[asterisk-users] [FreePBX] Digium addons
Hy, Does anybody knows how to show the digium addons in the freepbx GUI. The module is available in the GUI but sadly empty! Everything seems to be correctly installed bute the tables in the database are totally empty. Is there any script anywhere to fill those digium tables? Working with latest freepbx framework 2.9. regards Florent THOMAS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisknow mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisknow -- _ -- 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] How to remove asterisk ?
Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there . - 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
Re: [asterisk-users] How to remove asterisk ?
Maybe you could think about a root command line like that : yum uninstall asterisk Le 10/06/2011 11:26, virendra bhati a écrit : Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there . - 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
[asterisk-users] Asterisk issue or VoIP provider issue ?
Hi List, I want to set my caller ID and name with asterisk. So that when I make outgoing calls then destination end will see my name with number. from asterisk end I set both the things into dialplan. --- -- exten = _X.,n,Set(CALLERID(num)=9172341457) exten = _X.,n,Set(CALLERID(name)=Virendra Bhati) But when call reach to destination number then only number is display, name was display as *unknown * Is this issue of voip provider or Asterisk 1.6.2.18 ? I contact them they replay me that it's your end issue not my end. - 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
[asterisk-users] R: How to remove asterisk ?
What do you mean? Did you installed from sources or distro packet? sources: make uninstall distro: Every distro has its own commands (yum, apt-get ecc) Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di virendra bhati Inviato: venerdì 10 giugno 2011 11:26 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [asterisk-users] How to remove asterisk ? Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there . - 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
Re: [asterisk-users] Fwd: Re: ControlPlayback's options
On 2011-06-10 07:30, virendra bhati wrote: Hi John, Sorry for wrong information. Actually it's J not P option in ControlPlayBack... http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback That page is correct for asterisk 1.4 but the feature you need is in 1.6.0 and forward. Have you checked to documentation in the CLI? The J option is something different.. /Johan On Fri, Jun 10, 2011 at 12:23 AM, Johan Wilfer li...@jttech.se mailto:li...@jttech.se wrote: Humm... Seems like my message didn't make it. Here we go again.. /Johan Original Message Subject:Re: [asterisk-users] ControlPlayback's options Date: Sun, 05 Jun 2011 22:19:18 +0200 From: Johan Wilfer li...@jttech.se mailto:li...@jttech.se To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com On 2011-06-05 19:54, virendra bhati wrote: Hi John Wilfer, Thanks for replay. Now all things is working on asterisk 1.6.2.18 version. But When I try the same thing on Asterisk 1.4.X then facing problem. Is this the problem of ControlPlayback 's option fields of asterisk 1.4.X in this version have option P(jumping) not O(time) ? Is there any way by which we will implement like by upload ControlPlayback from asterisk 1.6 to 1.4 or else ? ControlPlayback(filename[,skipms[,ff[,rew[,stop[,pause[,restart[,options]]]) These features only exist in 1.6.0 and forward. Should be relative easy to make a backport if you need to run it on 1.4 I've never heard of the P()-option, which version of asterisk? /Johan -- Med vänlig hälsning Johan Wilfer email:jo...@jttech.se mailto:jo...@jttech.se JT Tech | Utvecklare webb:http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- 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 -- - 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 -- Med vänlig hälsning Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- 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] Callerid issue
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
44578900 100
44578901 101
44578902 102
44578902 103
44578903 104
44578905 200
44578906 275
44578907 277
44578908 354
I need to setup the callerid with this extensions . for example whenever I
am dial from 354 extension callerID will show 44578908.
fro this scenarion I need logical dialplan because I have 100 extenstions ,
so 100 extentions should be have different extensions.
