Re: [asterisk-users] Pass DTMF to IVR gateway through SIP phone conferencing.
Christian, Thanks for your response. In my case, I was asked to do it through SIP phone 3 way call functionality and not the Asterisk Meetme application. I wanted to know if any one had done something similar in past or not. I am short of PRI in my test environment and hence I can't test it practically. Well, I 'll try to implement it using Meetme. Regards, AsteriskMan -- _ -- 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] SOLVED: Re: Setting `userfield` from within a callfile
On Monday 20 Dec 2010, Olivier wrote:
2010/12/20 A J Stiles asterisk_l...@earthshod.co.uk
Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application
(written by someone else before me) which sets up calls by creating
files of
the general form
Channel: SIP/$INSIDE_NUMBER
Context: $CONTEXT
Extension: $OUTSIDE_NUMBER
Priority: 1
CallerId: $INSIDE_NUMBER
in /var/spool/asterisk/outgoing/ .
It works very well. However, it would be nice to be able to attach an
additional piece of information along with the call record There is a
userfield in the SQL database, which is a VARCHAR(255) and would be
plenty for what we need. Is there a way to set the userfield of the CDR
database from within such a callfile?
Yes, adding a Set field in your call file (see
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out), you'll be able
to pass everything you need to your dialplan, and then, from there, write
everything you need to your CDR.
I've got it working now! Thanks Olivier and Tilghman.
Now, for the benefit of anyone who may be searching the archives of this
mailing list at some point in the future, here's what I did.
I have modified the callfile-generating CGI script to added an extra line to
the callfile, something like;
Set: uid=$UID
and made sure that the calls it places are in a context of their own. In my
extensions.conf, I then have as part of that context, a line ending with the
command
... ,Set(CDR(userfield)=${uid})
and it all Just Works Beautifully.
Thanks again, everyone. Asterisk truly is a wonderful piece of software.
--
AJS
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[asterisk-users] app_voicemail.c how to enable debugging?
Hi Looking at the source of app_voicemail.c there are many statements like: ast_debug(1, %s doesn't exist, doing what we can\n, prefile); Where do I have to enably this to be showed in the console or logged to a file by logger. core set debug does not seem to help here. Well, my actual problem is, that if a customer has recorded his own greeting, he usualy tells the caller to record his message after the tone, so app_voicemail should not play the intro. spool/mailbox/unavail.gsm vm-intro.gsm beep.gsm but only spool/mailbox/unavail.gsm beep.gsm In case there is an unavailable message. Where do I have to poke at the source? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- 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 1.6 produces *many* zombie processes on Debian.
Am 20.12.2010 21:39, schrieb Ernie Dunbar: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. What does /var/log/asterisk/messages say? And /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] app_voicemail.c how to enable debugging?
On Tue, Dec 21, 2010 at 11:47:02AM +0100, Benoit Panizzon wrote: [snip] Well, my actual problem is, that if a customer has recorded his own greeting, he usualy tells the caller to record his message after the tone, so app_voicemail should not play the intro. spool/mailbox/unavail.gsm vm-intro.gsm beep.gsm but only spool/mailbox/unavail.gsm beep.gsm In case there is an unavailable message. Where do I have to poke at the source? No need to patch app_voicemail to do this I guess, passing the 's' argument to VoiceMail will skip vm-intro. So you only need to figure out is unavail.gsm exists from the dialplan to add 's' to the arguments. Implementing this in an AGI script should be trivial. -- Daniel Tryba -- _ -- 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] MeetMe - ConfBridge: hint not working
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten = 81,hint,ConfBridge:81
;;exten = 82,hint,MeetMe:82
exten = 82,hint,ConfBridge:82
;;exten = 83,hint,MeetMe:83
exten = 83,hint,ConfBridge:83
;;exten = 84,hint,MeetMe:84
exten = 84,hint,ConfBridge:84
And that does not work. The blf does not go on when a party is in
ConfBridge. Is there some new syntax for hints with ConfBridge?
sean
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[asterisk-users] Friend/user/peer in plain English?
