ? WANPIPE Release: 7.0.1 DAHDI Version: 2.6.2 Echo Canceller: HWEC libpri version: 1.4.12 Best regards -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial-App / Feature Disconnect
Hi, I configured in features.conf, that the Dial-App may be cancelled by pressing the pound key. That works fine. The caller can cancel the bridged call. BUT can I configure it that way, that the dialing itself can NOT be cancelled? My dial should only be cancelled by the timeout or by the gangup of the caller. Asterisk 11.3.0 Best regards, -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pri-Debug-Log / Is Early Media supported by provider?
Hi, I tried to use Early Media: exten = 1,1,Playback(demo-thanks,noanswer) same = n,Hangup() But when calling my extension I do not hear the voicefile - I only hear the ring tone. In the Asterisk-Log I can see, that the voicefile is played. I got the same result when using Progress() in the first priority. I tried pri set debug on span 1 and got the following: (I replaced originating caller id by 123456) PRI Span: 1 Protocol Discriminator: Q.931 (8) len=48 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent from originator) PRI Span: 1 Message Type: SETUP (5) PRI Span: 1 [a1] PRI Span: 1 Sending Complete (len= 1) PRI Span: 1 [04 03 80 90 a3] PRI Span: 1 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 1 User information layer 1: A-Law (35) PRI Span: 1 [18 03 a9 83 8e] PRI Span: 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 PRI Span: 1ChanSel: As indicated in following octets PRI Span: 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 PRI Span: 1Ext: 1 Channel: 14 Type: CPE] PRI Span: 1 [6c 0c 21 83 31 37 38 31 34 38 34 31 34 32] PRI Span: 1 Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) PRI Span: 1Presentation: Presentation allowed of network provided number (3) '123456' ] PRI Span: 1 [70 0c c1 36 30 32 31 32 35 30 30 30 33 30] PRI Span: 1 Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1' ] PRI Span: 1 [7d 02 91 81] PRI Span: 1 IE: High-layer Compatibility (len = 4) PRI Span: 1 -- Making new call for cref 14783 PRI Span: 1 Received message for call 0x7f48ec00a370 on link 0x7f49201859a0 TEI/SAPI 0/0 PRI Span: 1 -- Processing Q.931 Call Setup PRI Span: 1 -- Processing IE 161 (cs0, Sending Complete) PRI Span: 1 -- Processing IE 4 (cs0, Bearer Capability) PRI Span: 1 -- Processing IE 24 (cs0, Channel Identification) PRI Span: 1 -- Processing IE 108 (cs0, Calling Party Number) PRI Span: 1 -- Processing IE 112 (cs0, Called Party Number) PRI Span: 1 -- Processing IE 125 (cs0, High-layer Compatibility) PRI Span: 1 q931.c:8281 post_handle_q931_message: Call 14783 enters state 6 (Call Present). Hold state: Idle Span 1: Processing event PRI_EVENT_RING(5) PRI Span: 1 q931.c:5477 q931_call_proceeding: Call 14783 enters state 9 (Incoming Call Proceeding). Hold state: Idle PRI Span: 1 PRI Span: 1 DL-DATA request PRI Span: 1 Protocol Discriminator: Q.931 (8) len=10 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to originator) PRI Span: 1 Message Type: CALL PROCEEDING (2) PRI Span: 1 TEI=0 Transmitting N(S)=70, window is open V(A)=70 K=7 PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=10 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to originator) PRI Span: 1 Message Type: CALL PROCEEDING (2) PRI Span: 1 [18 03 a9 83 8e] PRI Span: 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 PRI Span: 1ChanSel: As indicated in following octets PRI Span: 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 PRI Span: 1Ext: 1 Channel: 14 Type: CPE] -- Accepting call from '123456' to '1' on channel 0/14, span 1 -- Executing [1@port1:1] NoOp(DAHDI/i1/123456-245, ) in new stack -- Executing [1@port1:2] Playback(DAHDI/i1/123456-245, demo-thanks,noanswer) in new stack -- DAHDI/i1/123456-245 Playing 'demo-thanks.gsm' (language 'de_female') -- Executing [1@port1:3] Hangup(DAHDI/i1/123456-245, ) in new stack == Spawn extension (port1, 1, 3) exited non-zero on 'DAHDI/i1/123456-245' PRI Span: 1 q931.c:6837 q931_hangup: Hangup other cref:14783 PRI Span: 1 q931.c:6594 __q931_hangup: ourstate Incoming Call Proceeding, peerstate Outgoing Call Proceeding, hold-state Idle PRI Span: 1 q931.c:5783 q931_disconnect: Call 14783 enters state 11 (Disconnect Request). Hold state: Idle PRI Span: 1 PRI Span: 1 DL-DATA request PRI Span: 1 Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to originator) PRI Span: 1 Message Type: DISCONNECT (69) PRI Span: 1 TEI=0 Transmitting N(S)=71, window is open V(A)=71 K=7 PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to originator) PRI Span: 1 Message Type: DISCONNECT (69) PRI Span: 1 [08 02 81 90] PRI Span: 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) PRI Span: 1
Re: [asterisk-users] cdr report
Well, the question is, what your secretary wants to do. Only see the CDRs or more? Realtime? One simple method would be to mail her the CSV-File, so she can open it with Excel or Calc (Open Office). Am 23.04.2013 16:35, schrieb aristidis tsitras: Hi. i am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on CPU, it has to be as low on CPU/RAM consumption as possible. any ideas? Sincerely yours, Aris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes
Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a Dial(DAHDI/g1/voicenumber,r). The call is bridged and everything is fine. dahdi show channels shows me, that channel 1 is used for the outcall. Then I try to hangup the outcall via dahdi destroy channel 1. Asterisk crahes immediatly. No message is logged (verbose is 10 and debug is 10). I get disconnected from the atserisk cli at this moment: vlr-3*CLI dahdi destroy channel 1 vlr-3*CLI Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups voxi@vlr-3:/tmp$ Is this a bug or is this my fault? Best regards -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes
Thanks! I do not have experience with bug reporting. Is that neccessary in that case? Where can I open a ticket for it (if neccessary)? Am 11.04.2013 12:23, schrieb Yves A.: Hi, I can reproduce your report (11.0.1, libpri 1.4.13, dahdi 2.6.1) and would say it is a bug... To remotely hang up a call use * **hangup request channel* where channel is the exact id of your channel as you would receive it via *core show channels* yves Am 11.04.2013 10:56, schrieb Thorsten Göllner: Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a Dial(DAHDI/g1/voicenumber,r). The call is bridged and everything is fine. dahdi show channels shows me, that channel 1 is used for the outcall. Then I try to hangup the outcall via dahdi destroy channel 1. Asterisk crahes immediatly. No message is logged (verbose is 10 and debug is 10). I get disconnected from the atserisk cli at this moment: vlr-3*CLI dahdi destroy channel 1 vlr-3*CLI Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups voxi@vlr-3:/tmp$ Is this a bug or is this my fault? Best regards -Thorsten- -- -- _ -- 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] WARNING[28151] from CLI
What hardware do you use? Do your have some E1 or T1 Ports? Maybe one or more of this ports is down. Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! -- _ -- 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] WARNING[28151] from CLI
You do use only span 1 and 6? So the other ports are not plugged? That is the cause for the warnings. I use a Sangoma E1-Card. The configure script gives me the option unused for any port. Maybe your configure script offers you the same option. Am 27.03.2013 11:54, schrieb Salaheddine Elharit: Hi i use 2 digium cards 1 card with 2 ports and the second card with 4 ports but actually i use just the span 1 and span 6 Asterisk 1.4-r110474M i use E1 ports zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 span=5,5,0,ccs,hdb3 # termtype: te bchan=125-139,141-155 dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone= us defaultzone= us etc/asterisk/zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=mycallerid immediate=no channel = 156-170 channel = 172-176 channel = 125-139 channel = 141-155 thanks and regards 2013/3/27 Yves A. yves...@gmx.de mailto:yves...@gmx.de Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... regards, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.2.1 / IAX / Problems with incoming calls
I am sure, that my log configuration is correct. NO messages will be logged other than the posted messages from iax debug. Am 08.03.2013 16:44, schrieb Rusty Newton: - Original Message - From: Thorsten Göllner t...@ovm-group.com I set verbose and debug to 100 but no(!) message was given. Read through https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information and read through the logger.conf sample file. Collect a full log with VERBOSE and DEBUG. Sanitize it as needed, and then link to a pastebin with a log excerpt covering from the very beginning of the attempted call to the end. You may also want to include your iax.conf configuration, sanitized too of course. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.2.1 / IAX / Problems with incoming calls
Thank - I will give it another try. Maybe the other endpoint is not compatible with this changes. Am 08.03.2013 17:36, schrieb Matthew Fredrickson: As I recall, there was an IAX2 protocol addition for newer versions of Asterisk a while ago due to a security issue - which can potentially trigger IAX2 interop issues if your config file for chan_iax2 is not setup properly. You can read more about it here: http://downloads.asterisk.org/pub/security/IAX2-security.pdf With regards to the CTOKEN addition. Hope that helps. Matthew Fredrickson Digium, Inc. On 3/8/13 8:38 AM, Thorsten Göllner wrote: Hi, I have upgraded vom Atserisk 1.6.1.20 to 11.2.1. Most things went fine. But 1 thing will not work: IAX. I used the same configuration but Asterisk will not answer the incoming IAX-Call. When enabling iax debugging I can see the following: [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:VERSION : 2 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING TYPEOFN : 0 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING TRANSIT : 1 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING NAME : 0049... [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:LANGUAGE : en [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:USERNAME : 02070992875 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:FORMAT : 8 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CAPABILITY : 65535 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:ADSICPE : 2 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:DATE TIME : 2013-03-07 16:14:38 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:Timestamp: 4ms SCall: 1 DCall: 05992 [77.240.54.23:4572] [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLTOKEN : 51 bytes [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:VERSION : 2 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING TYPEOFN : 0 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING TRANSIT : 1 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING NAME : 0049... [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:LANGUAGE : en [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:USERNAME : 02070992875 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:FORMAT : 8 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CAPABILITY : 65535 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:ADSICPE : 2 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:DATE TIME : 2013-03-07 16:14:38 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:Timestamp: 4ms SCall: 1 DCall: 05992 [77.240.54.23:4572] [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLTOKEN : 51 bytes [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c:Timestamp: 04006ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c:CAUSE CODE : 0 [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c: [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c:Timestamp: 04006ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c:CAUSE CODE : 0 [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c: [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23
