Re: [asterisk-users] Automatic dialing + SMS
Hi Bilal, You can possible to do automatic dialing, play the proper sound message for a list of numbers using Asterisk. You need to write Asterisk AGI for automatic dialing. About sending SMS is also possible but you need to use SMS Gateway or SMS service provider. If u need further discussion we will assist you. -- Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,VoIP,Asterisk Technology On Tue, May 17, 2011 at 2:13 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello All; If I need the Asterisk to do automatic dialing for a list of numbers and when the destination answer, then to play the proper sound message, is it possible? How? About sending SMS, can asterisk do 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 -- _ -- 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 call stored procedure
MYSQL(Nextresult resultid ${connid}) after MySQL(Fetch fetchid ${resultid}
pass) helped to resolve this.
you should always get all results sp produce otherwise mysql returns error.
On Tue, May 17, 2011 at 4:36 PM, Borin katerin.bo...@gmail.com wrote:
Hi Guys,
I am getting an error when executing another mysql query in dialplan after
calling stored procedure.
If calling the procedure from mysql cli it gives a result like:
mysql call call_control(78236721,1000,1233);
+--+
| pass |
+--+
|1 |
+--+
So I need asterisk to recognize this pass and take some actions based on
what the pass value is.
Dialplan looks like this:
MYSQL(Connect connid ${DBDefaultHost} ${DBuser} ${DBpass} ${DBname})
MySQL(Query resultid ${connid} CALL call_control(78236721,1000,1233))
MySQL(Fetch fetchid ${resultid} pass)
MYSQL(clear ${resultid})
MySQL(Query resultid ${connid} SELECT/INSERT whatever from table)
So, it gives me this pass value correct, but if I execute some other query
INSERT or SELECT after clearing the result, it gives me an error
[May 17 16:16:13] WARNING[19572]: app_addon_sql_mysql.c:374 aMYSQL_query:
aMYSQL_query: mysql_query failed. Error: Commands out of sync; you can't run
this command now
The error disappears if I reconnect to mysql after calling the stored
procedure but it seams not right to me to connect to mysql 2 times for 1
call.
Did anyone have the same issue?
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Re: [asterisk-users] Automatic dialing + SMS
Hi Bilal, sure is possible as is possible to do other activities after played the messages such call redirect, dtmf selection and so on. About sending sms you can do it but our tips is to use an external SMS gateway in your area. Where are you ? If needed we can support, bye. Enrico www.rdmnet.it Il 17/05/2011 10:43, bilal ghayyad ha scritto: Hello All; If I need the Asterisk to do automatic dialing for a list of numbers and when the destination answer, then to play the proper sound message, is it possible? How? About sending SMS, can asterisk do 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 -- _ -- 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] Sending SRTP to Asterisk Gateway ends up in authentication failure
Hi @ all, I´m trying to send SRTP packets to an asterisk 1.8.4 Gateway with my own developed softphone. I am using libsrtp to prtect the rtp packets. At the moment I do SRTP without a key management and creating my key with the crypto_get_random function of libsrtp. The key size is 30B. For creating the SDP offer with crypto I encode the key to base64. The encryption type I want to use is AES_CM_128_HMAC_SHA1_80. for crypto_policy I use the crypto_policy_set_rtp_default. By sending the protected SRTP packets Asterisk Gateway ends up with SRTP_unprotect: authentication failure. My payload is 240B large and after protecting 250B. What could be wrong? How I can find out if I´m using same key and key derivations like asterisk? Best Regards Karsten -- _ -- 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 know how many calls are on hold
core show channels concise Those with '(None)' haven't been briged yet. On 17 May 2011 15:16, virendra bhati virbh...@gmail.com wrote: hi list, please help me how to know how many calls are on hold. -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] Automatic dialing + SMS
Hello. Does it mean Asterisk has no in-built applications for auto dialing. What scripting language can easily and best be used for the AGI. Tell me more abt the sms providers Sent from my BlackBerry® smartphone from Vodafone -Original Message- From: Enrico Cicconi enrico.cicc...@rdmnet.it Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 18 May 2011 09:54:45 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: bilal ghayyadbilmar...@yahoo.com Subject: Re: [asterisk-users] Automatic dialing + SMS Hi Bilal, sure is possible as is possible to do other activities after played the messages such call redirect, dtmf selection and so on. About sending sms you can do it but our tips is to use an external SMS gateway in your area. Where are you ? If needed we can support, bye. Enrico www.rdmnet.it Il 17/05/2011 10:43, bilal ghayyad ha scritto: Hello All; If I need the Asterisk to do automatic dialing for a list of numbers and when the destination answer, then to play the proper sound message, is it possible? How? About sending SMS, can asterisk do 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 -- _ -- 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] Make Multiple Calls using Chan_alsa module
Hi I would like to use asterisk as a SIP client(IP phohone ) with multiple user,multiple line support . Using existing chan_alsa driver I am not able to achieve my requirement . Please give some hint to do this . Thanks Nikhil -- _ -- 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] Skype-like dialing from web page
On Tuesday 17 May 2011, Mike wrote:
Hi,
Is there any softphone or TAPI plug-in that allows one to dial from a web
page?
