Re: [asterisk-users] Ast12 issue "missing" library file??
On Wed, Oct 23, 2013 at 12:46 PM, Cassius Smith wrote: > Hi ALL, > still having trouble getting Ast 12 to run. I got it compiled and built > but now when I try to run, I'm getting a missing library error that seems > to be in error (see below). The .so file DOES exist with correct > permissions. > > > > Any ideas? > > Many thanks, > Cassius > > I've found a few solutions from the asterisk-dev mailing list, some posts[1] from last August. First, you may need to refresh your dynamic linker cache using ldconfig. Or, if you're using a 64-bit distro, you may have to rerun configure with the following option: "./configure --libdir=/usr/lib64" [1]: http://lists.digium.com/pipermail/asterisk-dev/2012-August/056610.html If neither of these work, let me know. I've got a working asterisk-12 install running on CentOS 6.4 in a virtual-box environment that I can play with. -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Disable peer from AMI
On Tue, Oct 22, 2013 at 11:36 PM, Michelle Dupuis wrote: > I need to disable/enable a peer after hours automatically, and am > thinking about doing so via the AMI. > > Is there a command to enable/disable (or perhaps delete/add) a peer via > the AMI? I could create code to modify sip.conf and force a reload, but > that seems like the wrong approach... > > Have you considered adding realtime sip peers? They can peacefully co-exist with a sip.conf file. Then just build in some logic to sip prune when you need to remove the peer...(I forget the actual command at the moment). As far as I can tell, no AMI would be needed... -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-12 issue after successful installation
On Mon, Oct 21, 2013 at 7:26 AM, virendra bhati wrote: > Hi Team, > > I have installed asterisk-12 Beta but when I try to asterisk start then > get below issue. > > *[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r > asterisk: error while loading shared libraries: libjansson.so.4: cannot > open shared object file: No such file or directory > [root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]#* > > So, as a specific answer to the original question, the proper resolution to this issue, assuming you manually installed libjansson, is the following, pulled from the install_prereq scripts: echo "/usr/local/lib" > /etc/ld.so.conf.d/usr_local.conf /sbin/ldconfig This worked for me on a fresh CentOS 6.4 installation where I didn't use the install_prereq script, and thus was having your same issue. Hope this helps someone in the future! -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] realtime voicemail asterisk 11
On Mon, Oct 14, 2013 at 2:15 PM, troxlinux wrote: > thnk Warren , I only see one warning message > > [Oct 14 13:12:14] WARNING[2736][C-]: app_voicemail.c:3768 > retrieve_file: SQL Get Data error! coltitle=category > [SELECT * FROM voicemessages WHERE dir=? AND msgnum=?] > > I'm not sure on this. Hopefully someone else can help. -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] realtime voicemail asterisk 11
On Mon, Oct 14, 2013 at 12:19 PM, troxlinux wrote: > res_config_mysql > > [general] > dbhost = 127.0.0.1 > dbname = asterisk > dbuser = root > dbpass = x > dbport = 3306 > dbsock = /var/lib/mysql/mysql.sock > > extconfig.conf > voicemail=> mysql,asterisk,voicemail_messages > First issue I see - you've got the context named [general] in res_config_mysql, but you're attempting to connect to asterisk in extconfig.conf (mysql,*asterisk*,voicemail_messages). The second item in extconfig.conf should match the database context name in res_config_mysql. So either fix res_config_mysql by changing [general] to [asterisk], or fix extconfig.conf by changing the line to "voicemail => mysql,general,voicemail_messages". Make whichever change you prefer (I would make the change in res_config_mysql.conf personally, but it's up to you), and then reload asterisk to see if that resolves the error. Otherwise, post whatever you're new error is. -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] realtime voicemail asterisk 11
On Mon, Oct 14, 2013 at 11:13 AM, troxlinux wrote: > Hi list, I'm trying to put my voicemail on asterisk realtime with 11.XX, > generate tables in a couple of files in the folder realtime / mysql , > voicemail_messages.sql and voicemail.sql > > the connection with mysql and odbc works well > > isql asterisk useradmin xxx > +---+ > | Connected!| > | | > | sql-statement | > | help [tablename] | > | quit | > | | > +---+ > > > > [Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:645 > ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02: > [MySQL][ODBC 5.2(w) Driver][mysqld-5.6.12]Table 'asterisk.voicemessages' > doesn't exist (86) > [Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:657 > ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to > asterisk [asterisk]... > [Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:761 > ast_odbc_sanity_check: Connection is down attempting to reconnect... > [Oct 14 10:06:16] NOTICE[10037][C-0003]: res_odbc.c:1527 > odbc_obj_connect: Connecting asterisk > [Oct 14 10:06:16] NOTICE[10037][C-0003]: res_odbc.c:1559 > odbc_obj_connect: res_odbc: Connected to asterisk [asterisk] > [Oct 14 10:06:16] WARNING[10037][C-0003]: app_voicemail.c:5609 > messagecount: SQL Execute error! > > Could you post a sanitized version of your res_config_mysql.conf and extconfig.conf files? I'm thinking maybe you've got an error in there somewhere that's causing this error. -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Capture Media IP in CDR
On Fri, Oct 11, 2013 at 9:05 PM, CDR wrote: > I am not proxying the media, but never the less I am forced to store > the source media IP in my CDR, for regulatory reasons. Asterisk gets > that information when the reinvite comes, but how do I store it? > If I don't figure this out my next email will be from Federal Prison. > Kindly help me stay away from those guys. Eventually we all need to > save that information or we shall not be able to stay in business. > > You can add custom fields to your CDR records using Set(CDR(customfieldname)=foobar). I don't know the name of the variable you want that specifically contains the source media IP, but I imagine you can pull it with the SIP_HEADER function, or possibly the CHANNEL(recvip) function. -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 sending comfort Noise
On Tue, Oct 8, 2013 at 11:02 AM, Eric Wieling wrote: > I have an Asterisk 1.4 box which is sometimes getting the message below. > Here is the weird part, the CNG is coming from ANOTHER ASTERISK SERVER. > 209.220.119.19 is an Asterisk 11 box. > > [Oct 8 11:59:27] NOTICE[20798]: rtp.c:849 process_rfc3389: Comfort noise > support incomplete in Asterisk (RFC 3389). Please turn off on client if > possible. Client IP: 209.220.119.19 > Is the other asterisk server under your control? -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Strange issues with newly rebooted machine
On Tue, Aug 6, 2013 at 10:47 AM, Mike Diehl wrote: > I appreciate your quick response. I issued the commands specified and > got NO output! > > === > CLI> core set verbose 10 > Verbosity was 25 and is now 10 > CLI> core set debug 10 > Core debug was 25 and is now 10 > CLI> module unload chan_sip.so > CLI> module load chan_sip.so > CLI> > === > > The reason we had to reboot the machine is that we changed it's > physical location, but didn't change it's IP address. As part of the > restart, I also took the opportunity to rebuild a RAID-1 array. Other > than that, there have been no configuration changes since the last > time this worked. > > Any other ideas? > > Are the phone still working? I've noticed that realtime registered peers don't always show when I do "sip show peers" or even "sip show peer *name*". I usually only see the peer if I make a call to the peer or the peer makes a call first. Do you have rtcachefriends=yes in your sip.conf? -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 CDR vs VoIP Innovations CDR
On Thu, Aug 1, 2013 at 10:28 AM, Adam Moffett wrote: > When I compare my total minutes on the bill from VoIP Innovations, to the > number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count > of minutes. I'm wondering why it's there. > > Are there different methods of counting the billable start or end point of > a phone call? > > If it matters, I'm counting more termination minutes than they are and > they're counting more origination minutes than I am. > > If I remember correctly, they bill in sub-minute increments, something like 60 second minimum, then every 6 seconds after that. In other words, if you have a 20 second call, it's billed as 60 seconds, however, if you have a 62 second call, it's billed as 66. I don't remember what they're specific increments are, but I don't believe it was a straight bill. Are you finding that you're off by just a few seconds per call, or by minutes? When you say you're off by 3-4%, are you saying your CDR reports 100 minutes on a call and they are showing 104 minutes, or vice versa? -- _ -- 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 launch a URl when dialing a number
On Fri, May 31, 2013 at 11:29 AM, Salaheddine Elharit < salah.elharit...@gmail.com> wrote: > hello , > > thanks alex for your help and support the scenario is correct. > > i will try to follow your suggestion and i will update you asap > > thank you again for your explication i really appreciate it > > Have you tried maybe setting up the entire call in an AGI that will execute the desired script as you make the dial command? Or, you could look at running the M or U options in your Dial() command to execute a macro or gosub routine when the call is connected? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] dial and bridge
On Tue, May 14, 2013 at 11:16 AM, Lenz Emilitri wrote: > > Hi all, > I need some advice - I have been working on originating multiple calls > using AMI and then joining them. > What I want to do is: > - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or > Local/1234@ext) and "park" it somehow > - dial call 2 (where again the caller is in channel format) and join it to > the previous call. > > > Why not just originate from one extension to the other? Something like this (not tested): Action: Originate Channel: Local/300@from-internal Context: from-internal Exten: 500 Timeout: 30 Should dial extension 500 in the from-internal context after the call to 300@from-internal is answered. Meaning, the person at 300@from-internalwould have their phone ring, they'd pick it up, and then they'd hear ringing on the line as asterisk then dialed extension 500@from-internal. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] multiple provider for incoming
On Tue, Apr 30, 2013 at 7:50 PM, David Wessell wrote: > Hi Matt, > > You can't have multiple providers for inbound traffic. You can have > multiple providers for outbound traffic though. > > Thanks > David > > David, I'm not sure where you got this information, but it's not accurate. I've had multiple inbound and outbound SIP providers for years going to a single box. You just get a separate DID from each provider. Matt, The process will depend on your provider, of course, but I know some have an option that if they are unable to reach your box, then they can auto-forward to another DID, or to a voicemail box, or to a user-defined function, etc. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] E911 Voip Trunking
There are E911 providers that offer this functionality. I know off the top of my head, 911Enable offers a service like this. A former client of mine that provided hosted PBX services had a contract with them. I'm sure there are other providers out there as well. On Fri, Apr 19, 2013 at 10:41 AM, Chris Nighswonger < cnighswon...@foundations.edu> wrote: > During the course of a conversation with an member of the IT group who > handles the E911 center for our county, I learned that all of the county's > E911 is voip based. This got me to wondering why we could not just > configure up a SIP or some such trunk directly to the E911 center to handle > our emergency traffic. The county seems interested in exploring the > possibility. > > So I'm wondering if anyone else has attempted this. > > Kind Regards, > Chris > > -- > _ > -- 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, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Digium Wildcard TDM800P not working with DAHDI
What have you done so far to try and make it work? What version of CentOS are you using, what version of DAHDI, etc? On Sat, Mar 9, 2013 at 5:54 AM, Gilberto Sanches wrote: > Hello everyone, > > How can I let Digium Wildcard TDM800P work successfully with DAHDI? > Because the Centos recognizes the card but I can't get the analog card > working with DAHDI. > > > Thanks in advance, > Gilberto > > -- > _ > -- 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, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] recrding calls
Around version 1.4 or 1.6, TIMESTAMP was phased out and replaced with
STRFTIME. See this page for details on how to properly generate a
timestamp:
http://www.voip-info.org/wiki/view/Asterisk+func+strftime
On Fri, Jan 18, 2013 at 8:46 PM, Joseph wrote:
> On 01/18/13 19:27, Carlos Alvarez wrote:
>
>On Fri, Jan 18, 2013 at 6:25 PM, Joseph <[1]syscon...@gmail.com> wrote:
>>
>> I would like to outgoing/icoming calls and email the files.
>> This is what I have:
>> ...
>> exten => _7.,n,Set(CALLFILENAME=${**EXTEN:1}-${TIMESTAMP})
>> exten => _7.,n,Monitor(wav,${**CALLFILENAME},m)
>> ...
>> How do I email these file?
>>
>> This is how we do it:
>> exten =>
>> _1NXXNXX,1,Set(**recordfilename=/var/spool/**
>> asterisk/monitor/${EXTEN}-
>> ${TIMESTAMP:0:8}${TIMESTAMP:8}**.WAV)
>> \exten => _1NXXNXX,n,MixMonitor(${**recordfilename},b)
>> exten => _1NXXNXX,n,(dial here or whatever)
>>
>
> Thanks Carlos
> I'm just concentrating right now on ${TIMESTAMP} variable but is is not
> working:
>
> I have:
> exten => 11,n,Set(recordfilename=/var/**spool/asterisk/monitor/${**
> EXTEN}-${TIMESTAMP:0:8}${**TIMESTAMP:8}.WAV)
> exten => 11,n,MixMonitor(${**recordfilename},b)
>
> and the file name I got was: -11.wav
>
> Why I'm not getting any timestamp?