[asterisk-users] CDR-Logging with leading 0 in src field clid and/or src
Hi, I am using Asterisk 11.2.1. I am logging CDRs to a mysql database (via odbc). The table contains the fields clid and src. Both fields are varchar(100). But alls entries are without the leading 0. For example 0211 for Germany-Düsseldorf. Where can I configure that behaviour, please? -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.2.1 / IAX / Problems with incoming calls
Hi, I have upgraded vom Atserisk 1.6.1.20 to 11.2.1. Most things went fine. But 1 thing will not work: IAX. I used the same configuration but Asterisk will not answer the incoming IAX-Call. When enabling iax debugging I can see the following: [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:VERSION : 2 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING TYPEOFN : 0 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING TRANSIT : 1 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING NAME: 0049... [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:LANGUAGE: en [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:USERNAME: 02070992875 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:FORMAT : 8 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CAPABILITY : 65535 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:ADSICPE : 2 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:DATE TIME : 2013-03-07 16:14:38 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:Timestamp: 4ms SCall: 1 DCall: 05992 [77.240.54.23:4572] [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLTOKEN : 51 bytes [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:VERSION : 2 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING TYPEOFN : 0 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING TRANSIT : 1 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING NAME: 0049... [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:LANGUAGE: en [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:USERNAME: 02070992875 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:FORMAT : 8 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CAPABILITY : 65535 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:ADSICPE : 2 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:DATE TIME : 2013-03-07 16:14:38 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:Timestamp: 4ms SCall: 1 DCall: 05992 [77.240.54.23:4572] [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLTOKEN : 51 bytes [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c:Timestamp: 04006ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c:CAUSE CODE : 0 [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c: [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c:Timestamp: 04006ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c:CAUSE CODE : 0 [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c: [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:VERSION : 2 [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:51] VERBOSE[3223]
Re: [asterisk-users] Error to install Asterisk
That should be ok. Try the following: open 2 shells. In the first one type watch df -h. In the second one you start the compilation. While compilation is running watch the first shell. The given command refreshes all 2 seconds the display and shows the used/free disk space. _Perhaps_ it will give you a hint, what mount point is running out of space. Am 06.03.2013 15:41, schrieb termo termosel: I have executed make in the same console where I had written mkdir /var/ext_tmp export TMPDIR=/var/ext_tmp make Is this way ok? Date: Wed, 6 Mar 2013 14:25:50 +0100 From: t...@ovm-group.com To: fermit...@hotmail.com CC: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Error to install Asterisk Did you execute the make command in the same environment so that make really uses the TMPDIR directory? (no su or other shell) Am 06.03.2013 13:37, schrieb termo termosel: Hi, the same error, I write your commands: mkdir /var/ext_tmp export TMPDIR=/var/ext_tmp make but the same error happens /usr/bin/ld: final link failed: No space left on device collect2: ld devolvió el estado de salida 1 make[2]: *** [asterisk] Error 1 make[1]: *** [main] Error 2 make[1]: se sale del directorio «/home/ubuntu/Downloads/asterisk-11.2.1» make: *** [_cleantest_all] Error 2 Jordi Date: Wed, 6 Mar 2013 13:29:24 +0100 From: t...@ovm-group.com mailto:t...@ovm-group.com To: fermit...@hotmail.com mailto:fermit...@hotmail.com CC: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Error to install Asterisk Try to set the tmp variable. In your case: mkdir /var/ext_tmp export TMPDIR=/var/ext_tmp make Am 06.03.2013 13:20, schrieb termo termosel: Hi, I read it but I don't find the solution. How Can I alocate more free space in tmp? Thanks, Jordi Date: Wed, 6 Mar 2013 13:12:34 +0100 From: t...@ovm-group.com mailto:t...@ovm-group.com To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com CC: fermit...@hotmail.com mailto:fermit...@hotmail.com Subject: Re: [asterisk-users] Error to install Asterisk Take a look here: http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device Am 06.03.2013 13:00, schrieb termo termosel: Hi, df -h output: root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# mailto:root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h S.ficherosTam. Usado Disp. % Uso Montado en /cow 14G 4,5G 8,7G 34% / udev 999M 4,0K 999M 1% /dev tmpfs 403M 860K 402M 1% /run /dev/sdb1 799M 693M 106M 87% /cdrom /dev/loop0668M 668M 0 100% /rofs tmpfs1006M 44K 1006M 1% /tmp none 5,0M 0 5,0M 0% /run/lock none 1006M 100K 1006M 1% /run/shm Jordi -- _ -- 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] Error to install Asterisk
Take a look here: http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device Am 06.03.2013 13:00, schrieb termo termosel: Hi, df -h output: root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h S.ficherosTam. Usado Disp. % Uso Montado en /cow 14G 4,5G 8,7G 34% / udev 999M 4,0K 999M 1% /dev tmpfs 403M 860K 402M 1% /run /dev/sdb1 799M 693M 106M 87% /cdrom /dev/loop0668M 668M 0 100% /rofs tmpfs1006M 44K 1006M 1% /tmp none 5,0M 0 5,0M 0% /run/lock none 1006M 100K 1006M 1% /run/shm Jordi -- _ -- 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] Error to install Asterisk
Try to set the tmp variable. In your case: mkdir /var/ext_tmp export TMPDIR=/var/ext_tmp make Am 06.03.2013 13:20, schrieb termo termosel: Hi, I read it but I don't find the solution. How Can I alocate more free space in tmp? Thanks, Jordi Date: Wed, 6 Mar 2013 13:12:34 +0100 From: t...@ovm-group.com To: asterisk-users@lists.digium.com CC: fermit...@hotmail.com Subject: Re: [asterisk-users] Error to install Asterisk Take a look here: http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device Am 06.03.2013 13:00, schrieb termo termosel: Hi, df -h output: root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# mailto:root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h S.ficherosTam. Usado Disp. % Uso Montado en /cow 14G 4,5G 8,7G 34% / udev 999M 4,0K 999M 1% /dev tmpfs 403M 860K 402M 1% /run /dev/sdb1 799M 693M 106M 87% /cdrom /dev/loop0668M 668M 0 100% /rofs tmpfs1006M 44K 1006M 1% /tmp none 5,0M 0 5,0M 0% /run/lock none 1006M 100K 1006M 1% /run/shm Jordi -- _ -- 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] Error to install Asterisk
Did you execute the make command in the same environment so that make really uses the TMPDIR directory? (no su or other shell) Am 06.03.2013 13:37, schrieb termo termosel: Hi, the same error, I write your commands: mkdir /var/ext_tmp export TMPDIR=/var/ext_tmp make but the same error happens /usr/bin/ld: final link failed: No space left on device collect2: ld devolvió el estado de salida 1 make[2]: *** [asterisk] Error 1 make[1]: *** [main] Error 2 make[1]: se sale del directorio «/home/ubuntu/Downloads/asterisk-11.2.1» make: *** [_cleantest_all] Error 2 Jordi Date: Wed, 6 Mar 2013 13:29:24 +0100 From: t...@ovm-group.com To: fermit...@hotmail.com CC: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Error to install Asterisk Try to set the tmp variable. In your case: mkdir /var/ext_tmp export TMPDIR=/var/ext_tmp make Am 06.03.2013 13:20, schrieb termo termosel: Hi, I read it but I don't find the solution. How Can I alocate more free space in tmp? Thanks, Jordi Date: Wed, 6 Mar 2013 13:12:34 +0100 From: t...@ovm-group.com mailto:t...@ovm-group.com To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com CC: fermit...@hotmail.com mailto:fermit...@hotmail.com Subject: Re: [asterisk-users] Error to install Asterisk Take a look here: http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device Am 06.03.2013 13:00, schrieb termo termosel: Hi, df -h output: root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# mailto:root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h S.ficherosTam. Usado Disp. % Uso Montado en /cow 14G 4,5G 8,7G 34% / udev 999M 4,0K 999M 1% /dev tmpfs 403M 860K 402M 1% /run /dev/sdb1 799M 693M 106M 87% /cdrom /dev/loop0668M 668M 0 100% /rofs tmpfs1006M 44K 1006M 1% /tmp none 5,0M 0 5,0M 0% /run/lock none 1006M 100K 1006M 1% /run/shm Jordi -- _ -- 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] crossed channels
Ist one channel significant louder than the other? Maybe it is some sort of crosstalking. Take a look here: http://es.wikipedia.org/wiki/Diafon%C3%ADa Am 19.02.2013 16:25, schrieb Juan Carlos Agudelo: El 19/02/13 03:59, Thorsten Göllner escribió: What exactly do you mean by crossing channels? Mixed audio? Can callers hear each other? Am 19.02.2013 02:07, schrieb Juan Carlos Agudelo: Hi, I have installed Asterisk 1.6.2.17-rc2 and I have a strange behavior, because sometimes they are crossing channels, thus producing unwanted calls connections...Any suggestions? -- _ -- 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] crossed channels
What exactly do you mean by crossing channels? Mixed audio? Can callers hear each other? Am 19.02.2013 02:07, schrieb Juan Carlos Agudelo: Hi, I have installed Asterisk 1.6.2.17-rc2 and I have a strange behavior, because sometimes they are crossing channels, thus producing unwanted calls connections...Any suggestions? -- _ -- 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] How to read/set ulimit for non-root asterisk process ?