Just write a simple CGI script (running from the Asterisk server) which
looks up the nearest phone from the remote IP address ( $ENV{REMOTE_ADDR} in
Perl), and inject a suitably-modified call file into the
folder /var/spool/asterisk.outgoing/ . The file needs to look something like
the below example:
-8-
Channel: SIP/$SRC
Context: $CTXT
Extension: $DEST
Priority: 1
CallerId: $SRC
-8-
where $SRC = source number, $CTXT = context and $DEST = destination.
NB, create the file in /tmp/ first then mv it; this way you can be sure
Asterisk will never try to parse an incomplete callfile. If doing this in
Perl, you *may* get away with keeping the callfile under 1 disk block as long
as $| = 0 on the filehandle, but this is by no means portable.
--
AJS
Answers come *after* questions.
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[asterisk-users] Using Asterisk/Digium repos = Astribank firmware not found
Hello, I'm writing here hoping to have a hint from Asterisk/Digium packager maintainer, Jason Parker (of course that any other's opinion is welcomed). We have installed an asterisk machine using Asterisk and Digium repos. Unfortunately we have found that an Astribank could not be connected due to the lack of the firmware files May 18 14:19:22 tenora1 kernel: usb 1-6: new high speed USB device using ehci_hcd and address 3 May 18 14:19:22 tenora1 kernel: usb 1-6: configuration #1 chosen from 1 choice May 18 14:19:22 tenora1 'xpp_fxloader'[15168]: Trying to find what to do for product e4e4/1160/101, device /dev/bus/usb/001/003 May 18 14:19:22 tenora1 'xpp_fxloader'[15172]: Loading firmware '/usr/share/dahdi/USB_FW.hex' into '/dev/bus/usb/001/003' May 18 14:19:22 tenora1 'xpp_fxloader'[15174]: /usr/share/dahdi/USB_FW.hex: unable to open for input. May 18 14:19:22 tenora1 'xpp_fxloader'[15175]: fxload failed with status 254 It was not supposed to have the firmware files into one of the dahdi RPMs? If not - from where could we download them (as I understand there are several ones)? Please find below some information regarding our machine. Best regards, Ioan. === [tenora1 ~]# rpm -qa | grep dahdi | sort asterisk14-dahdi-1.4.40.1-1_centos5 dahdi-firmware-2.0.2-1_centos5 dahdi-firmware-hx8-2.06-1_centos5 dahdi-firmware-oct6114-064-1.05.01-1_centos5 dahdi-firmware-oct6114-128-1.05.01-1_centos5 dahdi-firmware-tc400m-MR6.12-1_centos5 dahdi-linux-2.4.1.2-1_centos5 dahdi-tools-2.4.0-2_centos5 kmod-dahdi-linux-2.4.1.2-1_centos5.2.6.18_238.9.1.el5 kmod-dahdi-linux-fwload-vpmadt032-2.4.1.2-1_centos5.2.6.18_238.9.1.el5 [tenora1 ~]# ls -rtl /usr/share/dahdi/ total 20 -rwxr-xr-x 1 root root 8406 Nov 23 23:49 xpp_fxloader -rwxr-xr-x 1 root root 2415 Nov 23 23:49 waitfor_xpds -rwxr-xr-x 1 root root 2881 Nov 23 23:49 astribank_hook [tenora1 tmp]# rpm -qf /usr/share/dahdi/xpp_fxloader dahdi-tools-2.4.0-2_centos5 === -- _ -- 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] Automatic dialing + SMS
Here's a script to call a bunch of numbers (list-of-numbers.txt), trigger a
new call every 10 seconds. Adjust for your needs:
-snip-
#!/bin/bash
for i in `cat list-of-numbers.txt`
do
echo /usr/sbin/asterisk -rx originate local/@from-local extension
