[asterisk-users] Multiple Servers: Multiple Peers: call-limit
I would like to enforce call-limit across multiple servers. Is there
any way to pass a call-limit variable between servers 01 02, as shown
below? Use a global call-limit between multiple servers and peer
connections.
A -- 01 -- Z
A -- 02 -- Z
A is using round-robin to reach Z, but in the event that 01 or 02 fail,
I want the full call-limit available to A. The call-limit is only
applied between A and the middle servers. For the sake of discussion,
let's say call-limit=10 for both, and the total limit should also be 10.
Since my round-robin configuration will fall-back to the other server,
calls can reach a maximum of 20. Not a state I want to allow.
#server_a_extensions.conf
[SERVER01]
exten = _X.,1,NoOp(Use: First Server)
same = n,Dial(SIP/A-to-01-to-Z/${EXTEN})
same = n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?SERVER02,${EXTEN},1)
same = n,NoOp(yes, it's incomplete)
[SERVER02]
exten = _X.,1,NoOp(Use: Second Server)
same = n,Dial(SIP/A-to-02-to-Z/${EXTEN})
same = n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?SERVER01,${EXTEN},1)
same = n,NoOp(yes, it's incomplete)
I've though about passing the variable between the middle servers in a
SIP message, side communication channel. But, hoping there might be a
simpler solution.
Sincerely,
Brian LaVallee
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Re: [asterisk-users] T1 Card RED ALARM
Since there are a number of setting that could be causing the alarm, AMI/B8ZS, SF/ESF, etc... Start with a loop-test, make sure the card can communicate with itself (using the current settings). Connect the following pins: 01 (RX-) -- 04 (TX+) 02 (RX+) -- 05 (TX-) Sincerely, Brian LaVallee On 6/25/14, 1:16 PM, arun kumar wrote: Cables are workig fine in my other box. On 25 Jun 2014 00:46, Steve Totaro stot...@totarotechnologies.com wrote: Remember to always check your cables first. Thanks, Steve T On Tue, Jun 24, 2014 at 1:47 PM, arun kumar arunvsadni...@gmail.com wrote: Thank you Josh for your valuable reply. I will do try changing the server and let you know what happening. ~Arun On Tue, Jun 24, 2014 at 8:39 PM, Josh Metzger joshdmetz...@gmail.com wrote: On Tue, Jun 24, 2014 at 5:25 AM, arun kumar arunvsadni...@gmail.com wrote: Hello All, I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do connect T1 lines it goes in RED. When I do connect the same line on a different Server (Same Model T1 Card) it works fine. How do I examine/diagnose my T1 Card for any hardware failures. I heard about loopback test , how helpful it is? Here are my configuration /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured Thanks ~Arun It could still be some sort of system config issue, even if you think everything is configured the same. Have you tried moving the T1 card from the Bad system to the good system? That will at least help narrow down if it's a bad card / port, or a config issue. -Josh -- _ -- 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 -- _ -- 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] Multiple Servers: Multiple Peers: call-limit
Store the call count in a shared SQL db.
Sent from my Verizon Wireless 4G LTE DROID
Brian LaVallee b.laval...@globaltank.jp wrote:
I would like to enforce call-limit across multiple servers. Is there
any way to pass a call-limit variable between servers 01 02, as shown
below? Use a global call-limit between multiple servers and peer
connections.
A -- 01 -- Z
A -- 02 -- Z
A is using round-robin to reach Z, but in the event that 01 or 02 fail,
I want the full call-limit available to A. The call-limit is only
applied between A and the middle servers. For the sake of discussion,
let's say call-limit=10 for both, and the total limit should also be 10.
Since my round-robin configuration will fall-back to the other server,
calls can reach a maximum of 20. Not a state I want to allow.
#server_a_extensions.conf
[SERVER01]
exten = _X.,1,NoOp(Use: First Server)
same = n,Dial(SIP/A-to-01-to-Z/${EXTEN})
same = n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?SERVER02,${EXTEN},1)
same = n,NoOp(yes, it's incomplete)
[SERVER02]
exten = _X.,1,NoOp(Use: Second Server)
same = n,Dial(SIP/A-to-02-to-Z/${EXTEN})
same = n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?SERVER01,${EXTEN},1)
same = n,NoOp(yes, it's incomplete)
I've though about passing the variable between the middle servers in a
SIP message, side communication channel. But, hoping there might be a
simpler solution.