Hi Olivier, you have to edit /etc/security/limits.conf. Take a look at man limits.conf. Some users also modify the Asterisk-Start-Script. You can insert an ulimit -n 8192 in the Start-Case. Best regard -Thorsten- Am 15.02.2013 18:48, schrieb Olivier: 2013/2/15 Olivier oza_4...@yahoo.fr mailto:oza_4...@yahoo.fr Hello, On a production system, I'm seeing this: [Feb 13 16:47:00] WARNING[14742] res_agi.c: Unable to create toast pipe: Too many open files [Feb 13 16:47:00] WARNING[9283] acl.c: Cannot create socket [Feb 13 16:47:00] WARNING[9283] rtp.c: Unable to allocate RTCP socket: Too many open files [Feb 13 16:47:00] WARNING[14732] acl.c: Cannot create socket [Feb 13 16:47:00] WARNING[14732] channel.c: Channel allocation failed: Can't create alert pipe! Try increasing max file descriptors with ulimit -n [Feb 13 16:47:00] WARNING[14732] chan_sip.c: Unable to allocate AST channel structure for SIP channel [Feb 13 16:47:00] WARNING[14732] app_dial.c: Unable to create channel of type 'SIP' (cause 0 - Unknown) [Feb 13 16:47:00] ERROR[14732] rtp.c: Unable to allocate socket: Too many open files Typing ulimit -a, shows : # ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 16382 max locked memory (kbytes, -l) 64 max memory size (kbytes, -m) unlimited open files (-n) 1024 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 8192 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited file locks (-x) unlimited So it seems that increasing this open files limit from 1024 to 2048 could work around the above issue. Strangely, I can't find much online doc on ulimit and its usage. My main source is http://ss64.com/bash/ulimit.html and I also found this http://lists.digium.com/pipermail/asterisk-dev/2006-October/024091.html where I could read / And what does 'ulimit -n' say for your Asterisk process?/ 1. How can I specificially read ulimit -n for asterisk, for instance when asterisk is run by an asterisk user which has no login or shell ? Finally, it seems this command is enough : su asterisk --shell /bin/sh --command ulimit -n 2. Is there an easy and safe way to increase the number of files opened by asterisk ? Replace the question above by this one Is there an easy and safe way to artificially increase the number of files opened by asterisk ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan / check / tool
Hi, I am wondering, if there is any tool available, which performs a check for suspicious entries in the dialplan. For example a non existing AGI-Script or a double assigned extension ike that: [context] exten = *100*,1,AGI(test_app.pl) ... exten = 190,1,Answer() ... exten = *100*,1,AGI(never_reached.pl) ... A normal dialplan reload command would echo no warning or something similair. Best regards -Thorsten- -- _ -- 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] ConfBridge performance problem...?
Hi again, I did a try on my asterisk 11.2.1 compiled on Ubuntu 12.04 (64 bit) with a simple Pentium 4 CPU (Intel(R) Pentium(R) D CPU 2.80GHz). I connected 5 SIP-Users with a ConfBridge. This is my picture: Please give a a hint where I can change "you parameters" like denoise etc. So I will try to change these settings on my box also. I have no experience with Confbridge. -Thorsten- Am 08.02.2013 19:34, schrieb Hristo Trendev: Hi, the quad-core server is a dedicated asterisk server. I duplicated the tests on a virtual server (running on another physical server) only to rule out the possibility of hardware problem with the first sever. Hristo On Fri, Feb 8, 2013 at 11:41 AM, Thorsten Gllner t...@ovm-group.com wrote: Hi, perhaps it is a problem with your Host-Guest-Setup? Did you try the Asterisk-Setup on a dedicated server without virtualization? -Thorsten- Am 07.02.2013 11:42, schrieb Hristo Trendev: Hi Thorsten, Thanks for your reply. I did check core show translations, but the followinghttp://lists.digium.com/pipermail/asterisk-users/2012-November/276132.html suggests that the values displayed are no longer representing the computation cost only. However to answer your question: G722 to SLIN16 cost is 9000, reverse direction is 6000 ALAW to SLN16 cost is 17000, reverse direction is 14500 G722 to SLN cost is 9600, reverse direction is 8250 ALAW to SLN cost is 9000, reverse direction is 6000 With regards to the CPU usage per core - inside the VM, where only one core is available, the CPU was close to 100% when the problem started to apear, on the physical server with 4 cores, the cores were evenly loaded at about 30-40%. A single call into the conference consumed between 10-20% depending on whether I have denoise enabled or not. There is no dahdi board installed, I only use the dahdi module for conference timer (note that the problem is also present with thetimerfd timing module). BR, Hristo On Wed, Feb 6, 2013 at 1:57 PM, Thorsten Gllner t...@ovm-group.com wrote: Did you check asterisk -rx "core show translation recalc 10" Am 06.02.2013 13:56, schrieb Thorsten Gllner: Sorry - I just read you alsways checked the cpu usage. Are all cores at 100%? Is it the atserisk process which consumes it all? Am 06.02.2013 13:54, schrieb Thorsten Gllner: Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command "dahdi_test"? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the asterisk-11
Re: [asterisk-users] ConfBridge performance problem...?
Hi, perhaps it is a problem with your Host-Guest-Setup? Did you try the Asterisk-Setup on a dedicated server without virtualization? -Thorsten- Am 07.02.2013 11:42, schrieb Hristo Trendev: Hi Thorsten, Thanks for your reply. I did check core show translations, but the following http://lists.digium.com/pipermail/asterisk-users/2012-November/276132.html suggests that the values displayed are no longer representing the computation cost only. However to answer your question: G722 to SLIN16 cost is 9000, reverse direction is 6000 ALAW to SLN16 cost is 17000, reverse direction is 14500 G722 to SLN cost is 9600, reverse direction is 8250 ALAW to SLN cost is 9000, reverse direction is 6000 With regards to the CPU usage per core - inside the VM, where only one core is available, the CPU was close to 100% when the problem started to apear, on the physical server with 4 cores, the cores were evenly loaded at about 30-40%. A single call into the conference consumed between 10-20% depending on whether I have denoise enabled or not. There is no dahdi board installed, I only use the dahdi module for conference timer (note that the problem is also present with the timerfd timing module). BR, Hristo On Wed, Feb 6, 2013 at 1:57 PM, Thorsten Göllner t...@ovm-group.com mailto:t...@ovm-group.com wrote: Did you check asterisk -rx core show translation recalc 10 Am 06.02.2013 13:56, schrieb Thorsten Göllner: Sorry - I just read you alsways checked the cpu usage. Are all cores at 100%? Is it the atserisk process which consumes it all? Am 06.02.2013 13:54, schrieb Thorsten Göllner: Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command dahdi_test? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants that they've seen in a conference. I was never able to get more than 8 participants (mixed G722 and G711a) on a conference (actually that's per server limit) with almost all settings on default, except for dsp_drop_silence and denoise which are enabled. I tested on Debian squeeze, 64-bit, quad-core Xeon server @2.4GHz and also on another virtual server with similar processor (just one core available to the VM). While this is not the latest and greatest CPU, I would certainly expect it to handle more than 8 calls. To be honest, I was in fact able to get it working for up to 20 participants (most with G711), when I switched from res_timing_timerfd to res_timing_dahdi and turned off denoise, but that's still not normal I believe, especially with most participants on mute and with dps_drop_silence enabled and nothing else running on the server. The problem itself is, that once I get over the critical number of participants, the voice starts to break up and it's impossible to understand the person who's talking. This is certainly not bandwidth related because all tests were made on the LAN and besides I could see that the CPU was sometime close to 100%. Did someone observe something similar? BTW, once the first participant enters the conference I start seeing probably over 50 messages per second saying: bridging.c:757 bridge_channel_join_multithreaded: Going into a multithreaded waitfor for bridge channel 0x292d708 of bridge 0x28f3658 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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] ConfBridge performance problem...?