>
>
> --
> 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<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
--
_
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Re: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start
On Wed, Jan 16, 2013 at 1:37 PM, Danny Nicholas wrote: > Same issue exists with 11.2 > > I've created issue 20945 to track this, at least for 1.8.20.0. https://issues.asterisk.org/jira/browse/ASTERISK-20945 -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start
I'm trying to decide if I need to open an issue for this or if it's just a misconfiguration issue of some sort. Here's the situation - yesterday morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 installation and got a shell of a basic asterisk install setup (minimum required configuration files, etc, with no dialplan or sip peers setup yet). In the afternoon, I got the notification that asterisk 1.8.20.0 had been released, so today, I downloaded the latest 1.8-current.tar.gz and compiled and installed it (./configure, make menuselect and choose all the same options as my previous install, make, make install). Now, when I start the asterisk service using "service asterisk start" from the command line, this is the output: [root@pbx ~]# service asterisk start Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Starting asterisk: However, the /var/run/asterisk/asterisk.ctl file is being created and the process is starting: [root@pbx ~]# ls -lh /var/run/asterisk/ total 4.0K srwxr-xr-x 1 root root 0 Jan 16 12:07 asterisk.ctl -rw-r--r-- 1 root root 6 Jan 16 12:07 asterisk.pid However, I'm no longer getting the usual splash message when I connect to the asterisk console...this is what I get: [root@pbx ~]# asterisk -r Verbosity is at least 3 pbx*CLI> I don't have any peers setup yet, or even any dialplan configured to test, but when I go through the logs, I don't find any errors or warnings that I'm not expecting. I've gone back to the asterisk 1.8.19.1 install and everything works as expected (no error messages, full splash about license / version on connection to console, etc). I performed make clean in my 1.8.20 source directory, then ./configure, make menuselect, make, make install, and even make config, and I'm still seeing this message pop up when restarting / starting the service. I went through the CHANGELOG.TXT for 1.8.20.0 and it appears there are some items talking about changing the way the process starts up (commit r376428), but I'm not enough of a coder to understand if those would cause what I'm seeing. Is anyone else seeing this issue? Should I open an issue on the tracker? Anyone see something obvious I missed? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] faxdetect on/off on the fly?
On Thu, Jan 3, 2013 at 9:39 PM, David Cunningham wrote: > We have all calls going to an AGI, which decides where the number will get > routed to, and if fax detection should be enabled for this call. The choice > should only apply to the current call. > What criteria would determine if fax detection should be enabled? From reading this message, what it sounds like is you want the call to go to the AGI, and if a CNG tone is detected, you want it to go to a specific fax extension. That's what faxdetect does. You enable it on all your lines, and if a CNG tone is detected, it sends it to the "fax" exten in the current context. This would remove your routing AGI form the picture, so I don't think you want faxdetect enabled on your lines. Maybe I'm misunderstanding, but to me, it seems like you're trying to detect a CNG tone and base your routing decision on that inside your AGI. Faxdetect will detect the CNG tone after the call is answered and automatically route for you. It's not the kind of thing you want to set on a call by call basis. If you're looking to detect a CNG tone inside your AGI, I'm not sure what mechanism is available for that. --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to lookup a call
On Wed, Nov 7, 2012 at 7:27 AM, Jerry Geis wrote: > I am using 1.4.43 currently. > > I am using the AMI to originate a call over a SIP Trunk to my cell > XXX506. works fine. > when the call is active I do a "core show channels concise" and I get: > > How do I "lookup" my call so I can "hangup" the call at a later time. > > Since you're using AMI to originate the calls, you should then also be able to add an ActionID to the originate command. You should then be able to lookup the call in AMI using the ActionID. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] dahdi dummy
If I remember correctly, dahdi dummy was removed and the functionally added by default when you load dahdi with no TDM cards installed. I could be wrong though. What do you need dummy for? Thanks, --Warren Selby, dCAP On Oct 23, 2012, at 10:28 AM, Jerry Geis wrote: > I need to use the dahdi dummy driver. > Its not being compiled at this time. > > When I go into tools subdirectory under dahdi-linux-complete-2.4.1 > and do make menuselect all I get is > CC="" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect > CONFIGURE_SILENT="--silent" nmenuselect > make[1]: Entering directory > `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' > make[1]: Nothing to be done for `nmenuselect'. > make[1]: Leaving directory > `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' > CC="" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect > CONFIGURE_SILENT="--silent" gmenuselect > make[1]: Entering directory > `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' > make[1]: Nothing to be done for `gmenuselect'. > make[1]: Leaving directory > `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' > make[1]: Entering directory > `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools' > Terminal must be at least 80 x 27. > menuselect changes NOT saved! > make[1]: Leaving directory > `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools' > > How can I get the dahdi_dummy.c driver compiled? > > > > 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] CDR Unanswered calls
On Fri, Oct 5, 2012 at 4:51 AM, Shanavaz E A wrote: > Hi, > > No replies until now. Some one please help... There must be some people > who are using it... > > Thanks > > > Can you provide an example of what you expect it to be doing (from the old version) and what it is doing now (from the new version)? I'm talking examples of the table rows in question. Is it recording the call, just labeling it answered instead of unanswered? I've never seen asterisk simply not record a call in whatever CDR backend you're using, regardless of disposition. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Case-sensitivity of Dialplan variables.
On Mon, Oct 1, 2012 at 4:15 PM, Mark Michelson wrote: > Hi! > > I've been confronted with an interesting issue to resolve. The > issue is located here: > > > So respond here and let me know what you think. I got a couple of replies > on the -dev list and they said that this would be good to put out on the > -users list too. > > Mark Michelson > My vote is to maintain the case sensitivity as the way it is now - user generated variables are case-insensitive, and asterisk-generated variables are case sensitive. I think breaking the existing behavior would be causing more problems than it solves. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to remove the call waiting tone without disabling callwaiting?
On Mon, Oct 1, 2012 at 10:03 AM, Niccolò Belli wrote:
> Hi,
> The call waiting tone is very annoying (you hear nothing while it plays
> the beep). I need callwaiting because of the queues (the phone has to ring
> as soon as you hangup) but I want to remove the beep on my dahdi channels,
> how can I do?
>
> Thanks,
> Niccolò
> --
> http://www.linuxsystems.it
Niccolo,
This is what I did for one of my clients. They had a very busy queue, and
were getting annoyed with the Call Waiting beeps. To resolve this, we
changed the method for contacting the agents to Local Channels. The local
channel would then do a check (using the GROUP() function) and see if it
was already in a call or not, and if it was, it would delay sending the
call to that agent. It would then try again after a certain amount of time
had passed.
The agents are added to the queue dynamically using
AddQueueMember(${queue-name},Local/${agent-exten}@agent-callsSIP/${state-exten}).
We would load the appropriate variables in the preceding dialplan.
Here's the snippets from extensions.conf:
[agent-calls]
;Context to dial agents when calls come into their queues
exten => _,1,Wait(1)
exten => _,n,Set(GROUP()=${EXTEN}-calls)
exten => _,n,GotoIf($[${GROUP_COUNT(${EXTEN}-calls)} > 1]?wait_longer)
exten => _,n,Dial(SIP/${EXTEN})
exten => _,n,GotoIf(${DIALSTATUS}=UNAVAILABLE?wait_longer)
exten => _,n,Goto(1)
exten => _,n(wait_longer),Wait(15)
exten => _,n,Goto(1)
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Re: [asterisk-users] QUEUEHOLDTIME always zero
The idea is a busy call center that takes calls all day long should be able to
determine their average wait / hold time over the course of the day. It's a
metrics thing, not a live data feed. It doesn't really become useful until
you've had several live calls, and then it's only useful if you've got
predictable call times.
For example, one of my clients has a customer service call center. Each of
their calls are usually handled in less than 5 minutes. If their average hold
times start spiking higher than that, they look at possibly increasing staff
(among other things), because the idea is to not make the customer wait very
long.
Now, another customer runs a tech support hotline. These calls can take
anywhere between a simple three minute password reset call to a 2 hour
adventure to track down some hidden issue. This customer doesn't look at hold
time metrics, because they end up all over the place.
If you want a good, in depth look at metrics related to your queues, I would
suggest giving Queuemetrics an evaluation. Queuemetrics is a program that
analyzes your queue_log file and generates both live data as well as historical
reports. Both of the customers I've listed above utilize Queuemetrics and they
both love it. The licensing is very reasonable for the market and they offer
free evaluations as well.
Thanks,
--Warren Selby, dCAP
On Sep 27, 2012, at 1:02 PM, Mitch Claborn wrote:
> Warren - that coincides with what I am seeing. I guess it made sense to
> someone, but it is not terribly useful to me.
>
> mitch
>
>
> On 09/27/2012 11:22 AM, Warren Selby wrote:
>> On Thu, Sep 27, 2012 at 9:15 AM, Mitch Claborn > <mailto:mitch...@claborn.net>> wrote:
>>
>>Satish I believe you have the answer. See output below, where I
>>have 1 call answered and 1 in the queue. Unfortunately, the average
>>wait time is very inaccurate. These two calls where placed within
>>seconds of each other. The one still in the queue has a wait time
>>of 4:10, so the average should be about 4 minutes.
>>
>>
>> -- Executing [812@LocalSets:1]
>>NoOp("SIP/08000F3BE07C-__000e", "queue status") in new stack
>> -- Executing [812@LocalSets:2]
>>Set("SIP/08000F3BE07C-__000e", "LOGGEDIN=1") in new stack
>> -- Executing [812@LocalSets:3]
>>Set("SIP/08000F3BE07C-__000e", "READY=0") in new stack
>> -- Executing [812@LocalSets:4]
>>Set("SIP/08000F3BE07C-__000e", "WAITING=1") in new stack
>> -- Executing [812@LocalSets:5]
>>Set("SIP/08000F3BE07C-__000e", "STUFF=0") in new stack
>> -- Executing [812@LocalSets:6]
>>Verbose("SIP/08000F3BE07C-__000e", "waiting: 1 calls in queue: 1
>>avg hold: 58 logged in: 1 ready: 0") in new stack
>>waiting: 1 calls in queue: 1 avg hold: 58 logged in: 1 ready: 0
>>
>>
>>asset333*CLI> queue show sales
>>sales has 1 calls (max unlimited) in 'rrmemory' strategy (58s
>>holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
>>Members:
>> SIP/mlcx500 (dynamic) (In use) has taken no calls yet
>>Callers:
>> 1. SIP/mlcx450-0003 (wait: 4:10, prio: 0)
>>
>>
>> That's because the call is still on hold. Once the call is answered,
>> the avg hold time will update again. It's an average of how long the
>> answered calls had to wait, not an average of all current calls waiting
>> on hold. At least, that's my understanding of the issue...
>>
>>
>> --
>> Thanks,
>> --Warren Selby, dCAP
>> http://www.SelbyTech.com <http://www.selbytech.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
>>
>
> --
> _
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>
> asterisk-users mailing list
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] QUEUEHOLDTIME always zero
On Thu, Sep 27, 2012 at 9:15 AM, Mitch Claborn wrote:
> Satish I believe you have the answer. See output below, where I have 1
> call answered and 1 in the queue. Unfortunately, the average wait time is
> very inaccurate. These two calls where placed within seconds of each
> other. The one still in the queue has a wait time of 4:10, so the average
> should be about 4 minutes.
>
>
> -- Executing [812@LocalSets:1] NoOp("SIP/08000F3BE07C-**000e",
> "queue status") in new stack
> -- Executing [812@LocalSets:2] Set("SIP/08000F3BE07C-**000e",
> "LOGGEDIN=1") in new stack
> -- Executing [812@LocalSets:3] Set("SIP/08000F3BE07C-**000e",
> "READY=0") in new stack
> -- Executing [812@LocalSets:4] Set("SIP/08000F3BE07C-**000e",
> "WAITING=1") in new stack
> -- Executing [812@LocalSets:5] Set("SIP/08000F3BE07C-**000e",
> "STUFF=0") in new stack
> -- Executing [812@LocalSets:6] Verbose("SIP/08000F3BE07C-**000e",
> "waiting: 1 calls in queue: 1 avg hold: 58 logged in: 1 ready: 0") in new
> stack
> waiting: 1 calls in queue: 1 avg hold: 58 logged in: 1 ready: 0
>
>
> asset333*CLI> queue show sales
> sales has 1 calls (max unlimited) in 'rrmemory' strategy (58s holdtime, 0s
> talktime), W:0, C:0, A:0, SL:0.0% within 0s
>Members:
> SIP/mlcx500 (dynamic) (In use) has taken no calls yet
>Callers:
> 1. SIP/mlcx450-0003 (wait: 4:10, prio: 0)
>
>
That's because the call is still on hold. Once the call is answered, the
avg hold time will update again. It's an average of how long the answered
calls had to wait, not an average of all current calls waiting on hold. At
least, that's my understanding of the issue...