Sincerely,
Brian LaVallee
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Re: [asterisk-users] OT - How to configure Apache2 server to receive Polycom log files ?
Thanks for sharing this.
I'll give it a try ASAP and post my comments here.
Thanks again.
2014-06-17 14:51 GMT+02:00 Stepan Hradsky stepan.hrad...@ha-vel.cz:
Hi,
I have this configuration in apache site configuration
Directory /home/prov/polycom
RewriteEngine On
RewriteCond %{REQUEST_METHOD} =PUT [OR]
RewriteCond %{REQUEST_METHOD} =HEAD
RewriteRule ^(.*)$ put.php?url=$1
/Directory
this redirect PUT method to put.php script which read input and write to
file in logs directory.
and this is file put.php:
?php
/* PUT data comes in on the stdin stream */
$putdata = fopen(php://input, r);
$mode=w;
/* Open a file for writing */
$file=basename($_SERVER['REQUEST_URI']);
if (substr($file,-3)!='cfg' and substr($file,-3)!='xml') {
if (!is_dir(logs)) mkdir(logs);
$file=logs/$file;
$mode=a;
}
$fp = fopen($file, $mode);
/* Read the data 1 KB at a time and write to the file */
while ($data = fread($putdata, 1024)) fwrite($fp, $data);
/* Close the streams */
fclose($fp);
fclose($putdata);
?
BR
Stepan
Dne 16.6.2014 15:51, Olivier napsal(a):
Hello,
To troubleshoot Polycom phone provisionning (with an asterisk 11 box),
I'm looking to enable HTTP log file upload ie the capability for Polycom
phones to upload some data to a given HTTP server.
At the moment, Polycom phones are downloading config files from an
Apache2 HTTP server, thanks to a DHCP server configuration option bellow.
option tftp-server-name http://192.168.64.250/polycom;;
Looking at Apache2 log files, I can see that Polycom phones are trying to
upload log files but every attempt to upload data (fails with 405 error, no
matter how I configured target upload directory ownership. See:
192.168.64.215 - - [16/Jun/2014:15:25:03 +0200] PUT
/polycom/log/0004f2394356-boot.log HTTP/1.1 405 582 - FileTransport
PolycomSoundPointIP-SPIP_650-UA/4.3.0.0246
Has someone successfully received Polycom file uploads with an HTTP server
(ie without using FTP) or is it something can't simply be done ?
If positive, can you share key configuration settings ?
Regards
--
S pozdravem / with kind regards
Štěpán Hradský
oddělení specialistů hlasových služeb / voice department
ha-vel internet s.r.o.
Olešní 587/11A
712 00 Ostrava Muglinov
Czech Republic
T +420 552 305 370/ F +420 552 305 306
Hotline: +420 552 305 321http://www.ha-vel.cz
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[asterisk-users] Play announcement only once in a Call Queue after 10 seconds
Hello, how can I create the following scenario : I have a Call Queue and I want to play an announcement, but only once after about 10 seconds. The current option |periodic| |-| |announce| |-| |frequency| keeps on playing the announcement indefinitely. (it should have an option 'once' like the option |announce-holdtime|) 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] Multiple Servers: Multiple Peers: call-limit
Something like memcachedb is also an option.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gregory Malsack
Sent: Wednesday, June 25, 2014 5:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multiple Servers: Multiple Peers: call-limit
Store the call count in a shared SQL db.
Sent from my Verizon Wireless 4G LTE DROID
Brian LaVallee b.laval...@globaltank.jpmailto:b.laval...@globaltank.jp
wrote:
I would like to enforce call-limit across multiple servers. Is there
any way to pass a call-limit variable between servers 01 02, as shown
below? Use a global call-limit between multiple servers and peer
connections.