Hello I've done some googling, but still puzzled at my working configuration. Apparently, a user can only receive calls through Asterisk, a peer can only make calls, and a friend can do both. If that's correct, I don't understand why my VOSP requires the following settings in sip.conf to let my Asterisk server make/receive calls to/from the PSTN: = [general] ... register = me:p...@vosp.com [vosp_outgoing] type=peer host=vosp.com username=me secret=pass fromuser=me fromdomain=vosp.com nat=yes canreinvite=no qualify=yes [vosp_incoming] ;why not type=user? type=peer host=vosp.com context=from_vosp nat=yes canreinvite=no insecure=port,invite qualify=yes [6011] type=friend secret=pass context=my-phones host=dynamic qualify=yes nat=no = I would expect [vosp_outgoing] to be of type=peer, while [vosp_incoming] should be type=user. As a side-note, why do we need both a register and fromuser/secret to make calls through a VOSP? Thanks for any hint. -- _ -- 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] Call sip:u...@domain.com?
On Mon, 20 Dec 2010 12:39:44 -0600, Kevin P. Fleming kpflem...@digium.com wrote: You've missed a very important point here: you are using a *SIP* endpoint to call a *SIP* URI. The endpoint can do that directly, and doesn't need any help from Asterisk to do it. If you wanted to be able to restrict/control such calls, you'd need to use a SIP proxy... but Asterisk is not a proxy. Asterisk is a Back-to-Back User Agent, which means whatever URI the endpoint sends to Asterisk terminates there, and Asterisk constructs an outbound URI of some form, connecting the two channels together. Thanks much Kevin. I found this article helpful to have a better understanding of what a B2BUA is compared to an SIP proxy: www.voip-info.org/wiki/view/Asterisk+SIP+not-proxy One advantage I see in using Asterisk even when the two end-points are SIP, is that I end up with a single application to handle calls between end-points (SIP, VOSP, and FXO) and provide additional features like voice-mail, etc. But I could use a good article/book to better understand my options, how Asterisk is different from the alternatives (Freeswitch, openSIPS, etc.) www.amazon.com/s/ref=nb_sb_noss?url=search-alias%3Dstripbooksfield-keywords=voip Thank you. -- _ -- 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 420
On 12/20/2010 07:08 PM, Dovey Forman wrote: Thanks Kevin. Did it work with Asterisk 1.2 because it ignored it? I don't know specifically that Asterisk 1.2 ignored Required headers, but it's certainly possible. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Call sip:u...@domain.com?
On Tuesday 21 Dec 2010, Gilles wrote: But I could use a good article/book to better understand my options, how Asterisk is different from the alternatives (Freeswitch, openSIPS, etc.) www.amazon.com/s/ref=nb_sb_noss?url=search-alias%3Dstripbooksfield-keyword s=voip The same way Ubuntu, Slackware, CentOS c. differ from each other. They are all using the Linux kernel and the X Window System under the bonnet. Well, every Free and Open Source telephony system is using Asterisk (and Linux) under the bonnet. The differences are in the user configuration tools. -- AJS -- _ -- 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] Call sip:u...@domain.com?
On 21 Dec 2010, at 14:20, A J Stiles wrote: Well, every Free and Open Source telephony system is using Asterisk (and Linux) under the bonnet. The differences are in the user configuration tools. Uh, no? S -- _ -- 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] Call sip:u...@domain.com?
On Tue, 21 Dec 2010 14:20:55 +, A J Stiles asterisk_l...@earthshod.co.uk wrote: The same way Ubuntu, Slackware, CentOS c. differ from each other. They are all using the Linux kernel and the X Window System under the bonnet. Well, every Free and Open Source telephony system is using Asterisk (and Linux) under the bonnet. The differences are in the user configuration tools. According to this article, it appears that what really makes a B2BUA different from an SIP register/proxy is that a B2BUA can manage media (voicemail, etc.) while an SIP proxy doesn't: www.tinyurl.com/Asterisk-vs-OpenSIPS -- _ -- 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] MeetMe - ConfBridge: hint not working
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten = 81,hint,ConfBridge:81
;;exten = 82,hint,MeetMe:82
exten = 82,hint,ConfBridge:82
;;exten = 83,hint,MeetMe:83
exten = 83,hint,ConfBridge:83
;;exten = 84,hint,MeetMe:84
exten = 84,hint,ConfBridge:84
And that does not work. The blf does not go on when a party is in
ConfBridge. Is there some new syntax for hints with ConfBridge?
sean
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Re: [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.
