similar to: CLI - displaying all channel variables

Hi John This works fine providing extensions 1001,1002 and 1003 are "Incall" or "Paused" - the problem appears to be that is a handset say 1002 is "ringing" then the 2xxx then the penalty is not honoured. This is well described in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

Hi Dovid Yes was one of the options but as the required list is dynamic becomes very messy - and all combinations problem - where as "call all workers job xxx" is what is needed so the ability to call 20+ numbers is what is needed - agi does a database search for all jobx workers and constructs a dialstring with SIP, DAHDI and Local devices. Can someone tell me where to find maximum

Hi All I have been looking at this problem for a few days/weeks now and after some advice please. I currently have a customer on 11.25.3 and I am in the process of upgrading versions and OS (Debian) and all things that involves mysql -> PDO etc The problem I have is the customer want a simple call distribution like this Extn 1001, 1002, 1003 to be called on an incoming call - if they

Hi list I have a requirement that I just don't know how to address - I don't think its strange but can't find any pointers anywhere. I have a user that wishes to have a "multi phone" divert. By that I mean "calls made to his extension say Ext200 can be redirected to a different extension say Ext400 and also to his home landline. Doing the dial is fine using

Hi List I am having trouble running the command siptest:~# asterisk -rx 'dialplan reload' most times it does what I expect and I get a response as below siptest:~# asterisk -rx 'dialplan reload' Dialplan reloaded. every now and then I get no response i.e. siptest:~# asterisk -rx 'dialplan reload' siptest:~# and a "verbose 10" setting shows [Mar

Hi All I am trying to get voicemail switched over to ARA on version 13 and notice that the password is not stored in the db when it is changed. A little hair pulling and playing around and I think the problem is in the function ast_update2_realtime in main/config.c. Issued source is ==> int ast_update2_realtime(const char *family, ...) { RAII_VAR(struct ast_variable *,

Hello all, Just wondering on a behavior I noticed while testing with realtime sip peers with names like 111.222 at mydomain.com. Using Kamailio as outbound proxy, it sends Asterisk a sip message where To header value is < sip:111.222 at mydomain.com> and From header has value "username" < sip:111.333 at mydomain.com;transport=UDP>;tag=fc609171. When Asterisk sends out the

Paddy, It's pretty easy to spot from the CLI. A voicemail gets called. And the screen basically stops scrolling from there. Eventually you'll get the "Task processors exceeded 500 queued tasks" or something like that. And maybe channels attempting to hangup due to lack of RTP (If you have no-rtp timers configured). Once you find the problem mailbox, You can call it via any

Hi List I have had a look at the various posts on this and seem to be more confused than ever - but then again that's not hard ;-) I am using Version 1.4.33.1 build from the Debian "lenny" distros I am trying to implement a simple screening [macro-screen] exten => s,1,Background(press1) exten => s,n,WaitExten(5) exten => 1,1,NoOp(accepted) ; Dont set a reply so dial

We ultimately found this to be a voicemail issue. The voicemail is held in MYSQL as well (via ODBC). And we found when attempting to playback a customers voicemail unavail greeting is when the deadlock would occur (Immediately, every time. Throwing the same "task processors" errors, And making pjsip completely unresponsive). We had imported a number of greetings from a legacy asterisk

Hi, I'm setting up Asterisk as a voicemail with SER. My problem is, when a caller that is not registered with asterisk (no username and password in sip.conf) it prompts "403, Forbidden" . I need all calls from outside of my network to reach asterisk for my users' voicemails, because anonymous users will surely reach voicemail of my users to leave messages. What do I

Hello, It looks I encoutered a bug or misconfiguration. fts_squat search for subject and body works excellent for English mails. For non-English (in particular, Russian) it works only when query consists of 1 word. Phrases - 2 and more words - always returns nothing. Example: search for "planet" ("???????") returns results, search for "Earth" ("?????") also

Hi all A quick question about busy lamps I have lamps working 'sort-of' on my GXP2000 lamps flash with ringing and go solid red when call gets answered but stay green when a call is made from the extension. Setup is Ext 200, 201, 202, each monitor the other two when 200 calls 202 - the BLF on 200 and 201 flash red - when 202 answers both 200 and 201 show BLF for 200 as red but

Hi, Not sure if this has been asked before. When we create new users in our AD box, they get the UIDs in high 30000 range. Is this normal? If yes, can we change the starting range? We migrated using ClassicUpgrade, now running Ubuntu 18.04 VM The smb.conf looks like workgroup = ADDOM realm = ADDOM.COM netbios name = WNAD01 server role = active directory domain

These small fixes are required to fix the build and tests on Debian (using the debootstrap/debirf appliance). There is currently another bug in the Debian tests which I'm looking into. hexdump always fails as follows: libguestfs: error: hexdump: hexdump: /test123: hexdump: /sysroot/test123: Bad file descriptor Rich. -- Richard Jones, Virtualization Group, Red Hat

Hi all, I want to let linux server join ad by using a trust ad's child domain user, but failed with error. below is my env and what I have try I have 3 domain controller: test.com,demo.com and chn.demo.com test.com with demo.com is two way trust. and chn.demo.com is the child domain of demo.com demo at demo.com chn at chn.demo.com can join ad member to test.com I have tested demo at

Hi All, Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and asterisk 1.2.14 ? i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123

Hello, I cannot run FTS SOLR engine. OS is debian lenny. I re-compiled debian package (from backports) with --with-solr option. Dovecot loads fts_solr plugin. I know that it loads plugin because if I remove lib21_fts_solr_plugin.so, dovecot does not start. But it does not attempt to communicate with SOLR. Simple check: stop solr tomcat, and open 8080 port in netcat: $nc -l -p 8080 dovecot

Hi all, I''m running into this frustrating issue with the acts_as_authenticated plugin and one of the additional fields I added to the users table. I have added a column ''account_type'' to the users table (among several others) which denotes the type of account it is. This stores an integer value that is set by the system during account creation. The value is getting set