$i@voipout
/usr/sbin/asterisk -rx originate local/@from-local extension
$i@voipout
sleep 10
done
-snip-
[voipout] looks like:
exten = _X.,1,Dial(SIP/${EXTEN}@provider)
[from-local] looks like:
exten = ,1,Answer
exten = ,2,Playback(something)
exten = ,3,Hangup
- Original Message -
From: gadgetron...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 18, 2011 10:56 AM
Subject: Re: [asterisk-users] Automatic dialing + SMS
Hello.
Does it mean Asterisk has no in-built applications for auto dialing. What
scripting language can easily and best be used for the AGI. Tell me more abt
the sms providers
Sent from my BlackBerry® smartphone from Vodafone
-Original Message-
From: Enrico Cicconi enrico.cicc...@rdmnet.it
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 18 May 2011 09:54:45
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: bilal ghayyadbilmar...@yahoo.com
Subject: Re: [asterisk-users] Automatic dialing + SMS
Hi Bilal,
sure is possible as is possible to do other activities after played the
messages such call redirect, dtmf selection and so on.
About sending sms you can do it but our tips is to use an external SMS
gateway in your area. Where are you ?
If needed we can support, bye.
Enrico
www.rdmnet.it
Il 17/05/2011 10:43, bilal ghayyad ha scritto:
Hello All;
If I need the Asterisk to do automatic dialing for a list of numbers and
when the destination answer, then to play the proper sound message, is it
possible? How?
About sending SMS, can asterisk do this?
Regards
Bilal
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Re: [asterisk-users] Automatic dialing + SMS
See: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of gadgetron...@gmail.com Sent: Wednesday, May 18, 2011 4:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Automatic dialing + SMS Hello. Does it mean Asterisk has no in-built applications for auto dialing. What scripting language can easily and best be used for the AGI. Tell me more abt the sms providers Sent from my BlackBerry(r) smartphone from Vodafone -- _ -- 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 know how many calls are on hold
Can you send the logs in cli console for help you? Regards On Tue, May 17, 2011 at 9:16 AM, virendra ban hati virbh...@gmail.comwrote: hi list, please help me how to know how many calls are on hold. -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] Automatic dialing + SMS
For what reason do you need sending a sms from asterisk, please?
Am 18.05.2011 14:25, schrieb Markus:
Here's a script to call a bunch of numbers (list-of-numbers.txt),
trigger a
new call every 10 seconds. Adjust for your needs:
-snip-
#!/bin/bash
for i in `cat list-of-numbers.txt`
do
echo /usr/sbin/asterisk -rx originate local/@from-local extension
$i@voipout
/usr/sbin/asterisk -rx originate local/@from-local extension
$i@voipout
sleep 10
done
-snip-
[voipout] looks like:
exten = _X.,1,Dial(SIP/${EXTEN}@provider)
[from-local] looks like:
exten = ,1,Answer
exten = ,2,Playback(something)
exten = ,3,Hangup
- Original Message - From: gadgetron...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 18, 2011 10:56 AM
Subject: Re: [asterisk-users] Automatic dialing + SMS
Hello.