Below dialplan is for sequence callerid and extesions. like 101 to 199
should callerid is going 44578900 to 44578999 .
exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten = _0X,3,NoOp(Ext ident:${outgoing_ident})
exten = _0X,4,Set(CALLERID(name)=${outgoing_ident})
exten = _0X,5,AGI(agi://127.0.0.1:4577/call_log)
exten = _0X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
exten = _0X,9,Hangup
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] Permanent restart after upgrade
On Fri, Jun 10, 2011 at 1:48 AM, Hans Witvliet h...@a-domani.nl wrote: On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote: On Thu, 9 Jun 2011, Hans Witvliet wrote: I went originally from a almost working machine running: asterisk180-1.8.3.2-87.1 To a machine that continuously restarts asterisk (+core dumps) running: asterisk180-1.8.3-85.2 Any chance you have a mix of Asterisk and module versions? Was Zaptel/Dahdi compiled with the proper set of headers for your kernel? Can you start Asterisk from the command line instead of the usual startup script? What do the first couple of errors look like? Capturing the output via the 'script' command will help. For example*, script foo sudo /usr/sbin/asterisk -C /etc/asterisk/asterisk.conf\ -c -d -d -d -f -g -n -p -q -v -v -v exit Can you turn off auto module loading and start with no modules? *) I'm a 1.2 Luddite, so the command line arguments may have changed... No dahdi/zaptel involved. I'll be off to work in a while, report back later. hw It amazes me when people run into a problem but refuse to post logs or verbose when you start Asterisk. Nothing meaninful. I would wager a gentleman's bet that I can have your system working just fine in a half hour or less (unless your bandwidth sucks). If I do it, then you have to post to the list and you owe me a favor, plus, in the future you have to help someone else. If I don't, I have to post my failure to the list and I owe you a favor. I have spare cycles, just let me know. Thanks, Steve Totaro -- _ -- 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] Callerid issue
On Friday 10 Jun 2011, mahesh katta wrote: Hi, I have 44578900 to 44578999 DID's. and I have extensions(100) for this DID's. but problem is callerid Extensions 44578900 100 44578901 101 44578902 102 44578902 103 44578903 104 44578905 200 44578906 275 44578907 277 44578908 354 I need to setup the callerid with this extensions . for example whenever I am dial from 354 extension callerID will show 44578908. fro this scenarion I need logical dialplan because I have 100 extenstions , so 100 extentions should be have different extensions. Sounds like a case for either (1) a different context per originating extension (or maybe, per group of originating extensions which all happen to obey the same mathematical formula for determining outside callerID from inside extension number); (2) an AGI script, accessing a database which links internal extensions to external numbers; or (3) rethinking your internal extension numbering scheme so there is a consistent mapping from internal to external numbers, thus allowing you to do it all mathematically. -- AJS Answers come *after* questions. -- _ -- 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] Callerid issue
On Fri, Jun 10, 2011 at 5:35 AM, mahesh katta maheshka...@flexydial.com wrote:
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
44578900 100
44578901 101
44578902 102
44578902 103
44578903 104
44578905 200
44578906 275
44578907 277
44578908 354
I need to setup the callerid with this extensions . for example whenever I
am dial from 354 extension callerID will show 44578908.
fro this scenarion I need logical dialplan because I have 100 extenstions ,
so 100 extentions should be have different extensions.
Below dialplan is for sequence callerid and extesions. like 101 to 199
should callerid is going 44578900 to 44578999 .
exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten = _0X,3,NoOp(Ext ident:${outgoing_ident})
exten = _0X,4,Set(CALLERID(name)=${outgoing_ident})
exten = _0X,5,AGI(agi://127.0.0.1:4577/call_log)
exten = _0X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
exten = _0X,9,Hangup
--
Best Regards,
Mahesh Katta
BUZZWORKS Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
Are you using SIP phones? I assume you are but just checking.
Look into sip.conf
For each phone, add callerid=Joe Smith 1551212 no quotes in sip.conf
Be sure the telco allows it, some times they will only allow the BTN.
I have run into troubles with toll free callerid. Everything made
sense because the problem was calling other toll free numbers and who
pays. Good point. Especially because ANI can be manipulated in
Asterisk as well.
I never understood hy people who have block of DIDs in a row choose to
make life difficult by not incrementing extensions by one, send caller
ID by prepending the common numbers and only sending four digits.
I guess if you get huge there could be duplicates of the last four.