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Re: [asterisk-users] MySQL Query : Calls Answered for < 5 sec
On Fri, Sep 14, 2012 at 11:33 AM, RSCL Mumbai wrote: > @Raj > > I tried your query and variation by using replacing duration with billsec. > In both cases, I get results including disposition "NO ANSWER" > If you don't want the "NO ANSWER" disposition, add an AND NOT DISPOSITION = 'NO ANSWER' to your query. This is all pretty basic SQL Query writing, not specific to asterisk... -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Digium AEX410, MTNL Mumbai Caller-ID problems
On Fri, Sep 14, 2012 at 9:02 AM, Raj Mathur (राज माथुर) < r...@linux-delhi.org> wrote: > So if there's a good chance that the latest Asterisk and Dahdi packages > will give better results in testing or might actually solve the problem, > I'll be glad to compile from source. If not, then perhaps it's not > worth polluting a production box with locally-compiled packages. > > Try adding a Wait(2) between your NoOp and your Verbose lines. I don't know about your telco, but sometimes the CID is not sent with the first ring, and you have to add a Wait(2) to grab it. You may even want to call your upper level support at your telco and ask them how and when they send your callerid information... -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Help with GotoIf Command
On Wed, Sep 5, 2012 at 4:30 AM, David Klaverstyn wrote:
> Hi All,
>
> ** **
>
> For some reason I can’t get this GotoIf statement to work. Even if the
> name and number are the same it jumps to line 3. I’ve tried with and
> without the quotes around each variable.
>
> ** **
>
> exten => s,1,GotoIf($["${CALLERID(name)}" = "${CALLERID(num)}"]?:3)
>
> exten => s,2,NoOp(they are the Same)
>
> exten => s,3,NoOp(they are different)
>
> ** **
>
>
>
You need to verify if the {CALLERID(num)} can actually match what your
{CALLERID(name)} looks like. More than likely, the (num) has some sort of
brackets around it, such as < >, or perhaps it's starting with a +. You
can try to use the FILTER function on it, to strip away any additional
characters that you don't want to see or try to match on.
exten => s,1,GotoIf($["${CALLERID(name)}" =
"${FILTER(0123456789,${CALLERID(num)}}"]?:3)
That is, if you're just looking for numeric callerid. If you also want to
account for extra characters, you can add those to the first part of the
filter.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Re: [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API
On Fri, Aug 31, 2012 at 9:36 AM, Shitian Long wrote: > Hello. > > I am trying to use Asterisk Manager API query data from realtime. From > Asterisk CLI, we could use > realtime load > query realtime > it would have response like > >Column Name Column Value > > id 1 > mykey content >myvalue value > > I am wondering how I could make this type of query from Manager API. > > > Thanks for your time in advance. > > Is there a specific reason you want to access the realtime data through the Manager API and not directly from the database itself? It seems like the Manager API would add an extra layer to whatever you're trying to accomplish. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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 trunk failing to register causes sip phones to become unreachable
On Aug 23, 2012, at 10:30 AM, John Cahill wrote: > I have only seen this problem when using sipgate SIP trunks which actually > "register". If the ADSL connection goes down that the sip trunk uses, the sip > phones registered locally become unreachable. This happens on any 1.6.x or > 1.8 version of asterisk I've tried. Is there a work around that doesn't > involve putting an opensips server between the asterisk server and the sip > trunk? This is a common issue that I've seen many times. The problem has to do with DNS cache look-ups and timeouts. What typically solves it for me is to install a local cacheing-only DNS server on the asterisk box and point the resolvers on the asterisk box to itself. This will only solve the issue of an internet outage causing the sip phones to stop working, and only for as long as the local cache stays relevant. If there is a power outage that takes out both the asterisk server and the internet, and your asterisk box comes up but your internet doesn't, this won't work. -- Thanks, Warren Selby, dCAP -- _ -- 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] confbridge
On Mon, Aug 13, 2012 at 12:13 PM, Jerry Geis wrote: > I am getting a "beep beep beep" (like a busy or hangup sound) when I am > using my > AGI to start up a conf. (did not happen with Meetme). > > The confbridge works, but the beep beep beep is mixed in with the audio. > I have turned off every sound in the confbridge.conf file. > > How can I find out where this beep, beep beep is coming from and turn it > off??? > In addition to the two other excellent troubleshooting questions that have been asked so far, could you also please share with us the CLI output during the call, from the time you start to the time it all ends? Is this happening on even just adding one person to the confbridge? Does it happen when you add more than one person to the bridge? Does everyone hear the beep, or is it only in the mixed audio? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to control just one phone within current CCM?
You will need to setup a SIP trunk between the asterisk server and the CCM server. Then in your asterisk config, you'll need to direct any extensions that are handled by the CCM server to that trunk. You'll also need to configure the CCM server to send calls to the specific extension through the asterisk sip trunk. -- Thanks, Warren Selby, dCAP On Aug 9, 2012, at 6:05 PM, Eduardo Giacoman wrote: > Danny, thanks for your input... > > Can you tell me if I am wrong with the following or give me a brief guide of > what to look at? > I was planning on using Asterisk + chan_sccp to control the VOIP phone. > Asterisk will NOT replace the current CCM/PBX at work, it will have just one > phone but in a way that I still can call extensions at work from asterisk. > > I can point the phone to another TFTP server with the proper SEM file, etc. > so it will talk to Asterisk. But after that, if I call an internal extension > at work will it find it or I have to do something else? I am a little > confused because I think that since the phone is not pointing anymore to the > CCM at work, it won't find any other internal extensions, just the ones I may > add to the asterisk setup. > > Excuse me I have very basic voip knowledge. > > > On Thu, Aug 9, 2012 at 3:25 PM, Danny Nicholas wrote: > This shouldn’t be a problem. Asterisk is basically “flavor-blind” as to what > type and quantity of phones you put on it. > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jorge Díaz > Sent: Thursday, August 09, 2012 4:21 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Asterisk to control just one phone within current > CCM? > > > > > > Hi, > > I have used basic Asterisk as a PBX controlling few extensions. My question > is simple, at work there is an existing Call Manager/PBX and all which > manages all the extensions for SCCP VOIP phones. Can Asterisk be used to > manage just 1 VOIP phone and still can make internal calls to the other > extensions? > > Thanks, > Jorge > > > -- > _ > -- 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] Voicemail Emails
On Fri, Jul 20, 2012 at 12:53 PM, Josh Hopkins wrote: > Has anyone been able to make an html template for the voicemail emails. We > would love to customize them beyond just plain text. We have dome some > Google searches and have not been able to come up with much. > > ** ** > > I believe that Switchvox has customized the voicemail email into html. > Has anyone ever tried this? Thanks, > > /Josh > > > What about changing 'mailcmd=' to a shell script that rewrites the email in the format you want before sending it to sendmail? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] FreePBX: using context other than the default context and the generation for the configuration
On Wed, Jul 11, 2012 at 4:56 PM, bilal ghayyad wrote:
> Fine, did you read the question well and understand about what I am asking?
>
>
Perhaps I did not understand what you were asking. I thought you were
wanting to do something custom per extension (in the case of my example,
the "something custom" was control outbound call access to either local
only or local and long distance, etc. You can figure out you're own
"something custom"), but still have all the calls have all the standard
FreePBX features that you only get when using the [from-internal] context.
In my example, the extensions are in the 2XXX range, and they would either
have a context of [custom-local-only] or [custom-long-distance], depending
on what you wanted to allow that extension to dial.
To break down my example:
[custom-local-only] --> The name of our custom context. It could be
anything you want, as long as it's in square brackets
exten => _281NXX,1,Verbose(Outbound call from local-only context) -->
This step is purely informational, it has no bearing on CDRs or anything
else...it's just a useful step for debugging. I tend to do this for
everything, it's the same as some people use the "NoOp()" command to have
debugging information in their CLI output.
same => n,Goto(${EXTEN},from-internal,1) --> This step sends the call to
the [from-internal] context and handles it exactly as if you weren't using
any custom call controls. In my example, however, it will only go there if
it meets the criteria of matching the pattern (in other words, the call
would have to be placed to a number that matches the _281NXX pattern).
"same => n" is a shorthand way of writing "exten => _281NXX,n". It was
added in around 1.6 I think, I'm not entirely sure.
exten => _2XXX,1,Verbose(Internal extension-to-extension call) --> Again,
this is purely an informational step, useful for debugging. It can be
skipped or expanded as you see fit, it has no bearing on CDR records or
anything else, other than CLI output.
same => n,Goto(${EXTEN},from-internal,1) --> This does the same as the
previous example, however it will only go to the [from-internal] context if
the pattern that was dialed matches _2XXX. This is assuming you're using
internal extensions in the range of _2XXX. You can change this to whatever
works for you.
[custom-long-distance] --> another custom context, this time it allows
long distance NANPA calling as well as local and internal calls
exten => _1NXXNXX,1,Verbose(Outbound call from local and long-distance
context) --> I hope you're seeing the pattern by now. This is simply a
useful debugging step, with no bearing on anything else.
same => n,Goto(${EXTEN},from-internal,1) --> The call passes into the
[from-internal context if it matches the pattern of _1NXXNXX, a typical
NANPA long distance call.
include => custom-local-only --> include the local dialing context that
way we don't have to duplicate any code that we've previously written,
mostly useful for the internal extension dialing.
So you can see, the Verbose() statement has no bearing on CDR's what so
ever. I wasn't aware that FreePBX used any kind of custom CDR database, I
assumed it was simply using the asterisk CDR database, where any call
through the system generates a CDR. Since someone else had mentioned that
they did not get any CDR logging or any of the other FreePBX features
without making the extension have a context of [from-internal], I was
showing how to do simple things like local and long-distance access control
in the extensions_custom.conf file, and then sending the call into the
default [from-internal] context. What I provided was mostly just supposed
to be an example that you could build off of. You don't have to use
Verbose() if you don't want to, that's just something I've grown accustomed
to doing.
I'm by no means an expert at FreePBX. If you find that using custom
contexts are not helping in you situation, perhaps you can expand on what
the actual issue is that you're experiencing, and we can try to help
troubleshoot from there.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash
On Tue, Jul 10, 2012 at 12:39 PM, Tim Nelson wrote: > Most of the predone projects (Elastix is my favorite at the moment) > include some sort of endpoint manager that will generate configs for your > phones. I'm not sure specifically on Cisco phones, other than they are a > huge PITA in general. The system just needs a TFTP server installed, and > the phones pointed to it (manually, or by using DHCP option 66). > > Speaking from experience - I have a client that has Trixbox (setup by a previous phone consultant) that also uses all 7960 and 7940 phones. Trixbox works perfectly with the phones - it uses an endpoint manager that automatically discovers and configures the phones, even setting the background display. I think it's a special add-on module that was added to this installation, I'm not entirely sure. I can check if you would like me to pursue this? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Flowroute: howto set outbound callerid (ast 1.4)?
On Tue, Jul 10, 2012 at 12:34 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:
> Thank you for your feedback Warren. I removed the outbound name but still
> get random numbers & "VOIP CALLER" on outbound calls. Googling I tried some
> more:
>
> SipAddHeader(P-Asserted-**Identity: **)
> SipAddHeader(P-Asserted-**Identity: 19995551212)
> SipAddHeader(P-Preferred-**Identity: **)
> SipAddHeader(P-Preferred-**Identity: 19995551212)
>
> But none of them work. So unless someone has the magic incantation howto
> make this work I'll open a ticket with flowroute.