Does it mean Asterisk has no in-built applications for auto dialing. What
scripting language can easily and best be used for the AGI. Tell me
more abt
the sms providers
Sent from my BlackBerry® smartphone from Vodafone
-Original Message-
From: Enrico Cicconi enrico.cicc...@rdmnet.it
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 18 May 2011 09:54:45
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: bilal ghayyadbilmar...@yahoo.com
Subject: Re: [asterisk-users] Automatic dialing + SMS
Hi Bilal,
sure is possible as is possible to do other activities after played the
messages such call redirect, dtmf selection and so on.
About sending sms you can do it but our tips is to use an external SMS
gateway in your area. Where are you ?
If needed we can support, bye.
Enrico
www.rdmnet.it
Il 17/05/2011 10:43, bilal ghayyad ha scritto:
Hello All;
If I need the Asterisk to do automatic dialing for a list of numbers and
when the destination answer, then to play the proper sound message,
is it
possible? How?
About sending SMS, can asterisk do this?
Regards
Bilal
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Re: [asterisk-users] Automatic dialing + SMS
On Wed, 18 May 2011, gadgetron...@gmail.com wrote: Does it mean Asterisk has no in-built applications for auto dialing. Asterisk is a telephony Erector Set*. You get to build what you want. All the pieces are there. What scripting language can easily and best be used for the AGI. Easy may not be best. 'Easiest' is the language you know best. Best depends on your needs. A scripting language like PHP may be easiest for you if you know that language. A compiled language like C may be best if you want to run a bazillion calls per second. You can execute xxx AGIs written in C in the time it takes to load the Perl or PHP interpreter and parse your script. *) http://en.wikipedia.org/wiki/Erector_set -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Failover trunks
Hi All, Could you please help me with my following Scenario. I have a softswitch where my carriers send calls from International to my country for local termination. I route these calls to my Asterisk 1.8 which has a number of registered trunks from our SIP Provider. Please guide me how should I configure extensions.conf for calls to be sent to the next available trunk. Please help. Regards-Abid Saleem -- _ -- 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] Failover trunks
On 05/18/2011 10:05 AM, Abid Saleem wrote:
Could you please help me with my following Scenario. I have a
softswitch where my carriers send calls from International to my
country for local termination. I route these calls to my Asterisk 1.8
which has a number of registered trunks from our SIP Provider. Please
guide me how should I configure extensions.conf for calls to be sent
to the next available trunk.
exten = ...,1,Dial(SIP/${EXTEN}@trunk1,...)
exten = ...,n,Dial(SIP/${EXTEN}@trunk2,...)
exten = ...,n,Dial(SIP/${EXTEN}@trunk3,...)
?
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[asterisk-users] how to set moh without setting queue music with SetMusicOnHold
Hi all, I want to change my moh without changing my queue music...is it possible? SetMusicOnHold changes my moh but with the wrong effect to change my queue music I do not want to change...did anybody solve this problem or it is a bug? Thank you Giorgio Incantalupo -- _ -- 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 set moh without setting queue music with SetMusicOnHold
Giorgio, On 05/18/2011 10:32 AM, gincantalupo wrote: I want to change my moh without changing my queue music...is it possible? SetMusicOnHold changes my moh but with the wrong effect to change my queue music I do not want to change...did anybody solve this problem or it is a bug? Just define different MOH classes in musiconhold.conf. Refer to one in queues.conf and another one with 'musicclass' in sip.conf (general or peer). -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] how to set moh without setting queue music with SetMusicOnHold
Hi Alex, I tried but doesn't work because the queue doesn't give you control to change the moh. What I want is to change my moh depending on where the call is from. If it comes from Italy I have to play italian moh, if not, another moh. Normally I can change my moh but with queues, control is not in my hands. What I need is SetMusicOnHold to change moh only, not queues music which should be changeable with a sort of SetQueueMusic command or similar but this command does not exist. Giorgio Incantalupo On 05/18/2011 04:34 PM, Alex Balashov wrote: Giorgio, On 05/18/2011 10:32 AM, gincantalupo wrote: I want to change my moh without changing my queue music...is it possible? SetMusicOnHold changes my moh but with the wrong effect to change my queue music I do not want to change...did anybody solve this problem or it is a bug? Just define different MOH classes in musiconhold.conf. Refer to one in queues.conf and another one with 'musicclass' in sip.conf (general or peer). -- _ -- 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 set moh without setting queue music with SetMusicOnHold
On 05/18/2011 11:12 AM, gincantalupo wrote: I tried but doesn't work because the queue doesn't give you control to change the moh. What I want is to change my moh depending on where the call is from. If it comes from Italy I have to play italian moh, if not, another moh. Normally I can change my moh but with queues, control is not in my hands. What I need is SetMusicOnHold to change moh only, not queues music which should be changeable with a sort of SetQueueMusic command or similar but this command does not exist. Oh, I see. You are right; as far as I know, it is not possible to dynamically alter queue MOH from the dial plan. What is the feasibility of using multiple queues with the same members in them for various language/MOH combinations? If you need a unitary queue with this characteristic, your only choice might be to reimplement the queue logic in AGI with custom audio feedback. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] Jabber / facebook chat?
