Re: [asterisk-users] set time zone in sip debug logs
Hello Qasim, I need to change it permanently. System date/time is correct. INVITE header always follows GMT irrespective of system's date/time zone. It would be nice if you can mention the steps to sync the system and INVITE header time permanently. Thanks,Kamlesh Date: Tue, 26 Feb 2013 12:30:55 +0500 From: qasimak...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] set time zone in sip debug logs Hi Kamlesh, Asterisk give you very less control over SIP messaging. You can how ever add/remove/modify SIP headers from initial invite only. To modify a sip header you can use asterisk function "SIP_HEADER()". If you want to permanently change date why not change system date/time? Regards, -Qasim On Tue, Feb 26, 2013 at 11:13 AM, Kamlesh Kumar wrote: Hello, Please suggest the way to change the time zone in below sip debug logs. INVITE sip:xxx...@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rport Max-Forwards: 70 From: "xx" ;tag=as23a29r59 To: Contact: Call-ID: 2f17b2103ea4792d571e2dce7e14b...@xxx.xxx.xxx.xxx CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9 Date: Tue, 26 Feb 2013 04:54:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 444 Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] set time zone in sip debug logs
Hi Kamlesh, Asterisk give you very less control over SIP messaging. You can how ever add/remove/modify SIP headers from initial invite only. To modify a sip header you can use asterisk function "*SIP_HEADER()*". If you want to permanently change date why not change system date/time? Regards, -Qasim On Tue, Feb 26, 2013 at 11:13 AM, Kamlesh Kumar wrote: > Hello, > > Please suggest the way to change the time zone in below sip debug logs. > > INVITE sip:xxx...@xxx.xxx.xxx.xxx:5060 SIP/2.0 > Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rport > Max-Forwards: 70 > From: "xx" ;tag=as23a29r59 > To: > Contact: > Call-ID: 2f17b2103ea4792d571e2dce7e14b...@xxx.xxx.xxx.xxx > CSeq: 102 INVITE > User-Agent: Asterisk PBX 1.6.2.9 > *Date: Tue, 26 Feb 2013 04:54:29 GMT* > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 444 > > Thanks, > Kamlesh > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] set time zone in sip debug logs
Hello, Please suggest the way to change the time zone in below sip debug logs. INVITE sip:xxx...@xxx.xxx.xxx.xxx:5060 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rportMax-Forwards: 70From: "xx" ;tag=as23a29r59To: Contact: Call-ID: 2f17b2103ea4792d571e2dce7e14b...@xxx.xxx.xxx.xxxCSeq: 102 INVITEUser-Agent: Asterisk PBX 1.6.2.9Date: Tue, 26 Feb 2013 04:54:29 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Type: application/sdpContent-Length: 444 Thanks,Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk AMI - Create a daemon (background process)
On 13-02-24 07:30 AM, Shahid H wrote: I wanted to create a daemon (background process) in PHP. A daemon will use socket to connect with Asterisk AMI to send events and listen the actions. A daemon will also listen the commands from agents via HTTP, for example: A agent pressed a hang up button on a browser - it will send http command to a daemon. A daemon received a command and will then send Hang Up Action to AMI. How should a daemon process be designed to listen multiple actions and events? For example: 50 agents currently on the calls and how should a daemon to monitor the Actions/Events from 50 agents? You don't want to use PHP for your daemon, change to another scripting language (EG: python). -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Calendar: cert mismatch
> "PD" == Phil Daws writes: PD> It does generate a validity warning, as its self-signed, though I have PD> added it to the PBX ca-bundle.crt. Am I right in assuming that PD> Asterisk will use the default OpenSSL paths for where certificates are PD> stored ? The error said that the hostname in the uri does not match (any of) the hostname(s) in the cert. Does the self-signed cert have the hostname in either the CN or in (any of) the dnsName(s) in the subjectAltName section? It might work better if you created a local CA and used that to sign an end-entity cert for each server which needs one. Then add that CA cert to the bundle. Recent versions of tls (claim to have) deprecated the idea of using self-signed certs for anything other than root ca certs, but you can always create your own CA. -JimC -- James Cloos OpenPGP: 1024D/ED7DAEA6 -- _ -- 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] DST offset
ok, so after digging through google and the wiki and not finding anything I
went through the code for res_phoneprov and found my answer. in users.conf
set timezone= to the full name of the time zone.