>
>
I use Flowroute. My outbound callerID is set as follows:
[outgoing]
exten => _X.,1,Verbose(Outound call from ${callidnum} to ${EXTEN} on
${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)})
exten => _X.,n,Set(CALLERID(num)=${callidnum})
exten => _X.,n,Goto(outgoing-dial,${EXTEN},1)
[outgoing-dial]
exten => _NXXNXX,1,Dial(SIP/1${EXTEN}@flowroute)
exten => _1NXXNXX,1,Dial(SIP/${EXTEN}@flowroute)
${callidnum} is a variable from my SIP peer (setvar=callidnum=7133437300).
This always passes my proper phone number when I make outbound calls.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?
On Tue, Jul 10, 2012 at 11:45 AM, Patrick Lists < asterisk-l...@puzzled.xs4all.nl> wrote: > On 10-07-12 18:29, Alex Balashov wrote: > >> SIPAddHeader() comes to mind. :-) >> > > Yup I got that far :) I tried things like (with correct name & number): > > exten => _1ZX,1,SipAddHeader(P-**Asserted-Identity: "Global > Minties Corp" **) > > But that did not work as flowroute always sends "VOIP CALLER" and a ton of > different numbers on outbound calls. So I guess I am doing something wrong > but I can't figure out what. > You can't* set the outbound name. That's defined in the national caller id name database that the receiving phone company dips into. As far as I know, Flowroute does not add entries to this database, nor do they dip it when you receive a call to pass the caller ID name on inbound calls. Other providers do. * - You may be able to set it if you're calling other users on the Flowroute network, I'm not sure. But in general, once your call leaves the Flowroute network, the only way to get the CNAM info is from a CNAM dip to the national database (I don't recall the actual name). -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] channel not available and jump to next group channels
On Tue, Jul 10, 2012 at 9:04 AM, mahesh katta wrote:
> Hi list,
>
> TRUNkA=Dahdi/g0 {g0=1-15,17-31}
> TRUNKB=Dahdi/g1 {g1=32-46,48-62}
>
>
>
> I have 2 gsm channel banks its E1 connection , its connected to server. I
> define this 2 different trunks.
> for example like TrunkA,TrunkB.
> TRUNKA connected 1 gsm channel bank and TRUNKB connected 2 gsm channel
> bank. if TRUNKA channels are not available its needs to automatically
> TRUNKB. How its possible to do with Dialplan without macros.
>
>
>
exten => _XX,n,Dial(${TRUNKA}/${EXTEN})
exten => _XX,n,Dial(${TRUNKB}/${EXTEN})
Swap _XX for whatever your outbound extensions would be...
--
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--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Re: [asterisk-users] Rookie / sip and extensions
On Sat, Jul 7, 2012 at 1:48 PM, Thomas Perron wrote:
> extensions.conf
> [globals]
>
> ;
> ;
> [incoming]
> ;
> ;exten=> s,1,Goto(125010155_incoming)
> ;
> ;[125010155_incoming]
> exten => s,1,Answer
> exten => s,n,Dial(SIP/16175551212)
>
>
> sip.conf
> [general]
> ;register => 125010155:funnyti...@sip3.voipvoip.com/125010155
> register => 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11
> ;
> [incoming]
> username=125010155
> type=peer
> secret=funnytiger
> nat=auto
> insecure=invite,port
> host=69.90.209.11
> fromdomain=69.90.209.11
> dtmfmode=rfc2833
> context=incoming
> allow=g729
> allow=ulaw
> allow=alaw
> allow=ilbc
> srvlookup=yes
>
If these are actual copy / pastes from your extensions.conf and sip.conf
files, with just passwords changed, your issue probably comes from your
over abundant use of semi-colons (";") at the start of several lines. The
semi-colon indicates a comment line to the asterisk parser, and thus isn't
parsed. Your only exten => line in your [incoming] context is commented
out, as is the name of your [125010155_incoming] context, and your first
register statement.
Set the CLI verbosity to 6 (core set verbose 6) and then try to dial in
again, and paste the failed output as a response to this email, and we can
diagnose from there.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration
On Fri, Jul 6, 2012 at 12:11 AM, SamyGo wrote:
> umm Warren, yes including from-internal is the way of getting all the
> features,,,but in my experience the calls going out using the dialplan
> script we manually enter in our custome context don't get inserted into the
> FreePBX CDR and recording stuff !!
>
Okay, if you're writing custom dialplan to control outbound calling, but
you want to utilize the FreePBX standard features, without using custom
modules, you can do something like the following, adjusting for your
specific situations of course:
[custom-local-only]
; local NANPA calling for area code 281
exten => _281NXX,1,Verbose(Outbound call from local-only context)
same => n,Goto(${EXTEN},from-internal,1)
; extension-to-extension (internal) calling, assuming 2XXX internal
extension plan
exten => _2XXX,1,Verbose(Internal extension-to-extension call)
same => n,Goto(${EXTEN},from-internal,1)
[custom-long-distance]
; long distance NANPA calling, dial a 1 to dial anything outside of a local
number
exten => _1NXXNXX,1,Verbose(Outbound call from local and long-distance
context)
same => n,Goto(${EXTEN},from-internal,1)
; allow local calls also, without having to dial a 1
include => custom-local-only
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration
On Thu, Jul 5, 2012 at 5:20 PM, bilal ghayyad wrote: > Hi All; > You can get modules to do what you're looking for, but if you really want to make a custom context but still have all the available features of the default context, you can add the following at the end of your custom context: include => from-internal Be sure to do all of this in extensions_custom.conf, that way it doesn't get overwritten whenever you issue a reload in the GUI. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] basic sip quesiton
On Jul 4, 2012, at 9:20 PM, Thomas Perron wrote: > What am I missing please? sip show registry shows that I am registered. What are you missing? A question, or at the least, a description of whatever problem you are having? Also, a meaningful subject that somewhat talks to the content of your question. -- _ -- 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] Voicemail attachment format
On Mon, Jun 25, 2012 at 9:23 AM, khalid touati wrote: > Hi All, > I have a simple urgent question that I couldn't find the answer yet, can > we customize the voicemail attachment format *per user* in asterisk *1.2 > *(like > all receive wav attch but one or two users receive attch in gsm format)? if > yes can you show me how please? > > I don't think that was an option in 1.2, but I haven't used 1.2 in so long I may be off. Hopefully one of our resident 1.2 luddite's will see this and have a more definitive answer for you. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] ext-local and from-did-direct-ivr, how to change them?
On Sun, Jun 24, 2012 at 6:20 PM, bilal ghayyad wrote: > Hi All; > > Using the FreePBX, after I added the extension from the GUI, I discover > that it is automatically added in the extensions_additional.conf in the > context [ext-local] and [from-did-direct-ivr] > > How I can change these context name? I need to determine this. How? > What are you trying to do? FreePBX uses these context names for a reason, it's usually not a good idea to just change them. However, depending on what you want to do, you may be able to do something in the extensions_custom.conf file... -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] 10.5.0: channel name inserted as callerid number ??
On Wed, Jun 20, 2012 at 3:21 PM, sean darcy wrote: > [home_outgoing] > type=friend > transport=tcp > secret=<> > fromuser=office_incoming > host=dynamic > disallow=all > allow=ulaw > It's because you're using "fromuser" as your username setting. This will overwrite your CallerID settings. Instead try using "defaultuser". -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] GoogleVoice woes
On Wed, Jun 20, 2012 at 12:30 PM, Chris Gentle wrote: > On Wed, Jun 20, 2012 at 12:14 PM, Warren Selby wrote: > >> As you said, GV and asterisk integration is unstable at best. I haven't >> worked with it in a while, to be honest. But, with all that being said, >> I'm not opposed to popping my GV test box back online and helping to >> troubleshoot. Why don't you start by giving us the contents of the >> gtalk.conf and jabber.conf files, the incoming dialplan snippet from >> extensions.conf for the google voice calls, and the CLI output with >> verbosity set to at least 6 of both a successful incoming call and a failed >> incoming call. There's a debug option of jabber also, if you can have that >> enabled when you make the calls, that would be very helpful as well. >> > > For the number that is not working there is no jabber debug output and > nothing shows up on the console. That leads me to believe that Google > isn't sending the call to my box at all. > > I remember seeing this when I had my gmail web client open - the call would try to ring in the web client instead of the asterisk box. It was difficult to tell this was the case, because I never really noticed the ring on the web interface until a few hours into debugging the issue. However, closing the web app made it ring into the asterisk box. I'm assuming you don't have the web client open on a computer somewhere when you attempt this? Might be something to check out. > Here's my gtalk: > > [general] > context=incoming > allowguest=yes > bindaddr=0.0.0.0 > > [guest] > disallow=all > allow=ulaw > context=from-googlevoice > connection=tcg-asterisk > > Looks pretty similar to my notes on what I had for my own setup, I'll need to find the config I used on the old box to confirm. > And my jabber.conf: > > [general] > autoregister=yes > > [tcg-asterisk] > type=client > serverhost=talk.google.com > username=my_usern...@gmail.com/Talk > secret=deleted > port=5222 > usetls=yes > usesasl=yes > statusmessage="Connected via Asterisk" > timeout=100 > > [seg-asterisk] > type=client > serverhost=talk.google.com > username=my_wifes_usern...@gmail.com/Talk > secret=deleted > port=5222 > usetls=yes > usesasl=yes > statusmessage="Connected via Asterisk" > timeout=100 > > This looks pretty close to mine, the only thing I can think to do here would be to add a "status=available" option to both user definitions, and also maybe add a buddy= option, and add the name / email of another gmail user account that you can open in the actual gtalk client. This will let you see if these definitions are even coming online at all? If none of this helps, let me know and I'll find my old GV box and set it up again. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] GoogleVoice woes
On Wed, Jun 20, 2012 at 11:20 AM, Chris Gentle wrote: > I have two GV numbers. Both are configured to send calls to my Asterisk > 1.8.13.0 box using the Google chat interface. At one time I had both > working with Asterisk. Now, for whatever reason, one of them has stopped > sending incoming calls to my asterisk box and instead just rolls to GV > voicemail. The other number continues to work fine. One is associated > with my wife's google account and the other is mine. I've compared our > account settings in Google and can't find any differences. Running "jabber > show connections" shows connections to each account. I know about all the > instabilities with GV and Asterisk but if one number works the other one > should too. I'm sure this is something simple, probably a Google account > setting that I can't find. Can anyone think of something else I might > could check? > As you said, GV and asterisk integration is unstable at best. I haven't worked with it in a while, to be honest. But, with all that being said, I'm not opposed to popping my GV test box back online and helping to troubleshoot. Why don't you start by giving us the contents of the gtalk.conf and jabber.conf files, the incoming dialplan snippet from extensions.conf for the google voice calls, and the CLI output with verbosity set to at least 6 of both a successful incoming call and a failed incoming call. There's a debug option of jabber also, if you can have that enabled when you make the calls, that would be very helpful as well. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Missing voicemail prompt beginning
Please excuse the top post, I'm on my phone. Before we have a better idea of what's going on, please provide the dialplan snippet that the call is using as well as the cli logs of the calls where you hear the whole prompt and where you only hear part of the prompt. Also, if you can clarify the infrastructure setup as well, that would be helpful. Thanks, --Warren Selby, dCAP On Jun 17, 2012, at 11:25 AM, Stefan at WPF wrote: > Hmm, I tried calling myself (the asterisk voicemail) from another SIP > provider, same problem. What always works reliable is using and calling the > voicemail of my SIP Provider (Sipgate) from my mobile phone, I reliably hear > the complete prompt. Doesn't this contradict the assumption that the problem > is on the mobile phone side? > > 2012/6/17 Doug Lytle > Stefan at WPF wrote: > Which end do you mean with "channel not answered"? The asterisk > > The Asterisk side. If the answer didn't fix the issue, then my guess is that > it's on the cellular provider's side (Which isn't unheard of). > > > 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 -- _ -- 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] Voicemail: Tell external number instead of internal number
On Sat, Jun 16, 2012 at 4:23 PM, Stefan at WPF wrote: > Hello, > > I have an internal extension, e.g. 1005 which is being called from an > external/public number like 123456789. Now when it comes to the spoken > voicemail information it says something like "number 1000 not available", > however it should say "number 123456789 not available". How can I configure > this? I already googled and I guess this is really easy, but I just > couldn't figure out how to do this ): So thanks for any hint :-) > In your voicemail.conf, configure the mailbox as 123456789 => 1234,username,emailaddy,pager,options, and not as 1005 => 1234,username,emailaddy,pager,options And then in your extensions.conf you would call the Voicemail app like so: exten => 1005,1,Verbose(Incoming call to 123456789 transfered to SIP phone 1005) exten => 1005,n,Dial(SIP/1005,30) exten => 1005,n,Verbose(No answer, going to voicemail for 123456789) exten => 1005,n,Voicemail(123456789@default,u) exten => 1005,n,Hangup() -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Need queue name in CDR
On Tue, Jun 12, 2012 at 10:38 PM, Pratik Shrestha wrote:
> Dear All,
>
> I am making asterisk report using CDR values given by asterisk.