Just set it in sip.conf
Thanks,
Steve T
Thanks
Steve Totaro
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_
-- 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:
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Re: [asterisk-users] Callerid issue
On Fri, Jun 10, 2011 at 6:27 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 10 Jun 2011, mahesh katta wrote: Hi, I have 44578900 to 44578999 DID's. and I have extensions(100) for this DID's. but problem is callerid Extensions 44578900 100 44578901 101 44578902 102 44578902 103 44578903 104 44578905 200 44578906 275 44578907 277 44578908 354 I need to setup the callerid with this extensions . for example whenever I am dial from 354 extension callerID will show 44578908. fro this scenarion I need logical dialplan because I have 100 extenstions , so 100 extentions should be have different extensions. Sounds like a case for either (1) a different context per originating extension (or maybe, per group of originating extensions which all happen to obey the same mathematical formula for determining outside callerID from inside extension number); (2) an AGI script, accessing a database which links internal extensions to external numbers; or (3) rethinking your internal extension numbering scheme so there is a consistent mapping from internal to external numbers, thus allowing you to do it all mathematically. -- AJS Answers come *after* questions. Why do programmers try to make solution so elegant when an entries for each phone in sip.conf is all that is needed. No need for mathematical formulas, AGIs, and databases. You just took over engineering to a new level. Thanks, Steve T -- _ -- 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] Connected Line ID
Hai, Does anybody have problems with a wrong Connected Line ID with asterisk version 1.6 The following bug was for version 1.4, but I cannot make up if this bug is still in version 1.6 http://forums.digium.com/viewtopic.php?t=7780 In version 1.8 it is possible to change the Connected Line ID, but this isn't the case in version 1.6 Regards, Arjan Kroon Mobillion BV -- _ -- 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] Connected Line ID
Arjan Kroon | Mobillion wrote: Does anybody have problems with a wrong Connected Line ID with asterisk version 1.6 As far as I know, unless you're applying patches yourself, Connected Line ID is only available for the 1.8 series. I'm running it on 1.4 with patches. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] R: How to remove asterisk ?
Hi List, I don't install from yum repository. I download tar file from asterisk.org On Fri, Jun 10, 2011 at 3:03 PM, Alexandru Oniciuc alexandru.onic...@trivenet.it wrote: What do you mean? Did you installed from sources or distro packet? sources: make uninstall distro: Every distro has its own commands (yum, apt-get ecc) Alex *Da:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Per conto di *virendra bhati *Inviato:* venerdì 10 giugno 2011 11:26 *A:* Asterisk Users Mailing List - Non-Commercial Discussion *Oggetto:* [asterisk-users] How to remove asterisk ? Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there . - 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 -- - 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
Re: [asterisk-users] Callerid issue
Steve Totaro wrote: For each phone, add callerid=Joe Smith1551212 no quotes in sip.conf The problem with that solution is that station to station calls will show the same CID and not the extension. I'd vote for the database approach. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Callerid issue
On Friday 10 Jun 2011, Steve Totaro wrote: Why do programmers try to make solution so elegant when an entries for each phone in sip.conf is all that is needed. No need for mathematical formulas, AGIs, and databases. You just took over engineering to a new level. Because doing it your way would cause the external caller ID number always to show up on the phone you dialled, even when making an internal call. This probably is not what you want. -- AJS Answers come *after* questions. -- _ -- 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] Connected Line ID
We have two systems one with version 1.6 and one with version 1.8 With 1.8 we don't see the problem Unfortunately it is not possible to upgrade 1.6 to 1.8. But are there also pathes for version 1.6 Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle Verzonden: 10-06-2011 12:58 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID Arjan Kroon | Mobillion wrote: Does anybody have problems with a wrong Connected Line ID with asterisk version 1.6 As far as I know, unless you're applying patches yourself, Connected Line ID is only available for the 1.8 series. I'm running it on 1.4 with patches. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Callerid issue
-Original Message-
From: Steve Totaro stot...@asteriskhelpdesk.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 10 Jun 2011 06:30:53
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
Subject: Re: [asterisk-users] Callerid issue
On Fri, Jun 10, 2011 at 5:35 AM, mahesh katta maheshka...@flexydial.com wrote:
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
44578900 100
44578901 101
44578902 102
44578902 103
44578903 104
44578905 200
44578906 275
44578907 277
44578908 354
I need to setup the callerid with this extensions . for example whenever I
am dial from 354 extension callerID will show 44578908.
fro this scenarion I need logical dialplan because I have 100 extenstions ,
so 100 extentions should be have different extensions.