Am 20.12.2010 21:39, schrieb Ernie Dunbar: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. What does /var/log/asterisk/messages say? And /var/log/syslog? Not much. In /var/log/asterisk/messages here's a lot of lines like this: [Dec 17 19:10:13] NOTICE[25518] chan_sip.c: Registration from 'sip:xx...@voip.lightspeed.ca' failed for 'XX.XXX.X.XXX' - No matching peer found And /var/log/syslog has all the normal output from a2billing.php and making calls complete and such. The other funny thing is that except for the massive number of zombie processes, calls are being made and completed just fine. Even voice quality is quite high. -- _ -- 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 1.6 produces *many* zombie processes on Debian.
Your server is being brute-forced. Read this article (http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk) and set up fail2ban on your machine right now.Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brasil+55 54 2104-7000Information Security ManagerCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brazil+55 54 2104-7000 Am 20.12.2010 21:39, schrieb Ernie Dunbar: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. What does /var/log/asterisk/messages say? And /var/log/syslog?Not much. In /var/log/asterisk/messages here's a lot of lines like this:[Dec 17 19:10:13] NOTICE[25518] chan_sip.c: Registration from'sip:xx...@voip.lightspeed.ca' failed for 'XX.XXX.X.XXX' - No matchingpeer foundAnd /var/log/syslog has all the normal output from a2billing.php andmaking calls complete and such.The other funny thing is that except for the massive number of zombieprocesses, calls are being made and completed just fine. Even voicequality is quite high.--_-- 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/helloasterisk-users mailing listTo 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 1.6 produces *many* zombie processes on Debian.
On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar maill...@lightspeed.ca wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. Simple In sip.conf please set alwaysauthreject = yes ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.
Actually, no. This is part of a migration, and those are mostly customers' secondary lines (which for the most part, aren't even active). We get a lot of these bad logins because the retry times on the ATAs are quite short. Asterisk really *shouldn't* leave zombies around for every bad login, but if it does, then I suppose cleaning up these missing accounts might fix it. Your server is being brute-forced. Read this article (http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk) and set up fail2ban on your machine right now. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Mensagem original - Am 20.12.2010 21:39, schrieb Ernie Dunbar: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. What does /var/log/asterisk/messages say? And /var/log/syslog? Not much. In /var/log/asterisk/messages here's a lot of lines like this: [Dec 17 19:10:13] NOTICE[25518] chan_sip.c: Registration from 'sip:xx...@voip.lightspeed.ca' failed for 'XX.XXX.X.XXX' - No matching peer found And /var/log/syslog has all the normal output from a2billing.php and making calls complete and such. The other funny thing is that except for the massive number of zombie processes, calls are being made and completed just fine. Even voice quality is quite high. -- _ -- 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
[asterisk-users] What is equivalent function to mv command in php for Asterisk Spool directory usage?
Hi Everyone, I understand that there are a few warnings about using cp to move .call files to /var/spool/asterisk/outgoing as they might acted upon before copy is done. So, using PHP, What is the equivalent of mv command? Would it be rename() in php or is there a better method? 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
Re: [asterisk-users] What is equivalent function to mv command in php for Asterisk Spool directory usage?
PERL has a move() command; I wouldn't expect less out of PHP. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Tuesday, December 21, 2010 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] What is equivalent function to mv command in php for Asterisk Spool directory usage? Hi Everyone, I understand that there are a few warnings about using cp to move .call files to /var/spool/asterisk/outgoing as they might acted upon before copy is done. So, using PHP, What is the equivalent of mv command? Would it be rename() in php or is there a better method? 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
Re: [asterisk-users] What is equivalent function to mv command in php for Asterisk Spool directory usage?