>
> I have queues which consist of multiple members (extension). Also, an
> extension may be in multiple queues. So, I want CDR to record the
> name/number of queue from which the call was originated.
>
> E.g.
> *Channel* * DestinationChannel*
> * Src* *
> Destination
>*
> SIP/KOT-000c Local/102@from-queue-6a84;1
> 0856511524 (first
> line in CDR)
> Local/102@from-queue-6a84;2 SIP/102-000e
>0856511524 102
> (second line in CDR)
>
>
> In above example, is a queue and 102 is an extension which is member
> to that queue. So call comes from 0856511524 and goes to queue first
> and queue routes call to 102 extension. So what I need is when the queue is
> routed to extension 102 (in the seconds line), I want to show the queue
> () also. I know that I can track the queue by comparing Destination
> Channel of queue(first line) with Channel of extension (second line). But
> this will make my query very long and hard.
>
> Please help me. I am still new to asterisk.
>
>
While I agree with Lenz about using one of the existing tools out there to
analyze queue logs (his Queuemetrics is a very good tool, I would
definitely recommend it!), if all you really want is queue name in the CDR
fields, you can do that with a simple Set command in your local channel
that dials your agents using func CDR:
exten => agentcall,1,Set(CDR(queue)=${queuenum})
This will create a new field in your CDR called "queue" and will populate
it with the result of the channel variable ${queuenum}, which you should
set before you enter the queue. If you're using MySQL for your CDR
storage, I believe you have to create the column first for the new field.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Re: [asterisk-users] Polycom Caller ID
On Tue, Jun 12, 2012 at 4:15 PM, Jon Caum wrote: > Hello, > > I have an issue I remember seeing a while ago and forgot to investigate > further. Now it is turning into an issue and will need to be resolved. A > customer has Polycom 335 phones (and a couple Soundstation 6000s), and when > an extension is calling out, the screen on the 335 shows the company's > internal CID number instead of the person they are dialing. This also > applies to receiving calls - the internal CID is displayed as opposed to > who was calling. > > I remember seeing something about connectedline issues with Polycom > phones, but I can't find the bug I had seen 6 months ago. Does anybody know > about this issue and what can be done to resolve? > > > Thanks! > How is your username defined in the sip.conf entry? I had this issue once before when I used "fromuser=" instead of "defaultuser=" for each phone. Almost the exact same issue you're reporting... -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.10
On Mon, Jun 11, 2012 at 6:12 PM, motty.cruz wrote:
> Hello,
> How to change ring tone on interncal call? I'm using Centos 5.8 Asterisk
> 1.8
>
> exten =>
> 666,1,SIPAddHeader("Alert-Info:<http://1.2.3.4/ringtones/ghost.wav>")
> exten => 666,n,Dial(SIP/10)
>
> The above would not how to defirenciate from internal call or external
> call?
>
Just a thought, but maybe set a variable in your sip.conf for each internal
peer, and then check for that variable before you do the SipAddHeader
command (using an ExecIf statement).
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Re: [asterisk-users] Fax Server for Asterisk
On Tue, May 29, 2012 at 10:06 AM, Bakko wrote: > Any hint about email2fax? > > Thank you > > This can be handled natively by the HylaFax+ server. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Fax Server for Asterisk
On Tue, May 29, 2012 at 3:10 AM, Danny Dias wrote: > Hello, > > For those customers with only analog lines, who ask for fax2email and > email2fax, whats the most reliable solution available and tested with > Asterisk? > > Thanks > > I've been real happy with using HylaFax+ and Iaxmodem implementations. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] hangup not detected?
On Fri, May 18, 2012 at 12:00 PM, Justin Killen < jkil...@allamericanasphalt.com> wrote: > I have and automated call-in dispatch system where hundreds of people > call in daily for 2-3 minutes each. The extension is set up to get their > information, then text-to-speech the dispatch information (via odbc). It > then loops 5 times then ends the call. These calls are being handled by an > 8 port analog digium card. > > ** ** > > Sometimes though, I see calls via ‘core show channel dahdi/1-1’ that have > a time of > 16 hours. I’m not sure if this is a result of dahdi missing > the hangup, ODBC timing out, or TTS failing for some reason. When a > channel gets in this state, the call doesn’t seem to progress through the > dialplan, they always display the TTS line. Doing a ‘dahdi destroy channel > 1-1’ doesn’t seem to be effective – the only way I’ve been able to clear > the calls is to do a ‘dahdi restart’ and/or restart the asterisk service.* > *** > > ** ** > > For TTS I’m using cepstral with the Swift wrapper. > > ** ** > > Here is a snippet of my dialplan: > > ** ** > > Can you post the CLI output of a call that gets "hung"? I'd like to see where it's hanging on. Also, as a work-around to attempt to solve the symptom and not the underlying issue, you could maybe setup a cron job that runs once every ten minutes that checks for stale calls using AMI, and then hangs up any calls up that are over 10 minutes long? Using the AMI Hangup command? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Event response (AMI)
On Fri, May 11, 2012 at 8:31 AM, Matthew Jordan wrote: > > In your particular case, if I were writing a system that wanted to > associate > a created channel with an Originate Action, after I issue the Originate, > I'd listen for a NewChannel event. If that NewChannel event specified a > channel that was created in the context I specified and with a > technology/extension that I specified, I'd set that as the channel I just > asked to be created. From there on, subsequent events (VarSet, NewExten, > Hangup, etc.) that are associated with that channel will contain a Channel: > header with that value. > > Isn't that likely to cause race issues if for instance he Originates 30 calls all at the same time? I would think a better approach would be to set a unique channel variable for each originated call and track based on that? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)
Are you able to add a Wait(2) at all to the beginning of your incoming dialplan? A lot of missing callerID problems are because the callerID value gets sent after the initial call signaling comes in. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] hints and server-side DND (do not disturb)
On Wed, Apr 18, 2012 at 1:27 AM, Vieri wrote:
> Hi,
>
> Currently I'm using hints to determine SIP presence. As I understand it, a
> SIP extension can be labeled as busy, ringing, etc, based on a channel
> status. So a channel MUST be present. If it isn't then the extension is
> considered to be "available".
>
> If my statement is correct then is there a way to set the extesnion as
> "busy" even if there's no channel associated with this extension?
> eg. when an extension sets server-side DND (Do Not Disturb), it actually
> sets a boolean value in astdb. Whenever asterisk tries to route a call to
> this extension, it first checks this value. Obviously, there's no way I can
> use hints in this scenario, or is there? Is it possible to somehow create a
> "dummy" channel whenever an extension sets "server-side DND" (custom
> context) and delete it whenever it unsets it?
>
>
I've done something similar using "night-mode" type logic. All calls
coming into the system first do a check against the db to see if night-mode
is enabled or not. If it is, route calls to voicemail, if it's not, route
calls normally. You can also use custom hints to set busy lamps on
appropriate phones. The receptionist can then hit the monitored button on
her phone to turn on or turn off night-mode. Here's some snippets from
existing dialplan...
[mainmenu]
; Main IVR
exten => s,1,Verbose(Inbound call to main number - checking if night mode
or normal)
exten => s,n,Set(NIGHTMODE=${DB(nightmode/enable)})
exten => s,n,GotoIf($["${NIGHTMODE}" = "1"]?nightmode)
exten => s,n,Verbose(Normal mode - Proceeding Normally)
exten => s,n,...
exten => s,n,...
exten => s,n,...
exten => s,n(nightmode),Verbose(Night mode - going straight to voicemail)
exten => s,n,Voicemail(@default,su)
exten => s,n,Hangup()
[internal]
; Night Mode
exten => *280,1,Answer()
exten => *280,n,GotoIf($["${DB(nightmode/enable)}" = "1"]?disable:enable)
exten => *280,n(enable),Verbose(Enabling night mode)
exten => *280,n,Set(DB(nightmode/enable)=1)
exten => *280,n,Set(DEVICE_STATE(Custom:lamp)=BUSY)
exten => *280,n,Playback(enabled)
exten => *280,n,Hangup()
exten => *280,n(disable),Verbose(Disabling night mode)
exten => *280,n,Set(DB(nightmode/enable)=0)
exten => *280,n,Set(DEVICE_STATE(Custom:lamp)=NOT_INUSE)
exten => *280,n,Playback(disabled)
exten => *280,n,Hangup()
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
--
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Re: [asterisk-users] Set variables from one asterisk ta a second.
On Sun, Apr 15, 2012 at 2:49 PM, Olivier CALVANO wrote:
> The CLI of the server two:
>
> srv2*CLI>
>-- Accepting AUTHENTICATED call from 172.20.8.1:
> > requested format = alaw,
> > requested prefs = (alaw|g729),
> > actual format = alaw,
> > host prefs = (alaw|g729),
> > priority = mine
> *[Apr 15 21:46:26] ERROR[31094]: pbx.c:2860 ast_func_read: Function
> $CALLERID not registered
> *
Change ${$CALLERID(num)} to ${CALLERID(num)}. One too many '$' signs in
Danny's examples. Be sure to change it for each instance of
${$CALLERID.}.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
--
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Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6: chan_dahdi? dahdi?
On Sun, Apr 15, 2012 at 3:48 PM, bilal ghayyad wrote: > Hi All; > > Is it normal if I used asterisk 1.4 and dahdi, then I will not find > chan_dahdi under /usr/lib/asterisk/modules? And I will not be able to type > dahdi commands (dahdi restart for example) in the asterisk CLI? > > Actually what I found only the following: > > app_dahdibarge.so app_dahdiras.so app_dahdiscan.so codec_dahdi.so > > So, it is available only with asterisk 1.8? > > Well, does this mean it is preferred to use zaptel with asterisk 1.4? > Did you compile asterisk with DAHDI support? i.e Did you install DAHDI, then run ./configure on Asterisk Source and then install? Or did you install asterisk first, then DAHDI? I've successfully used DAHDI with Asterisk 1.4, so there must be some issue. Please give us information about how you installed everything. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Dial Plan - Routing via Caller ID
On Apr 11, 2012, at 5:40 PM, wrote: > And your examples should work for 1.8.10 correct? > I just typed those out really quick, so there may be some syntax errors, but generally yes they should all work with 1.8.x. -- Thanks, Warren Selby, dCAP -- _ -- 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] Dial Plan - Routing via Caller ID
On Wed, Apr 11, 2012 at 4:11 PM, wrote:
> Here is an example.
>
> ** **
>
> Let’s say that I want to send all calls to a context that would answer the
> call via voicemail.
>
> Let’s say that I want to only right a SIP phone if calls cam from a
> particular Area Code (maybe the Area Codes in your state).
>
> Let’s say that I would want to send calls from a particular A/C and
> certain NNX’s to a particular sales group.
>
> ** **
>
> Does that help define the purpose of directing calls **from** different
> Area Codes and NNX’s?
>
>
You've got a few ways you can do this:
1 - In the dialplan with ex-girlfriend logic. You should be able to use
patterns with your ex-girlfriend logic matches, as so:
exten => 15558675309/_255NXX,1,Verbose(Calls to 867-5309 from area code
255 end up here)
exten => 15558675309/_256123,1,Verbose(Calls to 867-5309 from phone
numbers 256123 end up here)
etc.
2 - In the dialplan with GotoIf logic:
exten => 15558675309,1,Verbose(Call from ${CALLERID(num)} to 867-5309)
exten => 15558675309,n,GotoIf($["${CALLERID(num):1:3}"="255"]?areacode255)
exten => 15558675309,n,GotoIf($["${CALLERID(num):1:6}"="256123"]?num256123)
exten => 15558675309,n(areacode255),Verbose(Calls to 867-5309 from area
code 255 end up here)
exten => 15558675309,n(num256123),Verbose(Calls to 867-5309 from phone
numbers 256123 end up here)
etc.
3 - Outside the dialplan with an AGI that allows you many more conditional
logic choices (plus keeps your dialplan nice and clean):
exten => 15558675309,1,Verbose(Call from ${CALLERID(num)} to 867-5309)
same => n,AGI(route_by_clid)
In your AGI, you'll be most interested in the agi_callerid environment
variable and you can control where the call goes next using the SET CONTEXT
and SET EXTENSION agi commands, or simply EXEC a GoTo command (either way
works).