On 05/17/2011 07:18 AM, Stefan Gofferje wrote: On 04/17/2011 02:13 AM, Stefan Gofferje wrote: has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. I finally figured it out. For facebook chat to work you have to use usetls = no usesasl = yes Chat.facebook.com offers TLS but it seems, it's incompatible to res_jabber. To clarify, does that mean that you were able to successfully use facebook chat with sasl? -- _ -- 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 set moh without setting queue music with SetMusicOnHold
Hi Alex, I could create 2 queues, one for italians and one for strangers calling but there is no point where you can change the moh except before executing the queue command but the queue moh changes as side-effect: -AGI (check language) -SetMusicOnHold(depends on language) -Queue(depends on language) This would work if the above mentioned side-effect didn't exist. Re-implement a queue would take us a lot of time we unfortunately have not. I think the only solution is to give up. :( Thank you. Giorgio Incantalupo On 05/18/2011 05:15 PM, Alex Balashov wrote: On 05/18/2011 11:12 AM, gincantalupo wrote: I tried but doesn't work because the queue doesn't give you control to change the moh. What I want is to change my moh depending on where the call is from. If it comes from Italy I have to play italian moh, if not, another moh. Normally I can change my moh but with queues, control is not in my hands. What I need is SetMusicOnHold to change moh only, not queues music which should be changeable with a sort of SetQueueMusic command or similar but this command does not exist. Oh, I see. You are right; as far as I know, it is not possible to dynamically alter queue MOH from the dial plan. What is the feasibility of using multiple queues with the same members in them for various language/MOH combinations? If you need a unitary queue with this characteristic, your only choice might be to reimplement the queue logic in AGI with custom audio feedback. -- _ -- 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 set moh without setting queue music with SetMusicOnHold
On 05/18/2011 11:34 AM, gincantalupo wrote: I could create 2 queues, one for italians and one for strangers calling but there is no point where you can change the moh except before executing the queue command but the queue moh changes as side-effect: Hmm. When you use SetMusicOnHold, does it change the queue MOH only for the particular member/channel that joins it, or for the entire queue globally, for everyone already in the queue, etc? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] Jabber / facebook chat?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 05/18/2011 06:23 PM, Jason Parker wrote: To clarify, does that mean that you were able to successfully use facebook chat with sasl? This is correct. - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk3UGSwACgkQbQKZlCdPOMMj7QCdGZpt3CZEN6rP6sKBAxz2CcsM FnsAn1/Duexn+Seb3GaIcQ17L2Po7ELA =Ozxv -END PGP SIGNATURE- -- _ -- 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] asterisk18 - realtime/mysql - take 3
Still a couple of questions.. I did configure extconfig.conf ... ;iaxusers = odbc,asterisk ;iaxpeers = odbc,asterisk ;sipusers = odbc,asterisk sipusers = mysql,asterisk,sip_devices sippeers = mysql,asterisk,sip_devices ;sippeers = odbc,asterisk ;sipregs = odbc,asterisk ;voicemail = odbc,asterisk ;extensions = odbc,asterisk ;meetme = mysql,general ;queues = odbc,asterisk ;queue_members = odbc,asterisk ;musiconhold = mysql,general ;queue_log = mysql,general So only defining sipusers sippeers for mysql And noticed two files for configuring mysql-stuff: file: res_config_mysql.conf database access config: host, user, pwd file: res_odbc.conf in section [mysql2]: mysql database config: host, user, pwd So, i configured them both... Quick check:kc3054*CLI sip show users Username Secret Accountcode Def.Context ACL ForcerPort j.witvliet geheim default No Yes 0277611 25b06d3a0b5ef73default No Yes kc3054*CLI kc3054*CLI sip show peers Name/username Host Dyn Forcerport ACL Port Status Realtime 0277611 (Unspecified) D N 0Unmonitored j.witvliet (Unspecified) D N 0Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] kc3054*CLI