Below dialplan is for sequence callerid and extesions. like 101 to 199
should callerid is going 44578900 to 44578999 .
exten =_0X,1,NoOp(Int exten:${CALLERID(num)})
exten =_0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten =_0X,3,NoOp(Ext ident:${outgoing_ident})
exten =_0X,4,Set(CALLERID(name)=${outgoing_ident})
exten =_0X,5,AGI(agi://127.0.0.1:4577/call_log)
exten =_0X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten =_0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
exten =_0X,9,Hangup
--
Best Regards,
Mahesh Katta
BUZZWORKS Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
Are you using SIP phones? I assume you are but just checking.
Look into sip.conf
For each phone, add callerid=Joe Smith 1551212 no quotes in sip.conf
Be sure the telco allows it, some times they will only allow the BTN.
I have run into troubles with toll free callerid. Everything made
sense because the problem was calling other toll free numbers and who
pays. Good point. Especially because ANI can be manipulated in
Asterisk as well.
I never understood hy people who have block of DIDs in a row choose to
make life difficult by not incrementing extensions by one, send caller
ID by prepending the common numbers and only sending four digits.
I guess if you get huge there could be duplicates of the last four.
Just set it in sip.conf
Thanks,
Steve T
Thanks
Steve Totaro
--
_
-- 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
--
_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] Callerid issue
On Fri, Jun 10, 2011 at 4:00 PM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:
On Fri, Jun 10, 2011 at 5:35 AM, mahesh katta maheshka...@flexydial.com
wrote:
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
44578900 100
44578901 101
44578902 102
44578902 103
44578903 104
44578905 200
44578906 275
44578907 277
44578908 354
I need to setup the callerid with this extensions . for example whenever
I
am dial from 354 extension callerID will show 44578908.
fro this scenarion I need logical dialplan because I have 100 extenstions
,
so 100 extentions should be have different extensions.
Below dialplan is for sequence callerid and extesions. like 101 to 199
should callerid is going 44578900 to 44578999 .
exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten = _0X,3,NoOp(Ext ident:${outgoing_ident})
exten = _0X,4,Set(CALLERID(name)=${outgoing_ident})
exten = _0X,5,AGI(agi://127.0.0.1:4577/call_log)
exten = _0X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
exten = _0X,9,Hangup
--
Best Regards,
Mahesh Katta
BUZZWORKS Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
(E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
Are you using SIP phones? I assume you are but just checking.
sir I am using IP phones means sip id's only
[100]
username=100
secret=123
mailbox=100
type=friend
host=dynamic
canreinvite=no
context=default
qualify=yes
Thanks,
Steve T
Thanks
Steve Totaro
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Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] Callerid issue
On Friday 10 Jun 2011, Steve Totaro wrote: I never understood hy people who have block of DIDs in a row choose to make life difficult by not incrementing extensions by one, send caller ID by prepending the common numbers and only sending four digits. Well, to be fair, that's what most people usually start out trying to do -- make it all line up neatly, with each department having numbers in a certain range (1xx for management, 2xx for purchasing, 3xx for sales, 4xx for IT, 5xx for shop floor, 8xx as short codes for direct access to selected external numbers from phones that shouldn't normally have access to outside lines but still need to call certain numbers occasionally) and so forth. But then, once you have invested considerable time and effort devising a plan for allocating numbers, somebody On High inevitably makes a decision that ruins the whole thing. -- AJS Answers come *after* questions. -- _ -- 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] Connected Line ID
Arjan Kroon | Mobillion wrote: But are there also pathes for version 1.6 The last patch available for the 1.6 series was for 1.6.0.1: https://issues.asterisk.org/jira/browse/8824 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Callerid issue
On Fri, Jun 10, 2011 at 5:23 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Friday 10 Jun 2011, Steve Totaro wrote: I never understood hy people who have block of DIDs in a row choose to make life difficult by not incrementing extensions by one, send caller ID by prepending the common numbers and only sending four digits. Well, to be fair, that's what most people usually start out trying to do -- make it all line up neatly, with each department having numbers in a certain range (1xx for management, 2xx for purchasing, 3xx for sales, 4xx for IT, 5xx for shop floor, 8xx as short codes for direct access to selected external numbers from phones that shouldn't normally have access to outside lines but still need to call certain numbers occasionally) and so forth. yes sir there is some depart ments. But then, once you have invested considerable time and effort devising a plan for allocating numbers, somebody On High inevitably makes a decision that ruins the whole thing. -- AJS Answers come *after* questions. -- _ -- 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 -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.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] Callerid issue