I think rename() is what you're looking for http://php.net/manual/en/function.rename.php On Tue, Dec 21, 2010 at 2:23 PM, Danny Nicholas da...@debsinc.com wrote: PERL has a move() command; I wouldn’t expect less out of PHP. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Tuesday, December 21, 2010 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] What is equivalent function to mv command in php for Asterisk Spool directory usage? Hi Everyone, I understand that there are a few warnings about using cp to move .call files to /var/spool/asterisk/outgoing as they might acted upon before copy is done. So, using PHP, What is the equivalent of mv command? Would it be rename() in php or is there a better method? 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
Re: [asterisk-users] What is equivalent function to mv command in php for Asterisk Spool directory usage?
I have been using:
exec ('mv *.call /var/spool/asterisk/outgoing')
and for a long time it has been working just fine for me on more than one
websites. Just make sure the folder where you create the call files has
correct permissions and ownerships so that the file is successfully moved by
the apache user to its destination.
Zeeshan A Zakaria
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On 2010-12-21 3:29 PM, Danny Nicholas da...@debsinc.com wrote:
PERL has a move() command; I wouldn’t expect less out of PHP.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B
*Sent:* Tuesday, December 21, 2010 2:20 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] What is equivalent function to mv command in
php for Asterisk Spool directory usage?
Hi Everyone,
I understand that there are a few warnings about using cp to move .call
file...
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Re: [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
2010/12/21 Paul Belanger pabelan...@digium.com: On 10-12-20 05:51 PM, Jarek Jarzebowski wrote: OK, so I have attached debug log. I am using: *CLI core show version Asterisk 1.8.1.1 built by root @ asterisk on a i686 running Linux on 2010-12-17 23:03:58 UTC Definitely a bug, ran into the same issue with chan_iax2 and DNS lookups. Please open a new issue on the tracker, include your debug log and sip.conf. So I have opened new issue #0018514. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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 Regards, Jarek -- _ -- 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 1.6 produces *many* zombie processes on Debian.
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. I know what the issue is. Please open a report on https://issues.asterisk.org and I'll get a patch uploaded pronto. -- Tilghman -- _ -- 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] SOLVED: Re: Setting `userfield` from within a callfile
On Tuesday 21 December 2010 04:49:42 A J Stiles wrote:
On Monday 20 Dec 2010, Olivier wrote:
2010/12/20 A J Stiles asterisk_l...@earthshod.co.uk
Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial
application (written by someone else before me) which sets up
calls by creating files of
the general form
Channel: SIP/$INSIDE_NUMBER
Context: $CONTEXT
Extension: $OUTSIDE_NUMBER
Priority: 1
CallerId: $INSIDE_NUMBER
in /var/spool/asterisk/outgoing/ .
It works very well. However, it would be nice to be able to attach
an additional piece of information along with the call record
There is a userfield in the SQL database, which is a VARCHAR(255)
and would be plenty for what we need. Is there a way to set the
userfield of the CDR database from within such a callfile?
Yes, adding a Set field in your call file (see
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out), you'll be
able to pass everything you need to your dialplan, and then, from
there, write everything you need to your CDR.
I've got it working now! Thanks Olivier and Tilghman.
Now, for the benefit of anyone who may be searching the archives of this
mailing list at some point in the future, here's what I did.
I have modified the callfile-generating CGI script to added an extra
line to the callfile, something like;
Set: uid=$UID
and made sure that the calls it places are in a context of their own.
In my extensions.conf, I then have as part of that context, a line
ending with the command
... ,Set(CDR(userfield)=${uid})
and it all Just Works Beautifully.
You shouldn't need to. You should be able to just do:
Set: CDR(userfield)=$UID
in your spoolfile and it should work from there.
--
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[asterisk-users] Simplifying dial-plan
Is there a way to include: _NXXNXX _NXX _011. _911 into my current plan: -- _ -- 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] Simplifying dial-plan
On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese rsre...@gmail.com wrote:
Is there a way to include:
_NXXNXX
_NXX
_011.