Here i see at for both users and peers ONLY the statis entries from sip.conf file, nothing from mysql... kc3054*CLI realtime mysql status general connected to asterisk@127.0.0.1, port 3306 with username voipadmin for 5 seconds. kc3054*CLI =No warnings/errors but nothing else either... kc3054*CLI kc3054*CLI realtime mysql cache kc3054*CLI =No warnings/errors but nothing else either... the module res_config_mysql.so is loaded, Try todo something else: kc3054*CLI realtime update sipusers set SET port = 4343 WHERE name = 0277611 Failed to update. Check the debug log for possible SQL related entries. Command 'realtime update sipusers set SET port = 4343 WHERE name = 0277611' failed. [May 18 18:47:16] WARNING[16718]: res_config_mysql.c:559 update_mysql: MySQL RealTime: Invalid database specified: 'asterisk' (check res_mysql.conf) kc3054*CLI == here the system talkes about _another_ config file! == So which file should i configure: A) res_config_mysql.conf B) res_odbc.conf C) res_mysql.conf But even when i put my credentials in all three of them, still no show! DB check: mysql -h localhost -u voipadmin -p Enter password: Server version: 5.0.67 SUSE MySQL RPM mysql use asterisk; select name,username,secret,host,nat from sip_devices; Database changed +-+-++-+-+ | name| username| secret | host| nat | +-+-++-+-+ | 0031756 | 0031756 | geheim | dynamic | Yes | +-+-++-+-+ 1 row in set (0.00 sec) mysql According to *,TDG, page 349: Also filled the file /etc/unixODBC/odbcinst.ini, and the command odbcinst -q -d produced the required result: [MySQL] I presume i made a silly mistake/omission, but i fail to see how i can detect that, or other steps to test the correct configuration of ARA. So it looks that i'm stuck. Can not imagine that i'm the first here! But even from the definitive guide, chapter 16 and onwards, it isn't clear if you should use the mysql-stuff directly of through the odbc-routines Kind regards, Hans -- _ -- 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] When someone helps you, at least let them know if the problem is resolved or not
Because poking the sleep tiger is fun for some, especially if you're just BARELY faster than the tiger ;-) *poke* On Tue, May 17, 2011 at 5:26 AM, Andrew Thomas a...@datavox.co.uk wrote: And why would you post a reply 5 days after my last post - and 4 days after the threads last one? Do you want to keep this thread going? I suggest letting it die on it's own. _ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: 17 May 2011 02:05 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not Seriously guys. Why would anyone other than the two of you need to read this. It's a personal conversation. We all know who you both are and your achievements etc. The longer the conversation goes on the more off topic it becomes :-) -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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 -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- 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's zombie processes
I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -- _ -- 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's zombie processes
Are you sure it's Asterisk creating the zombie processes, not the check_sip pinger in Nagios? Nagios is extremely bad with high throughput and concurrency, and check_sip is a wrapper around 'sipsak', which means it takes the full Timer T1 * 64 to time out if the Asterisk server is truly not available (about ~30-32 sec). On 05/18/2011 04:40 PM, vip killa wrote: I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -- _ -- 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 -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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's zombie processes
On Wed, 18 May 2011, vip killa wrote: I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. Asterisk creates threads, not processes. Trace back from the PPID of the zombies to see who created them -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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's zombie processes
Hi,
You can run the script below as an hourly cron. Works for me.