On Fri, Jun 10, 2011 at 5:23 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Friday 10 Jun 2011, Steve Totaro wrote: I never understood hy people who have block of DIDs in a row choose to make life difficult by not incrementing extensions by one, send caller ID by prepending the common numbers and only sending four digits. Well, to be fair, that's what most people usually start out trying to do -- make it all line up neatly, with each department having numbers in a certain range (1xx for management, 2xx for purchasing, 3xx for sales, 4xx for IT, 5xx for shop floor, 8xx as short codes for direct access to selected external numbers from phones that shouldn't normally have access to outside lines but still need to call certain numbers occasionally) and so forth. Yes sir, I have different departments and, different extensions. and I am using sipphones only and have PRI line with 100 DID's which i mention above. But then, once you have invested considerable time and effort devising a plan for allocating numbers, somebody On High inevitably makes a decision that ruins the whole thing. -- AJS Answers come *after* questions. -- _ -- 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 -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.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] AMI question
Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X 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
[asterisk-users] (no subject)
Good morning gentlemen, is my first post in the list, now I'm starting asterisk
wanted to have your help for some questions.
Well the first function is as follow me. Here
I will demonstrate how this configuration follow me on my
extensions.conf but it is not working, and do not know why, but
something is missing?
You must set up followme.conf ?
What
I want is that the follow-me is enabled for any of the extensions
within the same context, like if I am absent from my table and go to
extension 2801 DataCenter where I need to spend all afternoon and I will
have the extension 2820 which enabled me to follow this extension and after
back to my desk withdraw follow me.
; Ativa Siga-me incondicional
[sigame-on]exten = _*71*.,1,NoCDR()
exten = _*71*.,2,Set(DB(CF/${CALLERID(num)})=${EXTEN:4})
exten = _*71*.,3,Playback(call-fwd-unconditionalforextension)
exten = _*71*.,4,SayDigits(${CALLERID(num)})
exten = _*71*.,5,Playback(is-set-to)
exten = _*71*.,6,SayDigits(${EXTEN:4})
exten = _*71*.,7,Playback(vm-saved)
exten = _*71*.,8,Playback(beep)
exten = _*71*.,9,Hangup
; Desativa o siga-me incondicional
[sigame-off]exten = _*72*,1,NoCDR()
exten = _*72*,2,DBdel(CF/${CALLERID(num)})
exten = _*72*,3,Playback(cancelled) exten = _*72*,4,Playback(beep)
exten = _*72*,5,Hangup
Bom, agora vamos ao pulo do gato, esse passo é muito importante pois é
ele quem verifica se existe ou não o siga-me para o ramal.
Vamos ao contexto:
[disca]
exten = _3XXX,1,Noop(CF/${EXTEN})
exten = _3XXX,2,Set(siga=${DB(CF/${EXTEN})})
exten = _3XXX,3,Dial(SIP/${siga},30,Ttw)
exten = _3XXX,4,Dial(SIP/${EXTEN}) ; Unconditional forward
exten = _3XXX,5,Hangup
--
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Re: [asterisk-users] AMI question
Either use ExtensionState or watch for Hold/Unhold events. http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+ExtensionState -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Friday, June 10, 2011 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AMI question Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X 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 -- _ -- 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 issue or VoIP provider issue ?
Many providers do not allow for caller ID name to be sent. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Friday, June 10, 2011 5:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk issue or VoIP provider issue ? Hi List, I want to set my caller ID and name with asterisk. So that when I make outgoing calls then destination end will see my name with number. from asterisk end I set both the things into dialplan. --- -- exten = _X.,n,Set(CALLERID(num)=9172341457) exten = _X.,n,Set(CALLERID(name)=Virendra Bhati) But when call reach to destination number then only number is display, name was display as unknown Is this issue of voip provider or Asterisk 1.6.2.18 ? I contact them they replay me that it's your end issue not my end. - 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
Re: [asterisk-users] [FreePBX] Digium addons
Le 10/06/2011 08:07, Florent THOMAS a écrit : Hy, Does anybody knows how to show the digium addons in the freepbx GUI. The module is available in the GUI but sadly empty! Everything seems to be correctly installed bute the tables in the database are totally empty. Is there any script anywhere to fill those digium tables? Working with latest freepbx framework 2.9. regards Florent THOMAS -- _ -- 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 Does anybody has deal with the same issue? Thanks for your help, regards -- _ -- 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] R: How to remove asterisk ?