_911
into my current plan:
Sorry, here's the rest.
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = 201]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten = _1NXXNXX,n,Goto(h,1)
Thanks
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Re: [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.
On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar maill...@lightspeed.ca wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. Simple In sip.conf please set alwaysauthreject = yes Thanks for the tip, but we already did that a while ago. :) -- _ -- 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 as a caller ID
In a 1.4.24 system, out of several lines, one of ours gets 1 or more random calls a day with Asterisk as the caller ID. I have just seen this described in the last couple of weeks, but at the time it wasn't happening to us, and I the explanation didn't stick with me. Can anyone give me a pointer to this feature? Searching the message base for Asterisk seems futile. Thanks! Cary Fitch -- _ -- 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 as a caller ID
Cary Fitch wrote:
In a 1.4.24 system, out of several lines, one of ours gets 1 or more random
calls a day with Asterisk as the caller ID.
In my experience, it happens when the caller is blocking their CID. I
have programming in place that assigns the named restricted and the
phone number of 0 if the caller-id is blank:
exten = s,1,GotoIf($[${CALLERID(num)} = ]?2:3)
exten = s,n,Set(CALLERID(all)=Restricted 0)
Doug
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Re: [asterisk-users] MeetMe - ConfBridge: hint not working
On 12/21/2010 12:13 PM, Jeremy Betts wrote:
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten = 81,hint,ConfBridge:81
;;exten = 82,hint,MeetMe:82
exten = 82,hint,ConfBridge:82
;;exten = 83,hint,MeetMe:83
exten = 83,hint,ConfBridge:83
;;exten = 84,hint,MeetMe:84
exten = 84,hint,ConfBridge:84
And that does not work. The blf does not go on when a party is in
ConfBridge. Is there some new syntax for hints with ConfBridge?
sean
core show version
Asterisk 1.6.2.16-rc1
sean
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Re: [asterisk-users] MeetMe - ConfBridge: hint not working
On 12/21/2010 10:03 PM, sean darcy wrote:
On 12/21/2010 12:13 PM, Jeremy Betts wrote:
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten = 81,hint,ConfBridge:81
;;exten = 82,hint,MeetMe:82
exten = 82,hint,ConfBridge:82
;;exten = 83,hint,MeetMe:83
exten = 83,hint,ConfBridge:83
;;exten = 84,hint,MeetMe:84
exten = 84,hint,ConfBridge:84
And that does not work. The blf does not go on when a party is in
ConfBridge. Is there some new syntax for hints with ConfBridge?
sean
core show version
Asterisk 1.6.2.16-rc1
sean
BTW, wasn't device state handling added to ConfBridge last March?
https://issues.asterisk.org/view.php?id=16972
sean
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Re: [asterisk-users] What is equivalent function to mv command in php for Asterisk Spool directory usage?
On Tue, 21 Dec 2010, Bruce B wrote:
So, using PHP, What is the equivalent of mv command? Would it be
rename() in php or is there a better method?
Not really an Asterisk question...
On Tue, 21 Dec 2010, MrHanMan wrote:
I think rename() is what you're looking for
+1
On Tue, 21 Dec 2010, Zeeshan Zakaria wrote:
exec ('mv *.call /var/spool/asterisk/outgoing')
This will create a new process -- a relatively expensive activity.
Also, this assumes the path to mv is in the PATH environment variable
which it usually is, but if not can lead to a lot of confusion.
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Re: [asterisk-users] DIALSTATUS on CANCEL
Anyone??
Thanks.