Hi, perhaps it is a problem with your Host-Guest-Setup? Did you try the Asterisk-Setup on a dedicated server without virtualization? -Thorsten- Am 07.02.2013 11:42, schrieb Hristo Trendev: Hi Thorsten, Thanks for your reply. I did check core show translations, but the following http://lists.digium.com/pipermail/asterisk-users/2012-November/276132.html suggests that the values displayed are no longer representing the computation cost only. However to answer your question: G722 to SLIN16 cost is 9000, reverse direction is 6000 ALAW to SLN16 cost is 17000, reverse direction is 14500 G722 to SLN cost is 9600, reverse direction is 8250 ALAW to SLN cost is 9000, reverse direction is 6000 With regards to the CPU usage per core - inside the VM, where only one core is available, the CPU was close to 100% when the problem started to apear, on the physical server with 4 cores, the cores were evenly loaded at about 30-40%. A single call into the conference consumed between 10-20% depending on whether I have denoise enabled or not. There is no dahdi board installed, I only use the dahdi module for conference timer (note that the problem is also present with the timerfd timing module). BR, Hristo On Wed, Feb 6, 2013 at 1:57 PM, Thorsten Göllner t...@ovm-group.com mailto:t...@ovm-group.com wrote: Did you check asterisk -rx core show translation recalc 10 Am 06.02.2013 13:56, schrieb Thorsten Göllner: Sorry - I just read you alsways checked the cpu usage. Are all cores at 100%? Is it the atserisk process which consumes it all? Am 06.02.2013 13:54, schrieb Thorsten Göllner: Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command dahdi_test? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants that they've seen in a conference. I was never able to get more than 8 participants (mixed G722 and G711a) on a conference (actually that's per server limit) with almost all settings on default, except for dsp_drop_silence and denoise which are enabled. I tested on Debian squeeze, 64-bit, quad-core Xeon server @2.4GHz and also on another virtual server with similar processor (just one core available to the VM). While this is not the latest and greatest CPU, I would certainly expect it to handle more than 8 calls. To be honest, I was in fact able to get it working for up to 20 participants (most with G711), when I switched from res_timing_timerfd to res_timing_dahdi and turned off denoise, but that's still not normal I believe, especially with most participants on mute and with dps_drop_silence enabled and nothing else running on the server. The problem itself is, that once I get over the critical number of participants, the voice starts to break up and it's impossible to understand the person who's talking. This is certainly not bandwidth related because all tests were made on the LAN and besides I could see that the CPU was sometime close to 100%. Did someone observe something similar? BTW, once the first participant enters the conference I start seeing probably over 50 messages per second saying: bridging.c:757 bridge_channel_join_multithreaded: Going into a multithreaded waitfor for bridge channel 0x292d708 of bridge 0x28f3658 -- _ -- 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] SayDigits
Am 08.02.2013 13:11, schrieb Doug Lytle: Is there a way to slow down or speed up the speed at which SayDigits core show application saydigits [Synopsis] Say Digits. [Description] This application will play the sounds that correspond to the digits of the given number. This will use the language that is currently set for the channel. [Syntax] SayDigits(digits) [Arguments] Not available So, I'd have to say no. Doug You should write a little AGI-Script instead. -- _ -- 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] ConfBridge performance problem...?
Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command dahdi_test? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants that they've seen in a conference. I was never able to get more than 8 participants (mixed G722 and G711a) on a conference (actually that's per server limit) with almost all settings on default, except for dsp_drop_silence and denoise which are enabled. I tested on Debian squeeze, 64-bit, quad-core Xeon server @2.4GHz and also on another virtual server with similar processor (just one core available to the VM). While this is not the latest and greatest CPU, I would certainly expect it to handle more than 8 calls. To be honest, I was in fact able to get it working for up to 20 participants (most with G711), when I switched from res_timing_timerfd to res_timing_dahdi and turned off denoise, but that's still not normal I believe, especially with most participants on mute and with dps_drop_silence enabled and nothing else running on the server. The problem itself is, that once I get over the critical number of participants, the voice starts to break up and it's impossible to understand the person who's talking. This is certainly not bandwidth related because all tests were made on the LAN and besides I could see that the CPU was sometime close to 100%. Did someone observe something similar? BTW, once the first participant enters the conference I start seeing probably over 50 messages per second saying: bridging.c:757 bridge_channel_join_multithreaded: Going into a multithreaded waitfor for bridge channel 0x292d708 of bridge 0x28f3658 Best, -- _ -- 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] ConfBridge performance problem...?
Sorry - I just read you alsways checked the cpu usage. Are all cores at 100%? Is it the atserisk process which consumes it all? Am 06.02.2013 13:54, schrieb Thorsten Göllner: Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command dahdi_test? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants that they've seen in a conference. I was never able to get more than 8 participants (mixed G722 and G711a) on a conference (actually that's per server limit) with almost all settings on default, except for dsp_drop_silence and denoise which are enabled. I tested on Debian squeeze, 64-bit, quad-core Xeon server @2.4GHz and also on another virtual server with similar processor (just one core available to the VM). While this is not the latest and greatest CPU, I would certainly expect it to handle more than 8 calls. To be honest, I was in fact able to get it working for up to 20 participants (most with G711), when I switched from res_timing_timerfd to res_timing_dahdi and turned off denoise, but that's still not normal I believe, especially with most participants on mute and with dps_drop_silence enabled and nothing else running on the server. The problem itself is, that once I get over the critical number of participants, the voice starts to break up and it's impossible to understand the person who's talking. This is certainly not bandwidth related because all tests were made on the LAN and besides I could see that the CPU was sometime close to 100%. Did someone observe something similar? BTW, once the first participant enters the conference I start seeing probably over 50 messages per second saying: bridging.c:757 bridge_channel_join_multithreaded: Going into a multithreaded waitfor for bridge channel 0x292d708 of bridge 0x28f3658 -- _ -- 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] ConfBridge performance problem...?
Did you check asterisk -rx core show translation recalc 10 Am 06.02.2013 13:56, schrieb Thorsten Göllner: Sorry - I just read you alsways checked the cpu usage. Are all cores at 100%? Is it the atserisk process which consumes it all? Am 06.02.2013 13:54, schrieb Thorsten Göllner: Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command dahdi_test? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants that they've seen in a conference. I was never able to get more than 8 participants (mixed G722 and G711a) on a conference (actually that's per server limit) with almost all settings on default, except for dsp_drop_silence and denoise which are enabled. I tested on Debian squeeze, 64-bit, quad-core Xeon server @2.4GHz and also on another virtual server with similar processor (just one core available to the VM). While this is not the latest and greatest CPU, I would certainly expect it to handle more than 8 calls. To be honest, I was in fact able to get it working for up to 20 participants (most with G711), when I switched from res_timing_timerfd to res_timing_dahdi and turned off denoise, but that's still not normal I believe, especially with most participants on mute and with dps_drop_silence enabled and nothing else running on the server. The problem itself is, that once I get over the critical number of participants, the voice starts to break up and it's impossible to understand the person who's talking. This is certainly not bandwidth related because all tests were made on the LAN and besides I could see that the CPU was sometime close to 100%. Did someone observe something similar? BTW, once the first participant enters the conference I start seeing probably over 50 messages per second saying: bridging.c:757 bridge_channel_join_multithreaded: Going into a multithreaded waitfor for bridge channel 0x292d708 of bridge 0x28f3658 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set(CALLERID(num-pres)=prohib) / AGI version
Hi, on this site http://www.voip-info.org/wiki/view/Asterisk+func+callerid you can read, that since Atserisk 1.8 the command (in dialplan) to hide the caller id is: Set(CALLERID(num-pres)=prohib) I tried to implement it into my AGI-Script, but with no success. Can please anyone give me a hint, what is wrong with it: Set CALLERID(num-pres) prohib or Set CALLERID(num-pres)=prohib Both commands lead into: 510 Invalid or unknown command Besr regards -Thorsten- -- _ -- 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] Set(CALLERID(num-pres)=prohib) / AGI version
Am 06.02.2013 16:02, schrieb Steve Edwards:
On Wed, 6 Feb 2013, Thorsten Göllner wrote:
I tried to implement it into my AGI-Script, but with no success. Can
please anyone give me a hint, what is wrong with it:
Set CALLERID(num-pres) prohib
or
Set CALLERID(num-pres)=prohib
Both commands lead into:
510 Invalid or unknown command
I'm just a 1.2 Luddite, but...
Who's library/framework are you using?
Neither of the commands you show above are valid AGI commands.
Curiously, I've never tried to set caller ID (or its options) in an
AGI, I've only set channel variables that ended up setting CID in the
dialplan.
If you were reading the variables, the command would look like:
'get full variable ${CALLERID(num-pres)}'
Maybe you could try something like:
'set variable CALLERID(num-pres) prohib'
(I don't see a 'set full variable' AGI command.)
How about a console log with verbose and debug cranked up and with AGI
debug enabled?
Thanks. But I found the right syntax now:
Exec Set CALLERID(num-pres)=prohib
This AGI-Command leads into 200 OK and I can verify, that outgoing
calls (SIP and DAHDI) are anonymous.