Ultimately, I would go with the AGI option, because that then allows you to
do things like use a database to store your routing information, use case
statements, create routing loops, etc. It's up to you though.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Apr 5, 2012, at 2:32 PM, Carlos Alvarez wrote:
>
>
> On Thu, Apr 5, 2012 at 12:11 PM, Eric Wieling wrote:
> Are you sure you are not referring to the "s" extension?
>
> Absolutely. Every time I discuss 's' priority on this list or the Asterisk
> IRC channel people tell me it either doesn't exist or is wrong, but it's a
> powerful under-utilized feature. It's at the core of initially routing calls
> on our system.
>
> Show an example of needing "s" as a priority.
>
> This is from our system, the asterisks have been used to obscure for privacy,
> they are numbers.
>
> exten => 1602889,n,Goto(starnetworks#main|s|1)
> exten => 1602400,s,Goto(starnetworks#extensions,9520,1)
> exten => 1480241,s,Goto(starnetworks#extensions,9766,1)
> exten => _X.,s,Goto(starnetworks#extensions|${EXTEN:7}|1)
>
>
I still don't understand what you would need this for. What version of asterisk
are you using? From voip-info.org, it says the s priority is used when
"different patterns may match at the same point in the extension and act
differently for them", but couldn't you basically do the same thing with
priority labels? How would you ever end up with different patterns matching at
the same point in an extension? Where is your priority 1?
--
Thanks,
Warren Selby, dCAP--
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Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Apr 5, 2012, at 1:23 PM, Carlos Alvarez wrote: > > > On Thu, Apr 5, 2012 at 11:00 AM, Eric Wieling wrote: > Priorities are not complicated. Each extension starts with priority 1, all > additional priorities are "n" and you ALWAYS end your extension with a > > > This isn't correct, there are many cases where you must use an 's' priority. > Our system simply couldn't function without it. You're think of EXTENSION 's', not PRORITY. Priority is the order the call goes through the dial plan. Extension is the part of the dial plan you're traversing. Priority will always be either a number or an 'n'. exten => EXTENSION,PRIORITY,COMMAND -- Thanks, Warren Selby, dCAP-- _ -- 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] Does Cisco 79XX with SIP firmware support asterisk's BLF ?
On Apr 5, 2012, at 6:58 AM, Olivier wrote: > Hi, > > Does Cisco 79XX with SIP firmware support asterisk's BLF ? > Has someone been successful with this ? > > I've read that it can if you use tcp instead of udp, but I've never tested it myself. -- Thanks, Warren Selby, dCAP -- _ -- 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 pregi net account registration
On Apr 5, 2012, at 3:21 AM, Gopalakrishnan N wrote: > Hi guys, > > I am trying to configure sip.pregi.net account with my Asterisk 1.4.X, since > its a free account, its not getting registered, even my machine IP is allowed > in firewall. In the same machine if i register openser account which is in > public i am able to register. while checking the sip debug the register > request is keep on sending but there is no response. > > what i did is i registered the same account in my softphone installed in my > laptop, there it got registered. only with Asterisk its not registering, I > tried allowing externip as my routers IP, even then its not getting > registered. > What settings are you currently using, and what does your infrastructure look like? -- Thanks, Warren Selby, dCAP-- _ -- 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] extending fallback numbers
On Mon, Apr 2, 2012 at 7:05 PM, Paolo Supino wrote:
> Hi
>
> A couple of weeks ago I asekd how to setup a fallback numer and one of
> the reply I received was to se GotoIF and ${DIALSTATUS}.
> I succeeded in making it work for a single fallback number (i.e. the
> operator), but I want to extend it in the following manner:
>
> 2000-2099 -> fallback to 2000
> 2100-2199 -> fallback to 2100
> 2200-2299 -> fallback to 2200
> 2300-2399 -> fallback to 2300
>
> and so on...
>
>
> How do I implement such a configuration in a dialplan?
>
>
The simplest way is to just use pattern matching and multiple Dial
statements in consecutive order, like so:
exten => _20XX,1,Dial(SIP/${EXTEN},30)
exten => _20XX,n,Dial(SIP/2000,30)
exten => _21XX,1,Dial(SIP/${EXTEN},30)
exten => _21XX,n,Dial(SIP/2100,30)
exten => _22XX,1,Dial(SIP/${EXTEN},30)
exten => _22XX,n,Dial(SIP/2200,30)
exten => _23XX,1,Dial(SIP/${EXTEN},30)
exten => _23XX,n,Dial(SIP/2300,30)
This doesn't take things like DIALSTATUS into account, however it
accomplishes the same goal of having a fallback number, if that's what you
want. If you want to add a check for DIALSTATUS, just do it for each
pattern.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
--
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Re: [asterisk-users] Asterisk ACL
On Mon, Apr 2, 2012 at 7:44 AM, Mark Farmer wrote: > Hi > > ** ** > > We are trying to accept inbound calls from a SIP provider who sends us > calls from various IP’s within a given subnet but they are failing every > time with the following message on the console. > > ** ** > > chan_sip.c:20006 handle_request_invite: Call from '' to extension > '' rejected because extension not found > > > Does "destination-number" contain the context the call is failing in, or is that listed after the "extension not found" part? Can you provide a bit more of the CLI output before the failure? I've seen this type of error before and a lot of the time it has to do with the "insecure=" settings being used. Which version of asterisk are you using? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] keep dst cdr record if context change
On Fri, Mar 30, 2012 at 4:51 PM, Daniel Knoll wrote:
> Looks nice, was also my first idea, but field dst is read only. I can't
> overwrite this and get an error like this
>
> ERROR[2474]: cdr.c:345 ast_cdr_setvar: Attempt to set the 'dst' read-only
> variable!.
>
>
>
I was afraid of that. Does it absolutely have to be dst that you store
this information in? You can create custom cdr fields that are both
readable and writeable. Something like:
[incoming]
exten => _X.,1,Verbose(New call coming in)
exten => _X.,n,Set(CDR(original_dst)=${EXTEN})
exten => _X.,n,Goto(mainmenu,s,1)
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
--
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Re: [asterisk-users] keep dst cdr record if context change
On Fri, Mar 30, 2012 at 2:21 PM, Danny Nicholas wrote:
> So you have a situation like so:
> [default]
> Exten => _X.,1,Answer
> Exten => _X.,n,Goto(foo,s,1)
> [foo[
> Exten => s,1,playback(vm-goodbye)
> Exten => s,n,hangup()
>
> And you get two CDR records, 1 with default and 1 with foo?
>
No, he should be getting 1 record with "s" in the dst field.
To the OP: have you tried setting a channel variable to "${EXTEN} before
your Goto() command, and then in the "h" exten write it back into the cdr?
Something like:
[incoming]
exten => _X.,1,Verbose(New call coming in - verify routing)
exten => _X.,n,Set(finaldst=${EXTEN})
exten => _X.,n,Goto(mainmenu,s,1)
exten => h,1,Verbose(Hanging up)
exten => h,n,Set(CDR(dst)=${finaldst})
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
--
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Re: [asterisk-users] AGI variables being wrong
On Fri, Mar 30, 2012 at 8:43 AM, Mikhail Lischuk wrote: > ** > > Warren Selby wrote 29.03.2012 22:46: > > >> > To do this, you change your features.conf setting like so: > > parse => *9,peer/both,Macro,Parse > > > The same result when I changed to Macro. I believe that it's true that > callerid on outgoing call is "crap shoot". Here is output: > A couple things - what version of asterisk are you using? Are you actually using zaptel or do you have DAHDI as your interface to your TDM cards? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] AGI variables being wrong
On Thu, Mar 29, 2012 at 2:16 PM, Mikhail Lischuk wrote:
> **
>
> Warren Selby писал 29.03.2012 20:20:
>
> I'd be really curious to see the entire CLI log of the call, with
> verbose set to 6 and AGI debug enabled, from when the call first comes in
> to when it's hung up, including the execution of the *9 feature code.
> Also, knowing which version of Asterisk and DAHDI we're dealing with here
> couldn't hurt
>
> The output is pretty same. I can enable DTMF debugging, but can't
> imagine how could it help us:
>
> -- Launched AGI Script /etc/asterisk/agi/map.pl
>
>
What I meant was, let's see the CLI output of the entire call, from the
time it starts, to the time it stops. Something like the following:
-- Accepting call from 'XX' to 'YY' on channel 0/16,
span 1
-- Executing [YY@incoming-pri:1] Wait("DAHDI/16-1", "2") in new
stack
-- Executing [YY@incoming-pri:2] Verbose("DAHDI/16-1",
"Incoming call from XX to Main Line YY on 03/29/12 at
14:39:15.") in new stack
-- Executing [YY@incoming-pri:3] Goto("DAHDI/16-1",
"remote-phones,7999,1") in new stack
-- Goto (remote-phones,7999,1)
-- Executing [7999@remote-phones:1] Verbose("DAHDI/16-1", "Trying
extension 7999 on remote host remote.") in new stack
-- Executing [7999@remote-phones:2] Dial("DAHDI/16-1",
"SIP/7999@pbx-remote") in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Called 7999@pbx-remote
-- SIP/pbx-remote-0405 answered DAHDI/16-1
-- Channel 0/16, span 1 got hangup request, cause 16
== Spawn extension (remote-phones, 7999, 2) exited non-zero on
'DAHDI/16-1'
-- Hungup 'DAHDI/16-1'
But with the AGI debug thrown in the middle where appropriate.
>
> To the OP - just trying to think outside the box here, but what if instead
> of calling the AGI directly from the features.conf feature code, you wrote
> a Macro or GoSub that you could then use as your application, and within
> the Macro / GoSub you executed your AGI?
>
> I'd love to, but I need that script to run only when user hits some key
> combo during call. All I was able to find regarding that, was using
> features.conf and dynamic application. If you can advise me some workaround
> - I would appreciate.