-Bryan Anderson
On Mon, Feb 25, 2013 at 1:35 PM, Bryan Anderson wrote:
> Sorry I incorrectly typed my email. Let me correct, dstoffset should be
> tzoffset.
>
>
> Hello,
>
> I am trying to set the tcpIpApp.sntp.gmtOffset="0" setting on the polycom
> phone provisioning template and I see that it has a variable for
> ${TZOFFSET} in the template I tried adding "tzoffset = -28800" to
> users.conf and sip.conf under both general and individual users but can't
> get the setting to set. Where/How do I set the variable?
>
> Thanks,
> Bryan
>
> On Mon, Feb 25, 2013 at 10:44 AM, Bryan Anderson wrote:
>
>> Hello,
>>
>> I am trying to set the tcpIpApp.sntp.gmtOffset="0" setting on the
>> polycom phone provisioning template and I see that it has a variable for
>> ${DSTOFFSET} in the template I tried adding "dstoffset = -28800" to
>> users.conf and sip.conf under both general and individual users but can't
>> get the setting to set. Where/How do I set the variable?
>>
>> Thanks,
>> Bryan
>>
>
>
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Re: [asterisk-users] Calendar: cert mismatch
It does generate a validity warning, as its self-signed, though I have added it to the PBX ca-bundle.crt. Am I right in assuming that Asterisk will use the default OpenSSL paths for where certificates are stored ? - Original Message - From: "Christopher Harrington" To: "Phil Daws" , "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, 25 February, 2013 9:23:56 PM Subject: Re: [asterisk-users] Calendar: cert mismatch On Mon, Feb 25, 2013 at 3:18 PM, Phil Daws < ux...@splatnix.net > wrote: Hi, Am testing out the iCal functionality but when changing the URL am faced with the following warning: [Feb 25 20:55:20] WARNING[6234] res_calendar_icalendar.c: Unable to retrieve iCalendar 'dummycal' from ' https://webmail.domain.com/home/u...@domain.com/Calendar/ ': Server certificate verification failed: certificate issued for a different hostname, issuer is not trusted Has a cert been cached somewhere ? Have you actually verified that visiting https://[whatever] doesn't generate a certificate validity warning in your browser? -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] DST offset
Sorry I incorrectly typed my email. Let me correct, dstoffset should be
tzoffset.
Hello,
I am trying to set the tcpIpApp.sntp.gmtOffset="0" setting on the polycom
phone provisioning template and I see that it has a variable for
${TZOFFSET} in the template I tried adding "tzoffset = -28800" to
users.conf and sip.conf under both general and individual users but can't
get the setting to set. Where/How do I set the variable?
Thanks,
Bryan
On Mon, Feb 25, 2013 at 10:44 AM, Bryan Anderson wrote:
> Hello,
>
> I am trying to set the tcpIpApp.sntp.gmtOffset="0" setting on the polycom
> phone provisioning template and I see that it has a variable for
> ${DSTOFFSET} in the template I tried adding "dstoffset = -28800" to
> users.conf and sip.conf under both general and individual users but can't
> get the setting to set. Where/How do I set the variable?
>
> Thanks,
> Bryan
>
--
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Re: [asterisk-users] Calendar: cert mismatch
On Mon, Feb 25, 2013 at 3:18 PM, Phil Daws wrote: > Hi, > > Am testing out the iCal functionality but when changing the URL am faced > with the following warning: > > [Feb 25 20:55:20] WARNING[6234] res_calendar_icalendar.c: Unable to > retrieve iCalendar 'dummycal' from ' > https://webmail.domain.com/home/u...@domain.com/Calendar/': Server > certificate verification failed: certificate issued for a different > hostname, issuer is not trusted > > Has a cert been cached somewhere ? > > Have you actually verified that visiting https://[whatever] doesn't generate a certificate validity warning in your browser? -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Calendar: cert mismatch
Hi, Am testing out the iCal functionality but when changing the URL am faced with the following warning: [Feb 25 20:55:20] WARNING[6234] res_calendar_icalendar.c: Unable to retrieve iCalendar 'dummycal' from 'https://webmail.domain.com/home/u...@domain.com/Calendar/': Server certificate verification failed: certificate issued for a different hostname, issuer is not trusted Has a cert been cached somewhere ? 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
[asterisk-users] DST offset
Hello,
I am trying to set the tcpIpApp.sntp.gmtOffset="0" setting on the polycom
phone provisioning template and I see that it has a variable for
${DSTOFFSET} in the template I tried adding "dstoffset = -28800" to
users.conf and sip.conf under both general and individual users but can't
get the setting to set. Where/How do I set the variable?