On Fri, 2011-06-10 at 16:31 +0530, virendra bhati wrote: Hi List, I don't install from yum repository. I download tar file from asterisk.org On Fri, Jun 10, 2011 at 3:03 PM, Alexandru Oniciuc alexandru.onic...@trivenet.it wrote: What do you mean? Did you installed from sources or distro packet? sources: make uninstall distro: Every distro has its own commands (yum, apt-get ecc) Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di virendra bhati Inviato: venerdì 10 giugno 2011 11:26 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [asterisk-users] How to remove asterisk ? Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there . If you installed from source then you just need to make uninstall from the source directory or remove the following by hand: /etc/asterisk /var/lib/asterisk /var/spool/asterisk /usr/lib/asterisk If you installed the initd script you need to erase that too. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Request: please test modification to EWS calendar functionality
I have expanded the EWS calendar functionality within Asterisk 1.8 so it is now possible to access any calendar within an Exchange 2007 or 2010 server. I have put the changes onto the reviewboard for astrisk but currently no one responded. So if you use the EWS calendar functionality within Asterisk and would like to have access to any calendar in Exchange please try the patch in the following review request: https://reviewboard.asterisk.org/r/1152/. Please reply to the reviewboard if it is working for you or if you experience problems. Regards, Michel Verbraak. -- _ -- 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] AMI question
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Friday, June 10, 2011 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI question Either use ExtensionState or watch for Hold/Unhold events. http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action +ExtensionState Eric Thanks I have 2 questions: 1) trying to use the command and not getting anything of value: Response: Success ActionID: 1 Message: Extension Status Exten: 401 Context: polycom Hint: Status: -1 Current calls are this: SIP/401-001a!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!401!!3!24!SIP/404-0019 SIP/404-0019!smvoice-sip!401!6!Up!Dial!SIP/401|20|!204!!3! 24!SIP/401-001a status -1 is not found. 401 is valid for me and even active. what am I not sending correct? I also tried SIP/401, same result. 2) Is it possible to status for Incoming calls also - if they are on hold. how do I do that? Do a core show hints. If the extension@context is not on that list, then you can't monitor the state of the extension.See voip-info.org for info on setting up hints. No, you can only monitor extensions and devices. -- _ -- 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] Incoming Call Recording
Longtime lurker, first time poster. :) A client of mine is in need of having Asterisk record every call that comes in from a specific incoming route. I've added the following lines to the sip_additional.conf file, but no recordings are showing up in the /var/spool/asterisk/monitor/ folder. record_out=always record_in=always Another page I came across on Google ( http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor) suggested I add the following line to my sip.conf file: exten = 2060,3,Monitor(wav,myfilename) I can see how this could work, but I'm not sure what to replace 2060 with, as what I need setup is the record of all incoming calls across the board, not just calls associated with a particular extension number (ie: 2060). Your sure is appreciated! -- Rick Hall Senior Vice President ReadyWire Multimedia Solutions Affordable Website Reseller Hosting http://www.readywire.com/ (312) 278-4446 x5446 Technical Support: 24 hours a day / 7 days a week Customer Login...: https://secure.readywire.com/ Server Notices.: http://status.readywire.com/ Support Center: https://secure.readywire.com/ Twitter.: http://twitter.com/readywire Blog: http://blog.readywire.com/ This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. ReadyWire Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681. www.readywire.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] Incoming Call Recording
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Hall
Sent: Friday, June 10, 2011 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Incoming Call Recording
Longtime lurker, first time poster. :)
A client of mine is in need of having Asterisk record every call that comes
in from a specific incoming route. I've added the following lines to the
sip_additional.conf file, but no recordings are showing up in the
/var/spool/asterisk/monitor/ folder.
record_out=always
record_in=always
Another page I came across on Google
(http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor) suggested I add
the following line to my sip.conf file:
exten = 2060,3,Monitor(wav,myfilename)
I can see how this could work, but I'm not sure what to replace 2060 with,
as what I need setup is the record of all incoming calls across the board,
not just calls associated with a particular extension number (ie: 2060).