>
>
To do this, you change your features.conf setting like so:
parse => *9,peer/both,Macro,Parse
And you add something like this to your extensions.conf:
[macro-Parse]
exten => s,1,Verbose(Parsing AGI variables)
exten => s,n,AGI(map.pl)
Assuming your map.pl is in the place where your asterisk looks for agi (by
default this is /var/lib/asterisk/agi-bin), otherwise include the entire
path to the file.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
--
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Re: [asterisk-users] AGI variables being wrong
On Thu, Mar 29, 2012 at 10:10 AM, Steve Edwards wrote: > On Thu, 29 Mar 2012, Mikhail Lischuk wrote: > > I have the following line in features.conf: >> parse => *9,peer/both,AGI,/etc/**asterisk/agi/map.pl >> > > I've never invoked an AGI from 'features,' but I'll assume it's 'the same' > as executing an AGI from the dialplan. Neither have I. I'd be really curious to see the entire CLI log of the call, with verbose set to 6 and AGI debug enabled, from when the call first comes in to when it's hung up, including the execution of the *9 feature code. Also, knowing which version of Asterisk and DAHDI we're dealing with here couldn't hurt > > During outgoing call, those variables get messed up. >> > > Is that some bug, or misconfiguration, or maybe wrong programming? >> > > Usual 'fails' in AGIs are: > > 1) Not using an established AGI library. While the AGI protocol is simple, > nobody gets it right the first time. > > 2) Forgetting that your AGI's STDIN and STDOUT 'belong' to Asterisk and > printing a debugging message or something similar. > > 3) Not reading the AGI environment from STDIN before requesting an AGI > command. > > While I would normally completely agree with you on this, he showed in his original example the AGI environment that is being sent to the script is what is "wrong", not the script's handling of said environment. To be more specific, the agi_callerid: and the agi_dnid: variables appear to have incorrect values for an outbound call. My guess would be that this has to do with calling the AGI from a features.conf feature code, and not from within dialplan itself. To the OP - just trying to think outside the box here, but what if instead of calling the AGI directly from the features.conf feature code, you wrote a Macro or GoSub that you could then use as your application, and within the Macro / GoSub you executed your AGI? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to show used "wrong password"
On Wed, Mar 14, 2012 at 1:36 PM, Randall wrote: > all works as expected only there is 1 extension that is trying to register > with a wrong password causing fail2ban to block the IP address, normally > that is ok behaviour but i have several extensions on that IP address. > > First of all, white list the IP in fail2ban and you won't accidentally ban the whole office. This can be done by following this guide: http://www.fail2ban.org/wiki/index.php/Whitelist Second, this is kind of outside the box thinking, so it may not work at all, but try setting the NAT on that peer to no, and then tcpdump the incoming registration attempts and see if you can see the internal private IP address of the packet. If there's a SIP helper on the far end, this may not help. Possibly, remove the secret= line from that peer in sip.conf and see if it successfully registers. Again, with the right nat= setting, you may be able to tcpdump the communication with that peer and get the private IP address so that you can then attempt narrow it down. This is not a long term solution, obviously, as it would create a gaping security hole, but it's worth a shot. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Transfer to fax
On Tuesday, March 13, 2012, Kevin P. Fleming wrote: > On 03/13/2012 05:45 PM, Eric Wieling wrote: >> >> The faxdetect option is documented in the 1.8 sip.conf.sample. > > Right, I forgot about that. Now I've really confused things. > > /me heads back to his hole > It was actually added to chan sip in 1.6.2, I remember that being a selling point on a 1.6.2 upgrade for a client of mine about a year and a half ago. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
On Wed, Feb 22, 2012 at 10:30 AM, [Digital^Dude] ® wrote: > So you mean I can't use dahdi_dummy with meetme? > That's not what he means at all. What it means is, you are required to install the software named DAHDI before you are able to compile and load the asterisk application MeetMe(). It also means you do not need to have a chan_dahdi.conf file in your /etc/asterisk directory. So, to recap, you must install and run DAHDI on the same server as your asterisk box if you want to use MeetMe, but you don't have to use DAHDI anywhere in asterisk itself (for instance, if you don't have any TDM interface cards). The "dahdi_dummy" virtual device was removed a few versions ago as it was redundant - just installing DAHDI provided the same timing source that dahdi_dummy did. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Streaming musiconhold via mpg123
On Tue, Feb 21, 2012 at 2:34 PM, Stephen Brown wrote: > At my wits end with this, and can't proceed any further so I'm hoping > someone has seen this and can assist. I can not get streaming musiconhold > to work with Asterisk. > > My Asterisk version is 1.8.8.0 and the mpg123 version is 1.9.1, OS is > CentOS 5.7. When I call the musiconhold class (default for example) I get > nothing but silence. I've exhausted my troubleshooting capabilities at this > point, I've tried everything I can think of to include: > > - a newer version of mpg123, I went with the latest version > - verified I could play an MP3 file by itself in Asterisk by using the > MP3Player application > > What does not work, is if I use the mpg123 application for musiconhold to > play a standalone file or a streaming source. I seem to be missing > something and I just can't quite put a finger on it. > Share with us your musiconhold.conf configuration please. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Park() ignores 'r' option which should disable music on hold in favour of ringing tone
On Thu, Feb 16, 2012 at 9:44 AM, James Stocks wrote: > When I receive a call, I want to automatically park it from the dialplan > so that I can retrieve it later. However, I don't want callers to be aware > that they are being parked, so I want to play a ringing tone to the caller. > Park() is supposed to be able to do this: > > > > Park([timeout][,return_context[,return_exten[,return_priority[,options[,parking_lot_name]) > options >r: Send ringing instead of MOH to the parked call. >s: Silence announcement of the parking space number. > > I've created an extension to test this with, here's what I have in > extensions.conf: > > exten => *10,1,Answer > exten => *10,n,Park(12,special,*59,1,rs) > exten => *10,n,Hangup() > > Here's the output on the Asterisk console: > I'm seeing the same behavior in asterisk 1.8.8.0. I suggest you open a ticket on https://issues.asterisk.org/jira and report the issue. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Calling a group of phones and force the speaker
> > On Wed, Feb 8, 2012 at 12:48 PM, Danny Dias wrote: > >> Hi, >> >> I wonder, if there is a way to call from A phone to a group of phones (B, >> C and D) and force these phones to activate automatically the speaker >> >> Is that possible? >> >> Many thanks in advance >> >> I've done this quite a few times with Polycom phones and the SipAddHeader() application in Asterisk. There's plenty of guides out there with details on how to do this. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Getting one way audio even NAT is configured
On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir wrote: > Hi all, > > I'm getting one way audio when calling over the SIP trunk i.e. end device > B (remote end of SIP trunk) can hear device A (softphone registered with > Asterisk) but device A can't hear device B. Even though I configured same > NAT configurations on other servers and they are working good. The NAT > configuration is listed below; > > localnet=130.0.0.0/130.0.0.0 > externhost=12.131.12.13 > externrefresh=10 > fromdomain=test.localhost.com > nat=yes > qualify=yes > canreinvite=no > > > NAT on device end i.e. my softphone (extension) has already set to yes > with canreinvite=no but still unable to resolve this issue. SIP traces are > listed below; > > > > The Asterisk version I'm using is 1.8.5. Please assist me at earliest. > Which device (A or B) is behind NAT with regards to your asterisk server? Is that the actual localnet= statement you're using, because to my understanding that is not the proper format to use (should be localnet=x.x.x.x/y.y.y.y where x.x.x.x is your actual local network, and y.y.y.y is your subnet for your local network). -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Which device auto-registered an extension?
Why not try set a variable under each device in sip.conf to the same as the
endpoint name then Dial(SIP/${CustomVar})?
Thanks,
--Warren Selby, dCAP
On Dec 15, 2011, at 7:03 PM, Barry Miller wrote:
> Hi all,
>
> In sip.conf:
> [general]
> regcontext = autoreg
>
> [devabc]
> regexten = 543
>
> creates "exten=> 543,1,Noop(devabc)" in context autoreg when devabc
> registers. But I can't use "exten=> _5XX,2,Dial(SIP/${EXTEN})" in the
> dialplan, because there's no device SIP/543. Now I know I can add a line
> like "exten=> 543,2,Dial(SIP/devabc)" for each and every device that uses
> regexten, but it would be a lot cleaner to be able to use something like
> Dial(SIP/${WHAT_GOES_HERE?}) instead.
>
> So is there a way for the dialplan to determine which device caused SIP to
> auto-register an extension?
>
> --
> Barry
>
> --
> _
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Re: [asterisk-users] ODBC problem - static realtime file not loading
On Fri, Dec 16, 2011 at 6:06 AM, Brynjolfur Thorvardsson wrote: > > > After connecting, the asterisk user never sends another SQL statement, at > least nothing that shows up in the General log. Asterisk is running as > root. I’ve deleted the musiconhold.conf file from /etc/asterisk > > > I had always thought you still needed the musiconhold.conf file with at least one MOH class defined so that asterisk will load the MOH module. Once it loads the module, then it should read from the database as well. I don't know why this works, but it's the way I've always done it. If this behavior resolves your issue, perhaps a bug ticket is in order on https://issues.asterisk.org/jira/ . -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] AST-2011-013: Possible remote enumeration of SIP endpoints with differing NAT settings
On Thu, Dec 8, 2011 at 4:47 PM, Asterisk Security Team < secur...@asterisk.org> wrote: > Asterisk Project Security Advisory - AST-2011-013 > > > >Description It is possible to enumerate SIP usernames when the general > and user/peer NAT settings differ in whether to respond to > the port a request is sent from or the port listed for > responses in the Via header. In 1.4 and 1.6.2, this would > mean if one setting was nat=yes or nat=route and the other > was either nat=no or nat=never. In 1.8 and 10, this would > mean when one was nat=force_rport or nat=yes and the other > was nat=no or nat=comedia. > >Resolution Handling NAT for SIP over UDP requires the differing >behavior introduced by these options. > >To lessen the frequency of unintended username disclosure, >the default NAT setting was changed to always respond to the >port from which we received the request-the most commonly >used option. > >Warnings were added on startup to inform administrators of >the risks of having a SIP peer configured with a different >setting than that of the general setting. The documentation >now strongly suggests that peers are no longer configured >for NAT individually, but through the global setting in the >"general" context. > > This seems very counter-intuitive for anyone that has their asterisk server on a public IP address and serves clients both behind and not behind NAT. I've always viewed it as the nat= setting inside the [general] context is for whether or not the asterisk server itself is behind a NAT, and then the nat= setting inside each [peer] definition is based on whether or not that particular peer / endpoint was behind nat or not. Have I viewed it incorrectly all this time? On that note, why have the nat= setting on peers in the first place if it's insecure / not recommended to have a setting that differs from the general nat= setting. I'm not trying to be smug, I'm generally curious about the reasoning behind taking this approach to deal with this security issue, instead of changing code somewhere (I'm not a programming, and thus have no idea how complicated such a change would be). -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] CDR mysql with asterisk 1.4
In order to install MySQL support for asterisk 1.4, you'll need to download the asterisk-addons-1.4 tarball, extract it to it's own folder. Go to that folder, run ./configure, make menuselect, and select the cdr_addon_mysql and the res_config_mysql options. Exit make menuselect, then run make, make install, and make samples. This should add the necessary modules to asterisk, as well as the sample config files. This of course assumes you've got mysql and it's development packages installed. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Is it possible call land into extensions.ael configuration file not in extensions.conf
Sorry for the top post, this is from my phone. Asterisk parses all of the config files (.conf, .ael and .lua, assuming you have the appropriate modules loaded) at the time you load asterisk or reload the dialplan (dialplan reload). It does not read the files each time a new call is started. Thanks, --Warren Selby, dCAP On Nov 23, 2011, at 6:11 AM, virendra bhati wrote: > Hi Gohar, > > As per you suggestion I make context into AEL file and working file. > > But I do little bit R&D on that case I make same context into both > files(.conf and .ael) and asterisk read 1st .conf files extension. It means > if we make anythings into AEL files then asterisk 1st check into .conf file > then another one. It might be time consuming if we have Lot's off context. > > But any way thanks for you reply. > > On Wed, Nov 23, 2011 at 5:16 PM, Gohar Ahmed wrote: > Hi, > > Create a context in AEL, or LUA and change the context=ael-context or > context=lua-context in sip.conf [default] section or for each sip user > decalred who needs to start call in context defined in AEL/LUA? > > > > Regards, > > Gohar > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati > Sent: Wednesday, November 23, 2011 4:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; > Sam Govind > Subject: [asterisk-users] Is it possible call land into extensions.ael > configuration file not in extensions.conf > > > > Hi List, > > I want to change the asterisk flow. right now call startd from > extensions.conf. Is there any way by which we can changed it to > extensions.ael or extensions.lua ? > > > > - > Thanks and regards > > Virendra Bhati > +91-9172341457 > Software 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 > Software Engineer > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file with AuthenticateApplication
On Tue, Nov 22, 2011 at 11:02 PM, virendra bhati wrote: > Hi Warren, > > As per your suggestion I revert back the things. In such case nothing is > working. So it's completely wrong case. > > Can someone tell me how Authenticate check password from plan text file ? > If we know who it's work then we can implements the logic on it. > I'm not sure, one thought would be to try without the "a" option in the Read()? Other than that, I'd suggest maybe opening a ticket on the issue tracker. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file with AuthenticateApplication