Thanks,
Bryan
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Re: [asterisk-users] Asterisk AMI - Create a daemon (background process)
El 24/02/13 07:30, Shahid H escribió: I wanted to create a daemon (background process) in PHP. A daemon will use socket to connect with Asterisk AMI to send events and listen the actions. A daemon will also listen the commands from agents via HTTP, for example: A agent pressed a hang up button on a browser - it will send http command to a daemon. A daemon received a command and will then send Hang Up Action to AMI. How should a daemon process be designed to listen multiple actions and events? For example: 50 agents currently on the calls and how should a daemon to monitor the Actions/Events from 50 agents? You may want to take a look to the Elastix CallCenter code. This project has a daemon component to implement the autodialer, written in PHP. This code already solves the AMI connection issue and the multiple agent issue. http://elastix.svn.sourceforge.net/viewvc/elastix/trunk/apps/extras/callcenter/ -- _ -- 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] auto install all required dependences for asterisk.
Hello Mahendra, I've just installed Asterisk from source on my Raspberry Pi model B, this is what I did: sudo apt-get install build-essential sudo apt-get install libncurses5-dev sudo apt-get install libssl-dev sudo apt-get install libxml2-dev cd /usr/src/ sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8-current.tar.gz sudo tar zxvf asterisk-1.8-current.tar.gz sudo reboot sudo ./configure sudo make menuselect sudo make sudo make install sudo make samples sudo make config <-- It doesn't work for me I can access Asterisk from console (asterisk -vvvc) but I'm struggling to put it as a linux service. Hope this helps, if you have an answer to put asterisk as service please let me know. Elder D. Arohuanca Lima - Peru On Fri, Feb 15, 2013 at 3:13 PM, Christopher Harrington wrote: > On Fri, Feb 15, 2013 at 1:50 PM, Mahendra Dobariya < > mahendra_mahen...@hotmail.com> wrote: > >> but it did not works.. >> >> > Unfortunately, this is completely useless for anyone to help you. You will > need to actually tell us what isn't working and include output from your > console. > > -- > -Chris Harrington > ACSDi Office: 763.559.5800 > Mobile Phone: 612.326.4248 > > > -- > _ > -- 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] Fwd: Google Calendar issue
From 23 juanary 2013 on calendar.conf have to change type=caldav to type=ical http://forums.asterisk.org/viewtopic.php?f=1&t=85623 Regards Mensaje original Asunto: Google Calendar issue Fecha: Sat, 23 Feb 2013 10:22:17 -0500 De: Bakko Para: asterisk-users@lists.digium.com hello, I'm trying to connect Asterisk to Google Calendar. The connection work fine but Asterisk don't retrieve any programmed event present on the calendar. Asterisk version 1.8.20.1 Any hint? Thank you - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ERROR: chan_dahdi.c: PRI Span: 3 PROBLEM: General: Badly Structured Component
Hi Shitian, the line works but the ERROR is annoying since it appears very frequently. I think I'll have to patch it in order to lower its priority, maybe a NOTICE. G On 02/22/2013 03:06 PM, Shitian Long wrote: Did you get it to work may I ask ? On Feb 20, 2013, at 3:49 PM, gincantalupo wrote: Hi all, has anybody ever encountered this ERROR before? It happens frequently on my debian6-based pbx. I'm using Asterisk 1.8.11 with dahdi-linux-2.4.1 and a quadBRI card. ERROR: chan_dahdi.c: PRI Span: 3PROBLEM: General: Badly Structured Component I tried to google but without success. Do you know what it means? Should I worry? Thank You Giorgio -- _ -- 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