Your sure is appreciated!
--
Rick Hall
Senior Vice President
ReadyWire Multimedia Solutions
Affordable Website Reseller Hosting
http://www.readywire.com/
(312) 278-4446 x5446
Technical Support:
24 hours a day / 7 days a week
Customer Login...: https://secure.readywire.com/
Server Notices.: http://status.readywire.com/
Support Center: https://secure.readywire.com/
Twitter.: http://twitter.com/readywire
Blog: http://blog.readywire.com/
This message contains confidential information and is intended only for the
individual named. If you are not the named addressee you should not
disseminate, distribute or copy this e-mail. Please notify the sender
immediately by e-mail if you have received this e-mail by mistake and delete
this e-mail from your system. E-mail transmission cannot be guaranteed to be
secure or error-free as information could be intercepted, corrupted, lost,
destroyed, arrive late or incomplete, or contain viruses. The sender
therefore does not accept liability for any errors or omissions in the
contents of this message, which arise as a result of e-mail transmission. If
verification is required please request a hard-copy version. ReadyWire
Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681.
www.readywire.com.
This will do the trick, but you should play the you are being recorded
file to cover your assets
exten = s,n,MixMonitor(Zap_${UNIQUEID}.wav|av(0}V(0))
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[asterisk-users] Queue not sending call to Agent
Queue not sending call to Agent
I am having an issue and i am not sure if it is a bug or a config issue. I
was originally running Asterisk 1.8.1.1 when I noticed this issue. I
upgraded to 1.8.4.2 to see if that would fix it but it didn't.
The issue is that I have a call queue and the agent dials a number to log
into the queue. When someone calls the queue the first time the call is
sent to the agent without issue. The issue is that any calls after the
first are placed in the queue and never sent to the agent who is logged in
and available. Before I call the queue I do a show queue and it shows the
agent as
Asterisk18*CLI queue show
irock.com has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime,
0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/9013XX9XX8 (dynamic) (Not in use) has taken no calls yet
No Callers
Then the call comes into the queue and the callee just sits in the queue.
When I do a show queue again when the callee is in the queue it shows the
agent as busy
Asterisk18*CLI queue show
irock.com has 1 calls (max unlimited) in 'ringall' strategy (0s holdtime,
0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/9013XX9XX8 (dynamic) (Busy) has taken no calls yet
Callers:
1. SIP/9013XX9XX8-0001 (wait: 0:12, prio: 0)
So I am not sure what happened because the agent was free before the call.
If I do a reload at the Asterisk CLI and then call again the agent gets the
call and then the second call is once again placed in the queue. I will
attach a SIP Debug that shows what is going on. I don't see any SIP invites
leaving Asterisk to invite the agent to the call.
One other thing Currently in my config I have the agent show up as just
the username which is the phone number. If I set it so that the agent shows
up as phonenumber@blah then I can call the agent constantly without any
issue. The only problem here is that when I do a queue show the agent
shows up as unknown status. So when the agent is on a call and someone
else calls the agent will be interrupted.
This is what I have in queues.conf
[irock.com]
strategy=ringall
ringinuse=no
joinempty=yes
leavewhenempty=no
announce-frequency=30
min-announce-frequency=15
periodic-announce-frequency=60
announce-holdtime=yes
announce-position=yes
; (You are now first in line.)
queue-youarenext = queue-youarenext
; (There are)
queue-thereare = queue-thereare
; (calls waiting.)
queue-callswaiting = queue-callswaiting
; (The current est. holdtime is)
queue-holdtime = queue-holdtime
; (minutes.)
queue-minutes = queue-minutes
; (seconds.)
queue-seconds = queue-seconds
; (Thank you for your patience.)