On Mon, Nov 21, 2011 at 6:15 AM, virendra bhati wrote: > Hi, > > After deleting all space no improvements. > Try reversing the account code and password hash, like this: 81dc9bdb52d04dc20036dbd8313ed055:Virendra 9996535e07258a7bbfd8b132435c5962:Vijay 7bccfde7714a1ebadf06c5f4cea752c1:VirendraBhati -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Goto Queue, does not work, it should play message or any thing
On Tue, Nov 15, 2011 at 4:56 AM, bilal ghayyad wrote: > Hi All; > > When the call coming via the E1 dahdi and I handle the call (as first > step) by exten => 5631040,1,Goto(OrangeCMG,s,1) it is not working and the > call will be disconnected instead of queued. > > But, when I handle the call (as first step) by playing any sound file and > then send for the queue, then it is working fine, WHY? > > exten => 5631040,1,Playback(WelcomeMessage) > exten => 5631040,2,Goto(OrangeCMG,s,1) > > > So how I can overcome this? > Show us the CLI output of a call that's not doing what you want and a call that is, and we can compare the differences. My guess is it has something to do with Playback having an automatic Answer(), and whatever you're Goto'ing doesn't... -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Logging Specific Verbose Level To Seperate File
If you call DumpChan from an AGI you should be able to read the response programmatically and then dump the data into a database. Cleans up your dialplan but requires some scripting or programming knowledge (php, perl, bash or even C) in order to write the AGI. Thanks, --Warren Selby, dCAP On Nov 13, 2011, at 11:14 PM, Tristram Cheer wrote: > Hi Sammy, > > It's a good start, Atleast being split it is handy, Ideally I'd to be able to > spit DumpChan output direct to JabberSend or func_ODBC but I fear this will > require someone who know's C to alter the module. I think i'm going to have > to just use JabberSend for each variable I use and the channel details which > is going to blow out the size of the dialplan a bit but I cant see another > way around it > > > Cheers > > On 14 November 2011 18:07, Sammy Govind wrote: > Hello, > Reading about the application DumpChan() shows this: > > [Synopsis] > Dump Info About The Calling Channel. > > [Description] > Displays information on channel and listing of all channel variables. If > is specified, output is only displayed when the verbose level is > currently set to that number or greater. > > [Syntax] > DumpChan([level]) > > So in theory its just another Verbose output on CLI, you can separate Verbose > logging to another file in logger.conf. Your verbose level is 1001 so > whenever you set "core set verbose 1001" this DumpChan() application will > start dumping output in CLI and then fro there be logged in the Verbose > logging file. > > I don't think this is exactly what you require. > > -- > Regards, > Sammy > > On Mon, Nov 14, 2011 at 3:19 AM, Tristram Cheer > wrote: > Hi All, > > Hopefully this is considered on-topic for this list. > > I'm using DumpChan(1001) in a Macro called debug in order to debug issues > within the dialplan, I would like to dump this output to a file specifically > for DumpChan output but I'm having issues with figuring out how to do this > under logger.conf. Ideally I would like to put DumpChan into SQL using > func_ODBC but it seems that you can't do this so runner up is a file. > > Anyone have any pointers on how to do this? I would like to log DumpChan > output and only DumpChan output to a separate file. > > > Cheers! > > -- > _ > -- 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] FFA - Asterisk 1.6.2.6
Sorry for the top post. A valid fax extension is an extension named 'fax' in the incoming context for that peer. I.e: exten => fax,1,Verbose(1,Incoming fax call detected) exten => fax,n,ReceiveFax() You would obviously have a much longer definition, this was just a quick example from my phone. Thanks, --Warren Selby, dCAP On Nov 2, 2011, at 9:09 AM, Christian Tardif wrote: > On 02/11/2011 05:04, Anton Kvashenkin wrote: >> >> Turn off faxdetect on this peer. >> >> 2011/11/2 Christian Tardif >> Hi, >> >> I have a 1.6.2.6 fax installation with a FFA license which seems to be >> installed correctly (in fax show stats, I see that I have 1 Digium G.711 >> licensed channel, and 1 Digium T.38 licensed channel). >> >> When trying to call my business line with a fax machine, it looks like it's >> ringing to my asterisk box, then transfer the call to my extension. In the >> logs, I see (after the line where it says that my extension is ringing): >> chan_sip.c: Fax detected but no fax extension. >> >> How come does Asterisk even try to ring my phone? It seems that the >> detection (which should append BEFORE any phone ring) does not work, and I >> have no clue where to look at. >> >> In case this helps, I'm configuring the installation with FreePBX 2.8.1.4 > > I just checked, and faxdetect is not enabled on any peer. I was wondering if > any time condition could disturb my tests yesterday night (I was testing way > after office close hours) but no, there's no impact. Asterisk still does not > detect incoming fax. > > I just found, while reading the doc. I have to detect fax in legacy mode in > order for Fax For Asterisk to do the detection correctly. Well, for now, as > this is deprecated. I now know that the detection works and that I'll have, > in a near future, remove the legacy detection and route incoming fax to a > valid fax destination I'll have to understand what a valid fax extension > is... :-) > -- > Christian Tardif > -- > _ > -- 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] State of Asterisk+Virtualization+Timing
Look at upgrading to at least 1.6.2 or 1.8, these both have newer timing sources that don't rely on dahdi. Also, if conferencing is a big deal, look at 10, this contains a complete rewrite of ConfBridge which doesn't require dahdi for mixing at all. Thanks, --Warren Selby, dCAP On Nov 1, 2011, at 12:08 PM, Tim Nelson wrote: > Greetings- > > I'm about to dive into the process of virtualizing some of my Asterisk > (primarily 1.4.x) infrastructure. In the past, when looking at virt > solutions, the primary issue preventing me from moving was the lack of proper > timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like > to use either OpenVZ or KVM, but each seem to have independent "issues" that > need to be addressed: > > OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant > access to host node timing source (physical device, or dahdi_dummy in > /dev/dahdi/) to the containerized Asterisk process. > > KVM - Higher overhead, easier installation, 'true virtualization'. Primary > issue is not timing per se, but KVM scheduling. Timing source, while present > from dahdi_dummy natively may still not get proper scheduling by KVM process. > This could also affect general call quality (even non IAX2 trunked voice), > DTMF, etc. > > I have to believe there are others running virtualized Asterisk installations > with some degree of success on OpenVZ or KVM. Care to share your thoughts? > > --Tim > > -- > _ > -- 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] Temporarily disabling voicemail recordings (but not greetings)
On Mon, Oct 31, 2011 at 8:22 AM, Danny Nicholas wrote: > If you are using the “silent” option of voicemail (b – busy, u – > unavailable, s – silent) you could set up a context to play the “normal > silent” message, then goodbye. > > Completely irrelevant, but I always thought of 's' in the Voicemail() application as "skip intstructions", not "silent". Sorry, just one of those things that made me go hmmm. :) -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] Reporting for Asterisk Call Center
On Tue, Oct 25, 2011 at 7:30 AM, bilal ghayyad wrote: > Dear Tark; > > The asterisk version I am running is 1.8 and I can select mysql from > menuselect when I am compiling. > > But when I googled for cdr-mysql, I discovered that I have to login for > mysql and create the database and run a script to create this and give the > grants. All what I found in google is related to other asterisk versions, > while mine is 1.8, so the problem is how to know the required script to > create the database and give the right grants to be used for CDR that suite > the version I am running? From where I can get this? > > The following script will generate an "asterisk" database with a table named "CDR" that will work with asterisk 1.8. Be sure to change 'PASSWORD' with whatever password you want to use. SET SQL_MODE="NO_AUTO_VALUE_ON_ZERO"; CREATE DATABASE `asterisk` DEFAULT CHARACTER SET latin1 COLLATE latin1_swedish_ci; USE `asterisk`; CREATE TABLE IF NOT EXISTS `cdr` ( `recid` mediumint(8) unsigned NOT NULL auto_increment COMMENT 'Record ID', `calldate` datetime NOT NULL default '-00-00 00:00:00', `clid` varchar(80) NOT NULL default '', `src` varchar(80) NOT NULL default '', `dst` varchar(80) NOT NULL default '', `dcontext` varchar(80) NOT NULL default '', `channel` varchar(80) NOT NULL default '', `dstchannel` varchar(80) NOT NULL default '', `lastapp` varchar(80) NOT NULL default '', `lastdata` varchar(80) NOT NULL default '', `duration` int(11) NOT NULL default '0', `billsec` int(11) NOT NULL default '0', `disposition` varchar(45) NOT NULL default '', `amaflags` int(11) NOT NULL default '0', `accountcode` varchar(20) NOT NULL default '', `uniqueid` varchar(32) NOT NULL default '', `userfield` varchar(255) NOT NULL default '', PRIMARY KEY (`recid`), KEY `calldate` (`calldate`), KEY `dst` (`dst`), KEY `accountcode` (`accountcode`), KEY `src` (`src`), KEY `disposition` (`disposition`), KEY `uniqueid` (`uniqueid`) ) ENGINE=InnoDB DEFAULT CHARSET=latin1 AUTO_INCREMENT=1 ; CREATE USER 'asterisk'@'localhost' IDENTIFIED BY 'PASSWORD'; GRANT FILE ON * . * TO 'asterisk'@'localhost' IDENTIFIED BY 'PASSWORD' WITH MAX_QUERIES_PER_HOUR 0 MAX_CONNECTIONS_PER_HOUR 0 MAX_UPDATES_PER_HOUR 0 MAX_USER_CONNECTIONS 0 ; GRANT INSERT ON `asterisk`.`cdr` TO 'asterisk'@'localhost'; If you're going to be running the mysql database on the same server as the asterisk box, the following cdr_mysql.conf should also work for 1.8: [global] hostname=localhost dbname=asterisk table=cdr password=PASSWORD user=asterisk port=3306 sock=/var/lib/mysql/mysql.sock userfield=1 loguniqueid=yes -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] bug in queuemanager?
On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger wrote:
> Customer 200 calls to queue 900, Agent 300 answers but tells Customer 200
> that he should be at Queue 901 and transfers Customer 200 (using *2) to
> Queue 901. Agent 301 now gets the call from Queue 901 with Customer 200,
> answers the calls etc. After disconnect a new call arrivers immediately from
> Queue 901, without any wrap-up time. This should be considered as a bug IMO.
>
> Any ideas on how to fix, workaround this problem?
>
>
>
>
>
Please share the CLI output of such a situation, with the verbosity and
debugging both set to 10 ('core set verbose 10' and 'core set debug 10' from
the asterisk CLI), it may shed some light on whether this is a bug or a
"feature".
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
--
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Re: [asterisk-users] question about queues.conf
On Fri, Oct 21, 2011 at 1:15 PM, salaheddine elharit < salah.elharit...@gmail.com> wrote: > hi > here is my extensions.conf and aheeva_diaplan.conf > > if you can see theses files and tell my if there is any wrong > > regards > > > These configuration files you sent me don't seem to match up with the dailplan CLI you showed earlier. Please, do the other things I asked about in my last email, and let's move forward from there. Also, let's keep the emails on the list. For reference, I've included my requests below: 2011/10/21 Warren Selby > Please do the call again, this time please show us the output also with a > sip debug and a zap debug. > > These are both very old versions. The current release of asterisk is > currently five generations newer than what you're using, and Zaptel isn't > even used anymore, the tool was renamed to DAHDI. It may make more sense to > update to the latest version of at least the 1.4 branch of asterisk > (currently 1.4.42 I think?) and make the switch to DAHDI. This will require > some effort on your part, so don't do this without planning on a production > box. > > I don't know why you only need 3 numbers for your second provider, perhaps > that's all that they are sending you? You will probably need to ask the > provider why they are not sending you the full number like you're expecting. > -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk call transfers not working
On Mon, Oct 24, 2011 at 3:13 PM, Ramiro Paz wrote: So, are the separate FXS extensions able to call each other when NOT transferring calls? What are the actual "Extensions" numbers you've assigned these phones in the FreePBX GUI? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.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] question about queues.conf
On Fri, Oct 21, 2011 at 12:24 PM, salaheddine elharit < salah.elharit...@gmail.com> wrote: > yes if i chang it from queues or meetme to dial there is no issue it'works > withou issue > > Please do the call again, this time please show us the output also with a sip debug and a zap debug. > the asterisk version is > Asterisk 1.4-r110474M > zaptel-1.4.12.1 > These are both very old versions. The current release of asterisk is currently five generations newer than what you're using, and Zaptel isn't even used anymore, the tool was renamed to DAHDI. It may make more sense to update to the latest version of at least the 1.4 branch of asterisk (currently 1.4.42 I think?) and make the switch to DAHDI. This will require some effort on your part, so don't do this without planning on a production box. > > i want to know also why for the first provider we put all the number in > extensions .conf but for the second provider we put just the last 3 numbers > > > I don't know why you only need 3 numbers for your second provider, perhaps that's all that they are sending you? You will probably need to ask the provider why they are not sending you the full number like you're expecting. There may be more reasons hidden in your extensions.conf, if you want to share it maybe someone here can go over it and spot anything that sticks out? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to know RTP por of a SIP client in the dialplan
On Fri, Oct 21, 2011 at 2:46 AM, ISABEL ORDAS ARNAL wrote: > Hi all, > > How can I get the RTP port one SIP client is using for sending/receiving > RTP flow? Can I obtain it in from SIP_HEADER of something like that in the > dialplan? > > Thank you! > > ** ** > > I don't think you can pull this information from a dialplan native application, but you could probably write an AGI that pulls this information for you. The AGI Environment data includes things like the current channel in use, which should be able to start you off in the right direction. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dialplan macro output
On Fri, Oct 21, 2011 at 6:54 AM, ISABEL ORDAS ARNAL wrote:
> Hi all,
>
> ** **
>
> Is there a way to read in the dialplan a macro output parameter?
>
> For instance, in the following macro I would like to know the pid of the
> Linux process for killing it when hanging up.
>
>
I think what you're looking for is a GoSub that ends with a Return(value).
You then can pull up the value in ${GOSUB_RETVAL}. But I may be
misunderstanding what you're wanting to do.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
--
_
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