-Thorsten-
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[asterisk-users] CEL / CELGenUserEvent via AGI / no error and no cel entry
Hi, I am using Asterisk 11.2.0. Channel Event Logging (CEL) ist activated and running. CEL entries are logged into an mysql database. So far so good. I want to do some extra cel logging and try the following via an AGI-Script: EXEC CELGenUserEvent test In the asterisk logfile I can see the following: -- AGI Script Executing Application: (CELGenUserEvent) Options: (test) (no errors or warnings) But there is no cel entry in my database. What is going wrong here, please? Best regards, -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11 / Missing Application SetCallerPres
Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI == Using SIP RTP CoS mark 5 -- Executing [100@sip:1] AGI(SIP/userid-001e, test.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script Executing Application: (SetCallerPres) Options: (prohib_not_screened) [Jan 24 15:20:04] WARNING[15507][C-0030]: res_agi.c:2527 handle_exec: Could not find application (SetCallerPres) Why is the application not found, please? I think it should exist: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_SetCallerPres Best regards, -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 / Missing Application SetCallerPres
Thanks! It is not activated. Also I found a comment there: Support Level: deprecated, Replaced by: func_callerid So I use this instead. Am 24.01.2013 15:33, schrieb Danny Nicholas: Simplest question first. Does it show up in core show applications or core show application SetCallerPres? If not, do a make menuselect and see if something broke in the ability to make the application. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten Göllner Sent: Thursday, January 24, 2013 8:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 11 / Missing Application SetCallerPres Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI == Using SIP RTP CoS mark 5 -- Executing [100@sip:1] AGI(SIP/userid-001e, test.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script Executing Application: (SetCallerPres) Options: (prohib_not_screened) [Jan 24 15:20:04] WARNING[15507][C-0030]: res_agi.c:2527 handle_exec: Could not find application (SetCallerPres) Why is the application not found, please? I think it should exist: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_SetCaller Pres Best regards, -Thorsten- -- _ -- 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] ODBC Connection Problem
First of all test your odbc-connection via console: isql telco-ops dba c3podb@2012 -v You should see a Connected!-Message. Do you? Second: yes I also had problems setting up odbc. The main problem/error for me was, that documentation is sometimes confusing. Here is my config. Please notice the [section] - namings: /etc/odbcinst.ini [MySQL] Description = MySQL ODBCMyODBC Driver Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so FileUsage = 1 /etc/odbc.ini [MySQL-asterisk] Description = MySQL ODBC Driver Driver = MySQL Socket = /var/run/mysqld/mysqld.sock Server = localhost User = my_username Password = my_password Database = my_database Option = 3 Port = Charset = utf8 /etc/asterisk/res_odbc.conf [mysql] enabled = yes dsn = MySQL-asterisk username = my_username password = my_password pre-connect = yes /etc/asterisk/cdr_odbc.conf [global] dsn=mysql loguniqueid=yes dispositionstring=yes table=cdr /etc/asterisk/cel_odbc.conf [first] connection=mysql table=cel| | Additionally you will need some configurations for you realtime-config. This config above is only for cdr- and cel-logging via odbc. -Thorsten- Am 10.12.2012 12:23, schrieb Chandrakant Solanki: /etc/odbc.ini [telco-ops] Description = Asterisk realtime and other FUNC_ODBC access Driver = MySQL Server = 172.18.100.18 Socket = /var/lib/mysql/data3306/mysql.sock User= dba Password= c3podb@2012 Database= mytelcoexample Port= 3306 Option = 3 On Mon, Dec 10, 2012 at 4:34 PM, Thorsten Göllner t...@ovm-group.com mailto:t...@ovm-group.com wrote: Am 10.12.2012 06:37, schrieb Chandrakant Solanki: Hi All, OS : CentOS 5 64bit OS Machine Asterisk: 1.8.13.0 ODBC Packages: unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 unixODBC-devel-2.2.11-7.1 res_odbc.conf [telco-ops] enabled = yes dsn = telco-ops username = dba password = c3podb@2012 pre-connect = yes sanitysql = select 1 idlecheck = 15 ;isolation = repeatable_read pooling = yes limit = 3600 connect_timeout = 10 negative_connection_cache = 30 Above is my installation package and configuration file (res_odbc.conf), when I try to execute odbc show all it always gives below output. *CLI odbc show all ODBC DSN Settings - Name: telco-ops DSN:telco-ops Last connection attempt: 1970-01-01 00:00:00 Pooled: Yes Limit: 3600 Connections in use: 1 - Connection 1: connected When Insert/Update/Select query will be executed, it can't update last connection attempt field. In result, ODBC stuck after few minutes, and in this case I also need to restart asterisk, because I can't type any command, it can't give any command's output. Also updated asterisk with 10.9.0, but same result. Please show us /etc/odbc.ini too. -- _ -- 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] ODBC Connection Problem
Am 10.12.2012 06:37, schrieb Chandrakant Solanki: Hi All, OS : CentOS 5 64bit OS Machine Asterisk: 1.8.13.0 ODBC Packages: unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 unixODBC-devel-2.2.11-7.1 res_odbc.conf [telco-ops] enabled = yes dsn = telco-ops username = dba password = c3podb@2012 pre-connect = yes sanitysql = select 1 idlecheck = 15 ;isolation = repeatable_read pooling = yes limit = 3600 connect_timeout = 10 negative_connection_cache = 30 Above is my installation package and configuration file (res_odbc.conf), when I try to execute odbc show all it always gives below output. *CLI odbc show all ODBC DSN Settings - Name: telco-ops DSN:telco-ops Last connection attempt: 1970-01-01 00:00:00 Pooled: Yes Limit: 3600 Connections in use: 1 - Connection 1: connected When Insert/Update/Select query will be executed, it can't update last connection attempt field. In result, ODBC stuck after few minutes, and in this case I also need to restart asterisk, because I can't type any command, it can't give any command's output. Also updated asterisk with 10.9.0, but same result. Please show us /etc/odbc.ini too. -- _ -- 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] PRI can receive calls but cannot dial out
Hi! 1) How long does the outdial take? Does the Dial-Command return immediatly? 2) Maybe dial-out is blocked by your carrier? Did you try to open a trouble ticket there? 3) What number do you try to call? Did you try some different number? Alway the same problem? You receive ISDN-Cause-Code 18. Not sure though, but I would open a troubke ticket at your carrier. -Thorsten- Am 05.12.2012 08:48, schrieb Vieri: Hi, I'm trying to call out from a SIP extension to an outbound destination via a PRI E1 (Digium B410P). Please take a look at the PRI debug below. # cat /etc/dahdi/system.conf # Digium Wildcard TDM400P REV I (WCTDM/4) fxsks=1 echocanceller=oslec,1 fxsks=2 echocanceller=oslec,2 fxsks=3 echocanceller=oslec,3 fxsks=4 echocanceller=oslec,4 # Digium Wildcard TDM2400P (WCTDM/0) fxsks=5 echocanceller=oslec,5 fxsks=6 echocanceller=oslec,6 fxsks=7 echocanceller=oslec,7 fxsks=8 echocanceller=oslec,8 fxsks=9 echocanceller=oslec,9 fxsks=10 echocanceller=oslec,10 fxsks=11 echocanceller=oslec,11 fxsks=12 echocanceller=oslec,12 # Digium Wildcard B410P (B4/0/1) span=3,1,0,CCS,AMI bchan=29-30 hardhdlc=31 echocanceller=oslec,29-30 # Digium Wildcard B410P (B4/0/2) span=4,2,0,CCS,AMI bchan=32-33 hardhdlc=34 echocanceller=oslec,32-33 # Digium Wildcard B410P (B4/0/3) span=5,3,0,CCS,AMI bchan=35-36 hardhdlc=37 echocanceller=oslec,35-36 # Digium Wildcard B410P (B4/0/4) span=6,4,0,CCS,AMI bchan=38-39 hardhdlc=40 echocanceller=oslec,38-39 # lsmod | grep wcb4xxp wcb4xxp66250 12 dahdi 169899 65 dahdi_echocan_oslec,wcb4xxp,wctdm24xxp,dahdi_voicebus,wctdm # cat chan_dahdi.conf [trunkgroups] [channels] transfer = yes usecallerid = yes cidsignalling = dtmf callwaiting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes canpark = yes cancallforward = yes callreturn = yes callprogress = no overlapdial = yes echocancel = yes facilityenable = yes immediate = no busydetect = no ; Digium Wildcard TDM400P REV I (WCTDM/4) signalling = fxs_ks txgain = 1.0 rxgain = 14.0 group = 3 context = incoming-dahdi-3 faxdetect = incoming channel = 1,2,3,4 ; Digium Wildcard TDM2400P (WCTDM/0) group = 4 context = incoming-dahdi-4 faxdetect = incoming channel = 5,6,7,8,9,10,11,12 ; Digium Wildcard B410P (B4/0/1) signalling = bri_cpe switchtype = euroisdn rxgain = 2.0 group = 2 context = incoming-dahdi-2 faxdetect = incoming channel = 29-30 ; Digium Wildcard B410P (B4/0/2) channel = 32-33 ; Digium Wildcard B410P (B4/0/3) channel = 35-36 ; Digium Wildcard B410P (B4/0/4) channel = 38-39 --- # asterisk -rx dahdi show status Description Alarms IRQbpviol CRCFra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) Wildcard TDM2400POK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 1RED 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 2OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 3OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 4OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) Note that I have 3 cables connected and 1 port is free (RED). --- in AEL dialplan, I run: Dial(DAHDI/g2/XX); in the *CLI I see the following: -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/g2/XX -- Span 4: Channel 0/1 got hangup, cause 18 -- Hungup 'DAHDI/i4/XX-7' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/4053-0089' status is 'CHANUNAVAIL' If I enable PRI debug: -- Executing [@company:1] Dial(SIP/4053-0001, DAHDI/g2/XX) in new stack PRI Span: 4 -- Making new call for cref 32772 -- Requested transfer capability: 0x00 - SPEECH PRI Span: 4 PRI Span: 4 DL-DATA request PRI Span: 4 Protocol Discriminator: Q.931 (8) len=32 PRI Span: 4 TEI=0 Call Ref: len= 1 (reference 4/0x4) (Sent from originator) PRI Span: 4 Message Type: SETUP (5) PRI Span: 4 TEI=0 Transmitting N(S)=6, window is open V(A)=6 K=1 PRI Span: 4 PRI Span: 4 Protocol Discriminator: Q.931 (8) len=32 PRI Span: 4 TEI=0 Call Ref: len= 1 (reference 4/0x4) (Sent from originator) PRI Span: 4 Message Type: SETUP (5) PRI Span: 4 [04 03 80 90 a3] PRI Span: 4 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 4 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 4 User information layer 1: A-Law (35) PRI Span: 4 [18 01 81] PRI Span: 4 Channel ID (len= 3) [ Ext: 1 IntID: Implicit BRI
Re: [asterisk-users] PRI got event HDLC Abort
Maybe you should give irqbalance a try: https://irqbalance.org/ Maybe you also can assign irq 30 to a specific cpu (core): https://cs.uwaterloo.ca/~brecht/servers/apic/SMP-affinity.txt Am 06.11.2012 04:04, schrieb Edwin Lam: On 11/5/12 11:59 AM, Vincent Swart wrote: You're HDLC error is evident of timing slips. Use cat /proc/dahdi/1 or 2 or 3 aha.. it does have timing slips... Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) B8ZS/ESF ClockSource CRC4 error count: 6864 E-bit error count: 27603 IRQ misses: 1 Timing slips: 1459 1 TE4/0/1/1 Clear (In use) (EC: VPMOCT128 - INACTIVE) 2 TE4/0/1/2 Clear (In use) (EC: VPMOCT128 - INACTIVE) 3 TE4/0/1/3 Clear (In use) (EC: VPMOCT128 - INACTIVE) 4 TE4/0/1/4 Clear (In use) (EC: VPMOCT128 - INACTIVE) 5 TE4/0/1/5 Clear (In use) (EC: VPMOCT128 - INACTIVE) 6 TE4/0/1/6 Clear (In use) (EC: VPMOCT128 - INACTIVE) 7 TE4/0/1/7 Clear (In use) (EC: VPMOCT128 - INACTIVE) 8 TE4/0/1/8 Clear (In use) (EC: VPMOCT128 - INACTIVE) 9 TE4/0/1/9 Clear (In use) (EC: VPMOCT128 - INACTIVE) 10 TE4/0/1/10 Clear (In use) (EC: VPMOCT128 - INACTIVE) 11 TE4/0/1/11 Clear (In use) (EC: VPMOCT128 - INACTIVE) 12 TE4/0/1/12 Clear (In use) (EC: VPMOCT128 - INACTIVE) 13 TE4/0/1/13 Clear (In use) (EC: VPMOCT128 - INACTIVE) 14 TE4/0/1/14 Clear (In use) (EC: VPMOCT128 - INACTIVE) 15 TE4/0/1/15 Clear (In use) (EC: VPMOCT128 - INACTIVE) 16 TE4/0/1/16 Clear (In use) (EC: VPMOCT128 - INACTIVE) 17 TE4/0/1/17 Clear (In use) (EC: VPMOCT128 - INACTIVE) 18 TE4/0/1/18 Clear (In use) (EC: VPMOCT128 - INACTIVE) 19 TE4/0/1/19 Clear (In use) (EC: VPMOCT128 - INACTIVE) 20 TE4/0/1/20 Clear (In use) (EC: VPMOCT128 - INACTIVE) 21 TE4/0/1/21 Clear (In use) (EC: VPMOCT128 - INACTIVE) 22 TE4/0/1/22 Clear (In use) (EC: VPMOCT128 - INACTIVE) 23 TE4/0/1/23 Clear (In use) (EC: VPMOCT128 - INACTIVE) 24 TE4/0/1/24 HDLCFCS (In use) (EC: VPMOCT128 - INACTIVE) Also cat /proc /interrupts however i don't see any interrupt conflicts.. maybe i should try manually assign CPU affinity on that IRQ? CPU0 CPU1 CPU2 CPU3 0: 2108 0 0 0 IO-APIC-edge timer 1: 0 0 0 0 IO-APIC-edge i8042 8: 1 0 0 0 IO-APIC-edge rtc0 9: 0 0 0 0 IO-APIC-fasteoi acpi 14: 89 0 0 0 IO-APIC-edge ata_piix 15: 0 0 0 0 IO-APIC-edge ata_piix 16: 608555 0 0 0 IO-APIC-fasteoi megasas 17: 51 0 0 0 IO-APIC-fasteoi ehci_hcd:usb2, uhci_hcd:usb3, uhci_hcd:usb5 18: 0 0 0 0 IO-APIC-fasteoi uhci_hcd:usb4, uhci_hcd:usb6 19: 0 0 0 0 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb7 21: 0 0 0 0 IO-APIC-fasteoi ata_piix 30: 604673256 0 0 0 IO-APIC-fasteoi wct4xxp 54: 3 0 0 0 PCI-MSI-edge ioat-msix 55: 3 0 0 0 PCI-MSI-edge ioat-msix 56: 3 0 0 0 PCI-MSI-edge ioat-msix 57: 3 0 0 0 PCI-MSI-edge ioat-msix 58: 3 0 0 0 PCI-MSI-edge ioat-msix 59: 3 0 0 0 PCI-MSI-edge ioat-msix 60: 3 0 0 0 PCI-MSI-edge ioat-msix 61: 3 0 0 0 PCI-MSI-edge ioat-msix 62: 772684 0 0 0 PCI-MSI-edge eth0-0 63: 368866 0 0 0 PCI-MSI-edge eth0-1 64: 105367 0 0 0 PCI-MSI-edge eth0-2 65: 0 0 0 0 PCI-MSI-edge eth0-3 66: 0 0 0 0 PCI-MSI-edge eth0-4 71: 22558707 0 0 0 PCI-MSI-edge eth1-0 72: 15994275 0 0 0 PCI-MSI-edge eth1-1 73: 24318397 0 0 0 PCI-MSI-edge eth1-2 74: 12812423 0 0 0 PCI-MSI-edge eth1-3 75: 11109627 0 0 0 PCI-MSI-edge eth1-4 NMI: 0 0 0 0 Non-maskable interrupts LOC: 50455701 61286848 31629357 13702410 Local timer interrupts SPU: 0 0 0 0 Spurious interrupts PMI: 0 0 0 0 Performance monitoring interrupts PND: 0 0 0 0 Performance pending work
Re: [asterisk-users] PRI got event HDLC Abort
is the card sharing irq? no. this the only card that uses IRQ 30 1b:00.0 Network controller: Digium, Inc. Device 1420 (rev 14) Subsystem: Device 0005: Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr+ Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow TAbort- TAbort- MAbort- SERR- PERR- INTx- Latency: 64 (8000ns min, 32000ns max), Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 30 Region 0: Memory at 97a0 (32-bit, non-prefetchable) [size=32K] Kernel driver in use: wct4xxp is your system plugged directly into an outlet without ups? Please give us a complete lspci -vvv. Did you read this? http://alexrrr.blogspot.de/2007/10/solving-asterisks-hdlc-abort-issue.html -- _ -- 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] One side voice one side musiconhold
Currently I have no idea. But you wrote, that it does not happen all the time. Please provide us a log extract from that case where is going wrong. Perhaps you can do a diff for the good and the bad case yourself before?! Am 02.10.2012 09:02, schrieb Gianluca Baù: Hello Thorsten, i had a trace with core set debug 10 and core set verbose 10 but i didn't find anything usefull. The log is very full so it could be that i missed some important information. This is a the less verbose output of the problem: -- SIP/22-01b3 answered SIP/64-01b2 -- Started music on hold, class 'default', on SIP/22-01b3 -- Stopped music on hold on SIP/siprouter-01aa -- Executing [h@to-operators:1] Goto(SIP/64-01b2ZOMBIE, 9991) in new stack -- Goto (to-operators,h,9991) -- Executing [h@to-operators:9991] Set(SIP/64-01b2ZOMBIE, ~~parentcxt~~=) in new stack -- Executing [h@to-operators:9992] GotoIf(SIP/64-01b2ZOMBIE, 1?9996) in new stack -- Goto (to-operators,h,9996) -- Executing [h@to-operators:9996] NoOp(SIP/64-01b2ZOMBIE, ) in new stack Where: SIP/siprouter-01aa is A SIP/64 is B SIP/22 is C I think this is the moment of the transfer. -- Started music on hold, class 'default', on SIP/22-01b3 -- Stopped music on hold on SIP/siprouter-01aa After the transfer of the call from B it seems to start the music to C and to stop it on A. I'll try to provide you a better trace. Do you have any ideas about the cause? Thanks, regards Gianluca 2012/10/1 Thorsten Göllner t...@ovm-group.com: Did you take a look at the asterisk log? With core set verbose 3 or more? Am 01.10.2012 12:46, schrieb Gianluca Baù: Hello guys, my name is Gianluca and this is my first post in this ml. i've a strange problem with my asterisk box. I'll try to explain you. A (sip from ser) calls -- B (sip asterisk peer) B put A on hold with musiconhold B calls C B transfer the call with A to C A hears the C voice while C hears musiconhold C is every peer of the asterisk. This happens with version 1.6.22 and Asterisk 1.8.14.0 too. I tried to update but the problem persists. I've to say that the used phones are the same for both the versions. They are Snom and Grandstream. This problem is hard to debug because it doesn't happen everytime. Did you hear something about this problem? Can you suggest me how to understand this situation? Thanks, regards -- _ -- 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] One side voice one side musiconhold
