[asterisk-users] Queues and Distinctive Ring with Alert-Info
Hi All, I'm new to Queues and I have created one as follows which seems to work ok. [david-test] strategy = rrmemory timeout = 10 retry = 0 maxlen = 0 announce-frequency = 0 announce-holdtime = no member => SIP/121 member => SIP/122 member => SIP/123 I'm wondering how do you change the SipAddHeader/Alert-Info when a call comes from a queue so users know it is a queue that is calling? Is something like the following supposed to work? exten => 0453451564,1,SipAddHeader(Alert-Info: n=Classic-4;w=3;c=4) exten => 0453451564,2,Queue(david-test) Regards David Klaverstyn -- _ -- 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 on short network
I am running asterisk 1.4.43 on a really small network for testing, all on same switch. I launch a meetme between my server and 5 asterisk clients that are all on 10 foot network cables all connected to the same switch. The meetme is fine everything is in sync Then I reboot one of the clients. When it reboots I automatcially bring it back into the conference. however now its not really "in sync". I'm trying to understand why that might be??? I thought it would. The conference is a listen only conference. Its not "off" or out of sync by much - but it is noticable. Is there anything I can do to make that audio more in sync all the time? Thanks, Jerry -- _ -- 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] * Waiting for asterisk to shutdown .............
On 11/25/12 10:53, Mikhail Lischuk wrote: Joseph wrote 25.11.2012 02:12: ebegin "Stopping asterisk PBX gracefully" /usr/sbin/asterisk -r -x "core stop gracefully" &>/dev/null # Now we have to wait until asterisk has _really_ stopped. As you can see, it is trying to execute "core stop gracefully". I don't know why it wont shutdown, or maybe it would but you can't wait till it happens, so I can't help in troubleshooting. But if you need a quick workaround to be able to shut down without killing process - change it to "core stop now". It should shut the Asterisk down immediately, dropping all active calls. -- With Best Regards [1]Mikhail Lischuk Good suggestion, thank you. I've modified the scrip and will monitor it for a week. -- Joseph -- _ -- 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] Queue_log into MySQL - best practices
On Sunday 25 Nov 2012, Dmitry wrote: > [snip] > 3) Still know nothing about odbc support for queue_log Happily using ODBC (PostgreSQL, but should be mostly DB-independent) for queue_log here. The setup is a bit hairy, but can share if enough people show interest. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Java server side components
Thanks Rudi, but I want a java server component for building asterisk GUI. Nweike Onwuyali . excuse my typos & brevity. Sent from a mobile device On Nov 23, 2012, at 9:18 PM, "Rudi Lee" wrote: > You might want to look at Asterisk-Java, Java library to communicate to > Asterisk via AMI and AGI, https://github.com/srt/asterisk-java > > Best Regards, > > Rudi Lee > > -Original Message- > From: Nweike Onwuyali > Sender: asterisk-users-boun...@lists.digium.com > Date: Thu, 22 Nov 2012 19:45:36 > To: Asterisk Users Mailing List - Non-Commercial > Discussion > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [asterisk-users] Java server side components > > -- > _ > -- 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] Queue_log into MySQL - best practices
Lenz, Thank you for your answer Here is what I think: 1) If I use an external Perl/Python daemon to parse queue_log and put it into the MYSQL database - I must also monitor this script (for example,to restart it in case of failure) although this external daemon is rather easy to maintain and it means an "independence" from the DEPRECATED app_mysql. 2)If I use the app_mysql module- although it is very convenient (asterisk developers maintain it) but as it has a DEPRECATED status - I am afraid using this module (I am not sure about its future). 3) Still know nothing about odbc support for queue_log So I chose 1). From: Lenz Emilitri To: Dmitry ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 22, 2012 3:36 PM Subject: Re: [asterisk-users] Queue_log into MySQL - best practices Hi Dmitry, we usually advise against writing queue_log events straight to a database, as it is marginally more likely that the DB has issues that a simple flat file. And when data is lost it's lost forever. Still everybody seems to love writing data straight to the DB :) l. 2012/11/22 Dmitry Hi, > > >I use asterisk 1.8. > Currently I use a perl daemon to parse queue_log into MySQL. It works >reliably. > > >But I know that there is a method >(http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL and >http://work.mikeboylan.com/asterisk-queuelog-to-mysql) to write to MySQL >directly with app_mysql which has a DEPRECATED status. > > >My question is: >What is the best/preffered approach to put queue_log into MySQL in asterisk >1.8 and up? >1) To use external daemons to parse /var/log/queue_log? >2) To use the deprecated app_mysql? the status does not guarantee that this >application will be in the future >3) To use odbc to access mysql? but I could not find a procedure for it. And I >doubt it is possible. > > >BR, >Dmitry Pavlenko -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] SIP and RTP on different IP's