On Mon, Dec 20, 2010 at 10:42 AM, VoIP Question voip.quest...@gmail.comwrote:
Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
[incoming-private]
exten = _X., n, Dial(SIP/1001,30)
exten = _X., n, NoOp(${DIALSTATUS})
exten = _X., n, Gosub(incoming-status,s-${DIALSTATUS},1)
[incoming-status]
exten = s-CANCEL,1, NoOp()
exten = s-CANCEL,n, Return()
exten = s-NOANSWER,1, NoOp()
exten = s-NOANSWER,n, Return()
exten = s-BUSY,1, NoOp()
exten = s-BUSY,n, Return()
This is what we get on a BUSY call:
---
-- Executing [1...@incoming-private:3] Dial(SIP/Proxy-002b,
SIP/1001,50) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- Got SIP response 486 Busy Here back from 10.0.0.1
-- SIP/1001-002c is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [1...@incoming-private:4] NoOp(SIP/Proxy-002b,
BUSY) in new stack
-- Executing [1...@incoming-private:5] Gosub(SIP/Proxy-002b,
incoming-status,s-BUSY,1) in new stack
This is what we get on a NO ANSWER call:
---
-- Executing [1...@incoming-private:3] Dial(SIP/Proxy-002f,
SIP/1001,30) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- SIP/1001-0030 is ringing
-- Nobody picked up in 3 ms
-- Executing [1...@incoming-private:4] NoOp(SIP/Proxy-002f,
NOANSWER) in new stack
-- Executing [1...@incoming-private:5] Gosub(SIP/Proxy-002f,
incoming-status,s-NOANSWER,1) in new stack
This is what we get on a CANCEL call:
-
-- Executing [1...@incoming-private:3] Dial(SIP/Proxy-0031,
SIP/1001,30) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- SIP/1001-0032 is ringing
== Spawn extension (incoming-private, , 3) exited non-zero on
'SIP/Proxy-0031'
There's no event indicating that a DIALSTATUS is generated and the call
simply doesn't go to the next step in the dialplan. Unless I'm missing
something, it seems to me that it might be a bug.
I would be happy to get feedback from other users of the DIALSTATUS value
(or Digium), especially in the CANCEL scenario.
Thank you,
Michael
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Re: [asterisk-users] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI
2010/12/18 Matt Riddell li...@venturevoip.com On 17/12/10 5:56 PM, Olivier wrote: Hi, Did you use libpri 1.4.11.5 or 1.4.12-beta ? Recently l tried 1.4.11.5 on a live system and it failed (Asterisk 1.6.1.18 and dahdi trunk, Junghanns QuadBRI, PtmP lines). Going back to 1.4.11.2 solved it. Unfortunately, I couldn't note what error message were then generated. Heh, latest everything - so LibPRI trunk. I did try going backwards in terms of DAHDI, but not LibPRI - will try that on Monday. Could you try going backwards with LibPRI ? I'm very curious if the problem I met the other day with 1.4.11.5 comes from my config or is tied to LibPRI. Cheers By the way, Kevin/Russell etc, any chance we could get a test added to bamboo for physical connectivity? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] Simplifying dial-plan
Hi Stephen,
_NXXNXX
_NXX
_011.
_911
Of course it can, but it depends on what you want to do when those
numbers are called...
I didn't know about the setvar in the sip.conf and actually I think it
is a much cleaner solution. Since you are already using it I would
suggest to not only use it for CallerID but also for the
@vitel-outbound like this:
exten = _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@${outbound})
exten = _1NXXNXX,n,Goto(h,1)
Of course you'll need to set setvar=Outbound=vitel-outbound or
setvar=Outbound=vitel-outbound2 in sip.conf.
What do you want to do with the other numbers? If you want to do the
same as with _1NXXNXX you can just add things like this in your
extensions.conf:
exten = _NXXNXX,1,Goto(_1NXXNXX,1)
exten = 911,1,Goto(_1NXXNXX,1)
Or you can do different things if you want that like this:
exten = _NXX,1,Set(CALLERID(all)=No one cares 0)
exten = _NXX,n,Dial(SIP/${ext...@abcdefgh)
exten = _NXX,n,Goto(h,1)
Best regards,
Jeroen Eeuwes
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Re: [asterisk-users] DIALSTATUS on CANCEL
Hi Enable debug level to more than 1 ,you may get something. Thanks Nikhil On 12/22/2010 11:26 AM, Michael wrote: Spawn extension (incoming-private, , 3) exited non-zero on 'SIP/Proxy-0031' -- _ -- 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