Did you take a look at the asterisk log? With core set verbose 3 or more? Am 01.10.2012 12:46, schrieb Gianluca Baù: Hello guys, my name is Gianluca and this is my first post in this ml. i've a strange problem with my asterisk box. I'll try to explain you. A (sip from ser) calls -- B (sip asterisk peer) B put A on hold with musiconhold B calls C B transfer the call with A to C A hears the C voice while C hears musiconhold C is every peer of the asterisk. This happens with version 1.6.22 and Asterisk 1.8.14.0 too. I tried to update but the problem persists. I've to say that the used phones are the same for both the versions. They are Snom and Grandstream. This problem is hard to debug because it doesn't happen everytime. Did you hear something about this problem? Can you suggest me how to understand this situation? Thanks, regards -- _ -- 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] realtime_multi_mysql: MySQL RealTime: Invalid database specified
Maybe a stupid answer ;-) Did you make a "reload"? Did you try from shell: mysql -u myuser -pmysecret AsteriskHosted ? Am 27.09.2012 11:00, schrieb Jonas Kellens: Hello, this might seem a stupid question but I really don't see the solution to the problem. Using Asterisk 1.8.12.2 In extconfig.conf I have : voicemail = mysql,AsteriskHosted,voicemail_users sipusers = mysql,AsteriskHosted,sip_buddies sippeers = mysql,AsteriskHosted,sip_buddies queues = mysql,AsteriskHosted,queues queue_members = mysql,AsteriskHosted,queue_members In res_mysql I have : [AsteriskHosted] dbhost = 127.0.0.1 dbname = AsteriskHosted dbuser = myuser dbpass = mysecret dbport = 3306 dbsock = /var/lib/mysql/mysql.sock requirements=warn ; or createclose or createchar But still I get the error on Asterisk CLI : *CLI [Sep 27 10:47:20] WARNING[1693]: res_config_mysql.c:335 realtime_mysql: MySQL RealTime: Invalid database specified: AsteriskHosted (check res_mysql.conf) *CLI [Sep 27 10:47:57] WARNING[1693]: res_config_mysql.c:442 realtime_multi_mysql: MySQL RealTime: Invalid database specified: 'AsteriskHosted' (check res_mysql.conf) On 3 other servers I have installed, I have never had this problem. What can be the issue ?? Kind regards, Jonas. -- _ -- 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] realtime_multi_mysql: MySQL RealTime: Invalid database specified
Now I see: you try to use the wrong config file - try /etc/asterisk/res_config_mysql.conf instead. Am 27.09.2012 11:40, schrieb Jonas Kellens: On 27-09-12 11:27, Thorsten Gllner wrote: Maybe a stupid answer ;-) Did you make a "reload"? Yes, I reloaded and restarted several times. Did you try from shell: mysql -u myuser -pmysecret AsteriskHosted Yes, works perfect to connect via commandline. Only Asterisk does not see the database and I really don't know why. No hints in debug and verbose log either... Kind regards, Jonas. -- _ -- 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 -- Thorsten Gllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Dsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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] realtime_multi_mysql: MySQL RealTime: Invalid database specified
Now I see: you try to use the wrong config file - try /etc/asterisk/res_config_mysql.conf instead. Am 27.09.2012 11:40, schrieb Jonas Kellens: On 27-09-12 11:27, Thorsten Gllner wrote: Maybe a stupid answer ;-) Did you make a "reload"? Yes, I reloaded and restarted several times. Did you try from shell: mysql -u myuser -pmysecret AsteriskHosted Yes, works perfect to connect via commandline. Only Asterisk does not see the database and I really don't know why. No hints in debug and verbose log either... Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 / voicemail / final voice auth-thankyou
Hi, voicemail plays after hitting # as final file auth-thankyou. Is there any possibility to change this behaviour? Custom soundfile or disable it perhaps? Thanks for your answer(s)! -Thorsten- -- _ -- 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] Please dont tell me this is impossible
I just tried it on asterisk 1.8.13 with agi set debug on. The last log
line reveals it - streamfile return the endpos.
[2012-07-03 15:16:39] VERBOSE[7046] res_agi.c:
SIP/tgoellner-0002AGI Rx STREAM FILE /audio1/dtmf_detector/2.0
1234567890*#
[2012-07-03 15:16:39] VERBOSE[7046] res_agi.c: -- Playing
'/audio1/dtmf_detector/2.0' (escape_digits=1234567890*#) (sample_offset 0)
[2012-07-03 15:16:40] VERBOSE[7046] res_agi.c:
SIP/tgoellner-0002AGI Tx 200 result=0 endpos=4800
So please doublecheck your result.
Am 03.07.2012 00:47, schrieb CDR:
1.8 is my version, until the new one is stable.
On Mon, Jul 2, 2012 at 4:57 PM, Thorsten Göllner t...@ovm-group.com wrote:
What Asterisk version?
Am 02.07.2012 15:14, schrieb CDR:
Thanks. I already solved it using this command. The only issue was
that it gives you as return the ASCII code of the digit pressed
instead of the digit itself. For some reason my brain did not process
that detail. But it does work. However, the offset played is not
returned. Has anybody tested this and has a coding sample in perl?
Philip
On Mon, Jul 2, 2012 at 8:52 AM, Thorsten Göllner t...@ovm-group.com wrote:
So take a look here:
http://www.voip-info.org/wiki/view/stream+file
Am 29.06.2012 16:06, schrieb CDR:
This is from the documentation of Perl-AGI
$AGI-stream_file($filename, $digits, $offset)
Executes AGI Command STREAM FILE $filename $digits [$offset]
This command instructs Asterisk to play the given sound file and
listen for the given dtmf digits. The fileextension must not be used
in the filename because Asterisk will find the most appropriate file
type. $filename can be an array of files or a single filename.
Example: $AGI-stream_file('demo-echotest', '0123');
$AGI-stream_file(['demo-echotest', 'demo-welcome'], '0123');
Returns: -1 on error or hangup, 0 if playback completes without a
digit being pressed, or the ASCII numerical value of the digit if a
digit was pressed
It does not mention that it returns the offset at which the file
stopped playing. Also, if you could get that number, then restarting
the stream would result, I guess, in an audible interruption. Please
advise how to get the offset on the result and I will try.
Yours
Philip
On Fri, Jun 29, 2012 at 6:27 AM, Thorsten Göllner t...@ovm-group.com
wrote:
Am 29.06.2012 11:38, schrieb CDR:
I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI-command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has expired. So far I found impossible to achieve
this functionality. Am I missing something?
Philip
The Playcommand will be interrupted by the key but the agi result
contains
the offset. So you can play this file from offset again until you
$maxdigits
has been pressed. Take a look here:
https://wiki.asterisk.org/wiki/display/AST/AGICommand_STREAM+FILE
--
Thorsten Göllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf
Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54
--
Thorsten Göllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf
Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54
--
Thorsten Göllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf
Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54
--
_
-- 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
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Re: [asterisk-users] Please dont tell me this is impossible
Sorry, but I am using a self developed PHP-Library where I parse STDIN
myself. So I have no problem on this side. You are using a Perl-API?
There should be a method available for getting the AGI-Result-String?! I
never used Perl myself ...
Am 03.07.2012 16:13, schrieb CDR:
Yes, ai saw that information on the debug, but how do you bring it
inside a variable, so you may use it? I could not find a way. Maybe I
am missing something?
On Tue, Jul 3, 2012 at 9:20 AM, Thorsten Göllner t...@ovm-group.com wrote:
I just tried it on asterisk 1.8.13 with agi set debug on. The last log
line reveals it - streamfile return the endpos.
[2012-07-03 15:16:39] VERBOSE[7046] res_agi.c: SIP/tgoellner-0002AGI
Rx STREAM FILE /audio1/dtmf_detector/2.0 1234567890*#
[2012-07-03 15:16:39] VERBOSE[7046] res_agi.c: -- Playing
'/audio1/dtmf_detector/2.0' (escape_digits=1234567890*#) (sample_offset 0)
[2012-07-03 15:16:40] VERBOSE[7046] res_agi.c: SIP/tgoellner-0002AGI
Tx 200 result=0 endpos=4800
So please doublecheck your result.
Am 03.07.2012 00:47, schrieb CDR:
1.8 is my version, until the new one is stable.
On Mon, Jul 2, 2012 at 4:57 PM, Thorsten Göllner t...@ovm-group.com wrote:
What Asterisk version?