#!/bin/sh
# clean-up Asterisk zombies
# file clean_up.sh
# $Id: clean_up all dead parent processes
# use as cron task */30 * * * * root /usr/local/sbin/clean_up.sh
#
##LOG=/var/log/asterisk/agi-cleanup.log
date=`date +%d-%m-%Y_%Hh%Mm`
t1=`cat /proc/stat | grep btime | awk '{print $2}'`
t3=`date +%s`
echo $date - Asterisk Zombies Clean up started. $LOG
echo
for parent in `ps -ef | grep safe_asterisk | awk '$3 == '1'{print $2}'`
do
for ppid in `ps -ef | awk '$3 == '${parent}' { print $2 }'`
do
for i in `ps -ef | awk '$3 == '${ppid}' { print $2 }'`
do
t2=`cat /proc/$i/stat| awk '{print $22}'`
b=$(($t3-$t1));
c=$(($t2/100));
d=$((($b-$c)/60));
if [ $d -gt 30 ] ; then
kill -9 $i
echo Zombie found - killing $i ### $LOG
fi
done
done
done
exit
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, May 18, 2011 1:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk's zombie processes
On Wed, 18 May 2011, vip killa wrote:
I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of
too many zombie processes. I eventually had to disable the notification
for the alert but why does Asterisk create so many zombie processes,
I've see more than 30 at times and it generally stays in the 20s... just
seems unusual and wondering if it's harmful, thanks in advance.
Asterisk creates threads, not processes.
Trace back from the PPID of the zombies to see who created them
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
_
No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1325 / Virus Database: 1509/3645 - Release Date: 05/18/11
--
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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
Definitely, especially with a nice chianti and some fava beans...*slurping sound* On Wed, May 18, 2011 at 6:19 PM, Matt Riddell li...@venturevoip.com wrote: On 17/05/11 5:24 PM, Sherwood McGowan wrote: I like puppies Yeah, much more tasty than fully grown dogs :-P -- 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 -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- 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] v1.8.4: Extension Not found in Context?
On 11-05-18 08:01 PM, A E [Gmail] wrote: boxb*CLI dialplan show Test [ Context 'Test' created by 'pbx_config' ] '' = 1. Answer() [pbx_config] 2. Wait(2) [pbx_config] 3. Hangup() [pbx_config] -= 1 extension (3 priorities) in 1 context. =- But when the call comes into boxb from box a, on box b CLI I see the msg: NOTICE[1613]: chan_sip.c:21581 handle_request_invite: Call from 'boxA' to extension '' rejected because extension not found in context 'Test'. WHY?? Thanks :( Does the peer using 'boxA' dialplan include context 'Test'? -- 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
[asterisk-users] Pridialplan/ prilocaldialplan
Hi I'm beginner in list. I have doubts about the options pridialplan and prilocaldiaplan in chan_dahdi.conf. I interconnect the Asterisk with a Siemens PBX, but i saw that the changes in the file do not take effect in debug of the span or calling/called number. How to use this options? In that cases to use? Ps.: sorry for the english, i'm brazilian. Thanks -- Att, Rafael Saraiva -- _ -- 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] Pridialplan/ prilocaldialplan
Hello. To apply this settings you should restart dahdi (dahdi restart in CLI). About influence you could read here: http://markmail.org/message/rpd2aewiu2soostz On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote: Hi I'm beginner in list. I have doubts about the options pridialplan and prilocaldiaplan in chan_dahdi.conf. I interconnect the Asterisk with a Siemens PBX, but i saw that the changes in the file do not take effect in debug of the span or calling/called number. How to use this options? In that cases to use? Ps.: sorry for the english, i'm brazilian. Thanks -- Att, Rafael Saraiva -- _ -- 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