A -- 01 -- Z
A -- 02 -- Z
A is using round-robin to reach Z, but in the event that 01 or 02 fail,
I want the full call-limit available to A. The call-limit is only
applied between A and the middle servers. For the sake of discussion,
let's say call-limit=10 for both, and the total limit should also be 10.
Since my round-robin configuration will fall-back to the other server,
calls can reach a maximum of 20. Not a state I want to allow.
#server_a_extensions.conf
[SERVER01]
exten = _X.,1,NoOp(Use: First Server)
same = n,Dial(SIP/A-to-01-to-Z/${EXTEN})
same = n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?SERVER02,${EXTEN},1)
same = n,NoOp(yes, it's incomplete)
[SERVER02]
exten = _X.,1,NoOp(Use: Second Server)
same = n,Dial(SIP/A-to-02-to-Z/${EXTEN})
same = n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?SERVER01,${EXTEN},1)
same = n,NoOp(yes, it's incomplete)
I've though about passing the variable between the middle servers in a
SIP message, side communication channel. But, hoping there might be a
simpler solution.
Sincerely,
Brian LaVallee
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[asterisk-users] OPTIONS Request without username - Forbidden
Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is 403 Forbidden. Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt (pg 67), we found carol, so it makingme see that i am missing some config. OPTIONS sip:ca...@chicago.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877 Max-Forwards: 70 To: sip:ca...@chicago.com Is it wright? How can i instruct FREEPBX to send the username in the option request? Sorry for this silly question but a found no answer googling. Thans in advance. rv This is the debug of the case Reliably Transmitting (NAT) to 201.217.31.XX:5060: OPTIONS sip:201.217.31.10 SIP/2.0 Via: SIP/2.0/UDP 18x.16.204.XXX:6060;branch=z9hG4bK1d8715df;rport Max-Forwards: 70 From: Unknown sip:59x212376...@186.16.204.xxx:6060;tag=as4491c6af To: sip:201.217.31.10 Contact: sip:59x212376...@18x.16.204.xxx:6060 Call-ID: 4f02699e2632410c359e1ee43a021...@186.16.204.xxx:6060 CSeq: 102 OPTIONS User-Agent: FPBX-2.11.0(1.8.25.0) Date: Wed, 25 Jun 2014 13:47:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- SIP read from UDP:201.217.31.XX:5060 --- SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 18x.16.204.XXX:6060;received=18x.16.204.XXX;branch=z9hG4bK1d8715df;rport=5060 From: Unknown sip:59x212376...@18x.16.204.xxx:6060;tag=as4491c6af To: sip:201.217.31.XX;tag=aprqngfrt-nm50ea1c6 Call-ID: 4f02699e2632410c359e1ee43a021...@18x.16.204.xxx:6060 CSeq: 102 OPTIONS This is the peer. * Name : desde-XopaXo-2376XXX Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-trunk Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 201.217.31.10 Addr-IP : 201.217.31.10:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 595212376458 SIP Options : timer Codecs : 0xe (gsm|ulaw|alaw) Codec Order : (ulaw:20,alaw:20,gsm:20) Auto-Framing : No Status : OK (36 ms) Useragent: Reg. Contact : Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No * Name : desde-XopaXo-2376XXX Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-trunk Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 201.217.31.XX Addr-IP : 201.217.31.XX:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 59X212376XXX SIP Options : timer Codecs : 0xe (gsm|ulaw|alaw) Codec Order : (ulaw:20,alaw:20,gsm:20) Auto-Framing : No Status : OK (36 ms) Useragent: Reg. Contact : Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] RPM updates
On Mon, Jan 28, 2013 at 05:21:10PM +0200, Tzafrir Cohen wrote: On Mon, Jan 28, 2013 at 01:55:09PM +, Steven Howes wrote: Hi All, Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. See also: http://git.tzafrir.org.il/?p=rpm/dahdi-linux.git;a=summary http://git.tzafrir.org.il/?p=rpm/dahdi-tools.git;a=summary A new set of package repositories is now available under http://git.xorcom.com/ , or specifically: http://git.xorcom.com/rpm/ . The packages there are in initial stages of packaging and thus not yet published as a repository. I built them using git-buildpackage-rpm, see: http://git.xorcom.com/?p=rpm/tools.git;a=blob;f=README.txt -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.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
[asterisk-users] Echo Cancellation when calling from softphone to mobile.
Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- 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] Echo Cancellation when calling from softphone to mobile.
Put line side echo cancelation chip on ur PRI card. On 25-Jun-2014 10:35 PM, Anurag Rana anuragrana31...@gmail.com wrote: Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- 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] Echo Cancellation when calling from softphone to mobile.
Is there any Software solution? On Wed, Jun 25, 2014 at 11:38 PM, Mitul Limbani mi...@enterux.in wrote: Put line side echo cancelation chip on ur PRI card. On 25-Jun-2014 10:35 PM, Anurag Rana anuragrana31...@gmail.com wrote: Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- 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 -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- 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] Echo Cancellation when calling from softphone to mobile.
There are two common types of echo. Accoustic Echo: This is caused by microphone picking up audio from the speaker. This echo cannot generally be removed by echo cancelers. The solution to accoustic echo is to prevent the microphone from picking up audio from the speaker (or handset or earpiece). Line Echo: This is caused by your outgoing audio “reflecting” off the far end of an analog line. This happens in all calls with a 2-wire analog portion, however in analog and digital (aka PRI) the delay in echo is so small we can’t perceive it.VoIP has far larger latencies so we can hear the echo. This type of echo MUST be canceled out before the audio is converted to VoIP. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anurag Rana Sent: Wednesday, June 25, 2014 2:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Echo Cancellation when calling from softphone to mobile. Is there any Software solution? On Wed, Jun 25, 2014 at 11:38 PM, Mitul Limbani mi...@enterux.inmailto:mi...@enterux.in wrote: Put line side echo cancelation chip on ur PRI card. On 25-Jun-2014 10:35 PM, Anurag Rana anuragrana31...@gmail.commailto:anuragrana31...@gmail.com wrote: Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- 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 -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- 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 12 and chan_local
I am migrating my app to Asterisk12 and pjsip, but I cannot find chan_local, what happened? -- _ -- 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 12 and chan_local
On Wed, Jun 25, 2014 at 4:38 PM, CDR vene...@gmail.com wrote: I am migrating my app to Asterisk12 and pjsip, but I cannot find chan_local, what happened? from https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12 chan_local: The /b option has been removed. chan_local moved into the system core and is no longer a loadable module. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Play announcement only once in a Call Queue after 10 seconds
Hi Jonas, While I don't work with queues, but you could playback announce-holdtime before putting the caller into the queue. exten = _X.,1,NoOp(Post Queue Announcement) same = n,Answer() same = n,Wait(10) same = n,Playback(announce-holdtime) same = n,Queue(real_queue) Brian On 6/25/14, 10:11 PM, Jonas Kellens wrote: Hello, how can I create the following scenario : I have a Call Queue and I want to play an announcement, but only once after about 10 seconds. The current option |periodic| |-| |announce| |-| |frequency| keeps on playing the announcement indefinitely. (it should have an option 'once' like the option |announce-holdtime|) 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] PJSIP Dial via IP fails
Dear friends
This is my simple dialplan
[demopjsip]
exten = _X.,1,Dial(PJSIP/${EXTEN}@10.10.10.2)
exten = _X.,n,Hangup()
I need to dial out via an IP address, not using an endpoint, as shown above.
It fails with
Executing [1957408@demopjsip:3] Dial(PJSIP/federico-0002,
PJSIP/195XXX7408@10.10.10.2) in new stack
[Jun 26 00:39:00] ERROR[10136]: chan_pjsip.c:1722 request: Unable to
create PJSIP channel - endpoint '10.10.10.2' was not found
[Jun 26 00:39:00] WARNING[10167][C-0002]: app_dial.c:2421
dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No
route to destination)
I remember that this Dial format was possible with regular SIP. The IP
address is routable, so there is no specific network issue.
In my pjsip.coonf I defined a default outbound endpoint
[global]
default_outbound_endpoint=default_outbound_endpoint
In that default endpoint defined, I did not add any IP address,
because I want to keep it generic and dial any IP address with the
same settings,
Is this possible?
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