Am 02.07.2012 15:14, schrieb CDR:
Thanks. I already solved it using this command. The only issue was
that it gives you as return the ASCII code of the digit pressed
instead of the digit itself. For some reason my brain did not process
that detail. But it does work. However, the offset played is not
returned. Has anybody tested this and has a coding sample in perl?
Philip
On Mon, Jul 2, 2012 at 8:52 AM, Thorsten Göllner t...@ovm-group.com
wrote:
So take a look here:
http://www.voip-info.org/wiki/view/stream+file
Am 29.06.2012 16:06, schrieb CDR:
This is from the documentation of Perl-AGI
$AGI-stream_file($filename, $digits, $offset)
Executes AGI Command STREAM FILE $filename $digits [$offset]
This command instructs Asterisk to play the given sound file and
listen for the given dtmf digits. The fileextension must not be used
in the filename because Asterisk will find the most appropriate file
type. $filename can be an array of files or a single filename.
Example: $AGI-stream_file('demo-echotest', '0123');
$AGI-stream_file(['demo-echotest', 'demo-welcome'], '0123');
Returns: -1 on error or hangup, 0 if playback completes without a
digit being pressed, or the ASCII numerical value of the digit if a
digit was pressed
It does not mention that it returns the offset at which the file
stopped playing. Also, if you could get that number, then restarting
the stream would result, I guess, in an audible interruption. Please
advise how to get the offset on the result and I will try.
Yours
Philip
On Fri, Jun 29, 2012 at 6:27 AM, Thorsten Göllner t...@ovm-group.com
wrote:
Am 29.06.2012 11:38, schrieb CDR:
I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI-command. I need to play
a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has expired. So far I found impossible to
achieve
this functionality. Am I missing something?
Philip
The Playcommand will be interrupted by the key but the agi result
contains
the offset. So you can play this file from offset again until you
$maxdigits
has been pressed. Take a look here:
https://wiki.asterisk.org/wiki/display/AST/AGICommand_STREAM+FILE
--
_
-- 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
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To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Please dont tell me this is impossible
Am 29.06.2012 11:38, schrieb CDR: I have been fighting all night with version 1.8 and have not found a way to do this with any command or Perl AGI-command. I need to play a file and wait until the customer presses at least $maxdigits to return, BUT, the file must continue playing until $maxdigits is received or $timeout has expired. So far I found impossible to achieve this functionality. Am I missing something? Philip The Playcommand will be interrupted by the key but the agi result contains the offset. So you can play this file from offset again until you $maxdigits has been pressed. Take a look here: https://wiki.asterisk.org/wiki/display/AST/AGICommand_STREAM+FILE -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk with ss7 voice broadcast
Am 21.06.2012 11:30, schrieb [Digital^Dude] ®: Asterisk 1.8.7.1 built by root on a x86_64 running Linux. CentOS release 5.5 (Final) RAM: 4 GB CPU: Dual Xeon 2.66 GHz Asterisk is running as root data seg size (kbytes, -d) unlimited file size (blocks, -f) unlimited pending signals (-i) 38912 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 4096 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 38912 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited The changes in ulimit apparently don't get reflected when I run a broadcast on asterisk. Perhaps max. 4096 open files is too low? Try to increase it to 8192. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 / sending fax / spandsp
Hi, I need a fax-send - setup. I read the book Asterisk The Definitive Guide chapter 19 (fax) and found 2 options listed there. 1) Using spandsp. 2) Using FFA (Digium Fax For Asterisk). But the book nor any other article I read point out, what the differences or drawbacks are. Does anyone of you have experience with one or both solutions? We use: - Asterisk 1.8.13 - Sangoma AFT A104d (germany, E1) - libpri - DAHDI 2.6.1 Thanks for any hint. -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.13.0 / problem with cdr logging (mysql, odbc)
Am 18.06.2012 21:49, schrieb James Sharp: On 6/18/2012 11:52 AM, Thorsten Göllner wrote: Hi, I am trying now for over 4 hours setting up cdr-logging via odbc into a mysql database. But with no success. Do you have any hint for me? *SNIP* But after a call hangup I get the following error: cdr_odbc.c: Unable to retrieve database handle. CDR failed. What is going wrong here, please? The DSN that you specify in cdr_odbc.con should be the DSN you configured in res_odbc.conf, in this case mysql versus MySQL-asterisk. I beat head against the desk for hours because of this same issue. This solved the problem. The CDR is written to the mysql database. Thanks. It would never have occurred to me. BUT the cli command odbc show all still shows and uninitialized last connection attempt. Never mind? CLI odbc show all ODBC DSN Settings - Name: mysql DSN:MySQL-asterisk Last connection attempt: 1970-01-01 01:00:00 Pooled: No Connected: Yes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk with ss7 voice broadcast
Am 19.06.2012 11:53, schrieb [Digital^Dude] ®: Machine specs: CentOS release 5.5 (Final) RAM: 4 GB CPU: Dual Xeon 2.66 GHz Asterisk 1.8.7.1 built by root on a x86_64 running Linux. *CLI ulimit core Core file size (core) is effectively unlimited. *CLI ulimit data Program data segment (data) is effectively unlimited. *CLI ulimit descriptors *Number of file descriptors (descriptors) is limited to 178414.* *CLI ulimit file File size (file) is effectively unlimited. *CLI ulimit locked *Amount of memory locked into RAM (locked) is limited to 32768.* *CLI ulimit memory Resident memory (memory) is effectively unlimited. *CLI ulimit processes *Number of processes (processes) is limited to 38912.* *CLI ulimit stack Program stack size (stack) is effectively unlimited. *CLI ulimit time Cpu time (time) is effectively unlimited. *CLI ulimit virtual Virtual memory (virtual) is effectively unlimited. Now take a look at: /etc/security/limits.conf Try this: give user root AND the sser running asterisk (user asterisk?) the following limits and try again: [...] rootsoftnofile 4096 roothardnofile 8196 asterisksoftnofile 4096 asteriskhardnofile 8196 [...] Reboot after change! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk with ss7 voice broadcast
Did you check "ulimits" in Asterisk CLI? Am 14.06.2012 16:02, schrieb [Digital^Dude] : Hello, Asterisk under 90% load of SS7 calls can only withstand the voice broadcasting for 30 minutes. After around 30 minutes, it stops receiving any call hits via AMI. No errors are reported. Giving it a minute's rest makes it work for another 30 minutes. Can anyone hint to what may be causing this? -- Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.13.0 / problem with cdr logging (mysql, odbc)
Hi, I am trying now for over 4 hours setting up cdr-logging via odbc into a mysql database. But with no success. Do you have any hint for me? cat /etc/odbc.ini -- [MySQL-asterisk] Description = MySQL ODBC Driver Driver = MySQL Socket = /var/run/mysqld/mysqld.sock Server = localhost User = xxx Password = xxx Database = asterisk Option = 3 Port = and /etc/odbcinst.ini [MySQL] Description = MySQL ODBC MyODBC Driver Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so FileUsage = 1 When testing this setup I can see, that this basic setup ist fine: ~$ isql MySQL-asterisk asterisk qpalym -v +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL So here are the config file for asterisk. /etc/asterisk/res_odbc.conf - [mysql] enabled = yes dsn = MySQL-asterisk username = asterisk password = qpalym pre-connect = yes and /etc/asterisk/cdr_odbc.conf [global] dsn=MySQL-asterisk loguniqueid=yes dispositionstring=yes table=cdr But after a call hangup I get the following error: cdr_odbc.c: Unable to retrieve database handle. CDR failed. What is going wrong here, please? -Thorsten- -- _ -- 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] Another IP address to block
Where can I find such ip-lists, please? Am 05.06.2012 18:40, schrieb Alejandro Imass: We use complete regional blocks from Wizcraft and blocking at minimum all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We block almost anything that is not our actual customer market and screw the rest. On Tue, Jun 5, 2012 at 12:14 PM, Carlos Chavezcur...@telecomabmex.com wrote: Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with OpenSMS API?
What do you want to do? Sending and receiving SMS? Am 03.06.2012 11:20, schrieb Michelle Konzack: Hello Experts, since connecting of 4 Huawei K3765-HV Sticks to my Server does not work, I now use the Vodafone EasyBox 803A (cost less then 30 Euro on eBay) and connect them to my ISDN cards. It has the advantage, that I can use in the same time the UMTS Internet connectivity and local analog telephones. However, if I use Windows, the program shiped with the EasyBox use the OpenSMS API to get the SMS from the USB-Stick trough the EasyBox. So, my questions are: 1) Does Asterisk (or an AddOn/PlugIn) support the OpenSMS API? 2) Does someone know, where I can get infos about the OpenSMS API? I have found nothing on Google. Thanks, Greetings and nice Day/Evening Michelle Konzack -- _ -- 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 -- Thorsten Gllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Dsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] German voice recognition
Hi, I am looking (for the best) solution to recognize *german* words or simple phrases with a given number of words (eins, zwei drei etc. or hauptmenü, zurück etc.). Can somebody give me a good link? Can I find external service providers who can be accessed via ASR()? Best regards, -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libpri / ISDN feature ECT (explicit call transfer)
Hi, since version 1.4.12 the libpri package supports ETSI Explicit Call Transfer feature: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.12 Does anyone know, how to use this feature in the dialplan? I can not find any hints in the asterisk doc. Best regards, -Thorsten- -- _ -- 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