queue-thankyou = queue-thankyou
; (Hold time)
queue-reporthold = queue-reporthold
; (All reps busy / wait for next)
periodic-announce = queue-periodic-announce
This is what I have in extensions.conf
exten = 9012XX1XX1,1,Answer()
exten = 9012XX1XX1,n,Set(QUEUE_MAX_PENALTY=0);
exten = 9012XX1XX1,n,Queue(irock.com,t)
exten = 9012XX1XX1,n,Hangup()
exten = *50,1,Answer
exten = *50,n,Macro(queue-login,${EXTEN},${EXTEN:0:4})
exten = *50,n,Hangup
exten = *51,1,Answer
exten = *51,n,Macro(queue-logout,${EXTEN},${EXTEN:0:4})
exten = *51,n,Hangup
[macro-queue-login]
exten = s,1,Set(agent=${EXTEN:4})
exten = s,n,Set(queue=irock.com)
exten = s,n,NoOp(Queue login agent ${EXTEN:4} to queue ${phone});
exten = s,n,AddQueueMember(${queue});
exten = s,n,Playback(agent-loginok)
[macro-queue-logout]
exten = s,1,Set(agent=${EXTEN:4})
exten = s,n,Set(queue=irock.com)
exten = s,n,NoOp(Queue logout agent ${EXTEN:4} from queue ${phone});
exten = s,n,RemoveQueueMember(${queue});
exten = s,n,Playback(agent-loggedoff)
--
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[asterisk-users] Solved, was: Permanent restart after upgrade
On Fri, 2011-06-10 at 05:52 -0400, Steve Totaro wrote: On Fri, Jun 10, 2011 at 1:48 AM, Hans Witvliet h...@a-domani.nl wrote: On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote: On Thu, 9 Jun 2011, Hans Witvliet wrote: I went originally from a almost working machine running: asterisk180-1.8.3.2-87.1 To a machine that continuously restarts asterisk (+core dumps) running: asterisk180-1.8.3-85.2 Any chance you have a mix of Asterisk and module versions? Was Zaptel/Dahdi compiled with the proper set of headers for your kernel? Can you start Asterisk from the command line instead of the usual startup script? What do the first couple of errors look like? Capturing the output via the 'script' command will help. For example*, script foo sudo /usr/sbin/asterisk -C /etc/asterisk/asterisk.conf\ -c -d -d -d -f -g -n -p -q -v -v -v exit Can you turn off auto module loading and start with no modules? *) I'm a 1.2 Luddite, so the command line arguments may have changed... No dahdi/zaptel involved. I'll be off to work in a while, report back later. hw It amazes me when people run into a problem but refuse to post logs or verbose when you start Asterisk. Nothing meaninful. I would wager a gentleman's bet that I can have your system working just fine in a half hour or less (unless your bandwidth sucks). If I do it, then you have to post to the list and you owe me a favor, plus, in the future you have to help someone else. If I don't, I have to post my failure to the list and I owe you a favor. I have spare cycles, just let me know. Thanks, Steve Totaro Hi Steve, thanks for your time and consideration. Hadn't a chance to report back, as i just returned from work ;-( I think i found the reason behind it; a missing file from the update. As the machine involved is not connected to Internet, each and every file has to be put on a portable medium, checked, and only then i'm allowed to put it on our corporate lan. It turned out, that not all required files were transferred on the usb-disk, or removed by someone. Anyway it, i copied the missing file ( ../repo/network:/telephony:/asterisk/SLE_11_SP1/x86_64/asterisk180-1.8.4.2-90.1.x86_64.rpm ) manually, updated it again and: voila So regarding not posting config/log/trace/core's.. Initially i just put the symptom's on the list. Next step would have been any specific log's or config files. As you know, any most of the logfiles for *, can be rather long, same for logfiles. If you are still interested in it (educational purposes) i can still get the /var/log/asterisk/messages file, and put the relevant section on the list Bottomline is however, when doing an upgrade, and only some of the asterisks RPM's are processed, you get funny results And what you see on the asterisk console/logfile does not indicate directly what went wrong. If i run into a situation, it is mostly that i can not get something working in the first place, or i made an incorrect change. In those cases the remedy is obvious. In this case however, the cause is not PEBKAC or asterisk issue, but something with the prebuild packages, i'll guess a missing dependancy, which allowed some asterisk files to be updated, while not all were present, resulting into an unstable result. So, i'll try to find the person responsible for packaging, and try to convince that he ahs some work to do. 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