Am 25.11.2012 15:41, schrieb Tiago Geada: yes I have no control over that. Ok we will figure another way. Thanks How about two Asterisk instances with different bindaddr= settings? One instance listens on 172.16.1.10 and the other instance listens on 10.34.18.250. Maybe this will make the RTP IP in SDP automatically right? But I'm just guessing... Here are some instructions for multiple instances: http://forums.asterisk.org/viewtopic.php?f=1&t=71510 Regards Markus -- _ -- 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 and RTP on different IP's
yes I have no control over that. Ok we will figure another way. Thanks On 25 November 2012 07:10, Duncan Turnbull wrote: > > > On 25/11/2012, at 1:23 PM, Tiago Geada wrote: > > linux does sort this out and asterisk listens in both interfaces. however > asterisk connects and tells remote end to send rtp back at the same IP > where sip is going trough... > > remote end does try to send it but gets stopped in a firewall.. thus if > asterisk did present a different IP to recieve RTP in its SIP header, this > would not happen! > > > > I think this is outside of asterisk's natural ability > > You may need a proxy server in between you and the Cisco to achieve this > if you can't change the firewall. > > http://forums.asterisk.org/viewtopic.php?f=1&t=84018 > > Have you tried making the preferred route to these addresses go out eth1, > thus getting the required address? > > Ultimately seems odd the firewall allows access in but not out, guessing > you have no control over that? > > Good luck > > Cheers Duncan > > > On 23 November 2012 19:39, Duncan Turnbull wrote: > >> >> On 24/11/2012, at 2:19 AM, Tiago Geada wrote: >> >> Hello Folks, I am looking for a way that makes asterisk tell remote SIP >> party that the IP where they will send RTP is not the same as the one I am >> comunicating via SIP >> >> Can this be done anyhow? >> >> I can try and explain: >> >> We have placed a asterisk box in our partners office. >> >> It has eth0 with IP 172.16.1.10 and eth1: 10.34.18.250 >> >> linux has its routes set so it can comunicate with several networks in >> their offices. >> >> now there is a cisco call manager that we need to communicate with. >> Normally via our IP 172.16.1.10, however seems that this cisco uses some >> sort of 'directmedya=yes' and sets both ends speaking RTP with themselves. >> >> There are some extensions in cisco that have a network 10.134.0.0/16that we >> can only comunicate via eth1 >> >> thus when calling cisco (always via eth0) sometimes we need to say that >> OUR IP to recieve RTP is not 172.16.1.10, but 10.34.18.250 >> >> >> This is a routing issue, not asterisk I think. You are saying you route >> to cisco via eth0, it sets up connections to its end points and then drops >> out of the media flow, but the end points have no route to the eth0 address >> so they fail >> >> Linux usually sorts this out and asterisk replies on the address of the >> interface it sends out with. So for the most part the response in my >> experience if its going out eth1 should use the eth1 ip address. >> >> If you can get to it via eth0 and thats the preferred route then it will >> have the eth0 address. If so why can't you change your routing table to use >> eth1 when you need to go to the cisco then you will have the right address >> and the far extensions can respond to you correctly >> >> Or change the cisco network endpoints so they can successfully access >> your address on eth0 >> >> >> can this be done? >> -- >> _ >> -- 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 > -- _ -- 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] * Waiting for asterisk to shutdown .............
Joseph wrote 25.11.2012 02:12: > ebegin "Stopping asterisk PBX gracefully" > /usr/sbin/asterisk -r -x "core stop gracefully" &>/dev/null > # Now we have to wait until asterisk has _really_ stopped. As you can see, it is trying to execute "core stop gracefully". I don't know why it wont shutdown, or maybe it would but you can't wait till it happens, so I can't help in troubleshooting. But if you need a quick workaround to be able to shut down without killing process - change it to "core stop now". It should shut the Asterisk down immediately, dropping all active calls. -- With Best Regards Mikhail Lischuk -- _ -- 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
