Re: [asterisk-users] Overriding global voicemail options on a per-mailbox basis
Hi John, Em 17/08/2014 12:04, Tech Support aster...@voipbusiness.us escreveu: So my question is this: Does anyone know which options can and can't be overridden on a per-mailbox basis? Are there 9 options, are there 39, or are there 26? Where can I find a definitive answer? It would be extremely time consuming to test each option individually. This is not an answer, just an observation ... The fourth edition of The Asterisk book is based on Asterisk 11. Best. -- _ -- 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] 'restart when convenient'
Hi Matthew, Em 28/05/2014 15:09, Matthew Jordan mjor...@digium.com escreveu: * now - tell all CDRs to go submit themselves. Tell all channels to Where CDR is quoted is it also valid for CEL? Tks. -- _ -- 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] CEL Park APP_START and APP_END events
Dear friends, In Asterisk 11.7.0, is it possible to receive CEL APP_START and APP_END events from Park application? Queue and Dial apps are generation these events, but Park no. It doesn't seem to make difference when configured. cel.conf ... apps = dial,queue,confbridge,park events = ALL ... localhost*CLI cel show status ... CEL Tracking Application: queue CEL Tracking Application: confbridge CEL Tracking Application: dial CEL Tracking Application: park ... Tks. -- _ -- 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] No voice when the calls come from Internet
Do you have any idea when the voice is heard only when the call is from my local network to the Internet and not in the other direction ? Hi Neo, In the documentation look for this options: externip localnet nat It helps to understand them because this kind of situation is common with VOIP. Best. -- _ -- 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, CEL and ConfBridge recordings
Thank you very much Rusty.
It really works. Even if ${MyCustomFileName} gets a different value when
the second participant enters the conference, the filename remains the name
defined when the first participant enters (because he started the
conference).
Another thing, if I need to know the time a conference ended should I use
CEL or is there another better approach?
Best.
2014-03-06 21:28 GMT-03:00 Rusty Newton rnew...@digium.com:
On Wed, Mar 5, 2014 at 1:30 PM, Jairo jairomolin...@gmail.com wrote:
Dear friends,
Need to know filenames of conference recordings in Asterisk 11.
Besides directory scanning the recordings could use CEL:
Filter MySQL rows with eventtype equal CHAN_START and channame like
ConfBridgeRecorder and then get the eventtime field and convert to
timestamp
to complete filename(s).
Would you suggest any other approaches?
You might set the record file path yourself through the CONFBRIDGE
function, for example, in dialplan:
...stuff up here to build a unique file name into MyCustomFileName...
exten = 1,n,Set(CONFBRIDGE(user,record_file)=${MyCustomFileName}.wav)
Then of course you now know the file name so you could do whatever you
wanted with it afterwards.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CONFBRIDGE
--
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direct: +1 256 428 6200
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Re: [asterisk-users] Asterisk 11, CEL and ConfBridge recordings
Great :)
Thank you very much.
Best.
2014-03-07 18:07 GMT-03:00 Rusty Newton rnew...@digium.com:
On Fri, Mar 7, 2014 at 7:21 AM, Jairo jairomolin...@gmail.com wrote:
Thank you very much Rusty.
It really works. Even if ${MyCustomFileName} gets a different value when
the
second participant enters the conference, the filename remains the name
defined when the first participant enters (because he started the
conference).
Another thing, if I need to know the time a conference ended should I use
CEL or is there another better approach?
If you can get it from CEL, there is that, otherwise you can track
when you receive the AMI event ConfbridgeEnd
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_ConfbridgeEnd
That is all I got from poking around the docs. :)
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direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
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[asterisk-users] Asterisk 11, CEL and ConfBridge recordings
Dear friends, Need to know filenames of conference recordings in Asterisk 11. Besides directory scanning the recordings could use CEL: Filter MySQL rows with eventtype equal CHAN_START and channame like ConfBridgeRecorder and then get the eventtime field and convert to timestamp to complete filename(s). Would you suggest any other approaches? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to selectively disable callerid block?
Hi Eric,
Take a look at the call pickup code, maybe you need to change it to not
conflict with your dialplan:
localhost*CLI features show
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #1
Attended Transfer #2
One Touch Monitor
Disconnect Call * *
Park Call #3
One Touch MixMonitor
2014-02-12 21:23 GMT-02:00 Eric Cooper e...@cmu.edu:
In Asterisk 1.8, I used the following line in extensions.conf to allow
me to pass *82 in front of a dialed number, to disable the callerid
block that's normally on that POTS line:
; disable callerid block
exten = _*82.,1,Dial(${POTS}/${EXTEN})
But this seems to have stopped working when I upgraded to Asterisk
11.7. I get the following debug output, with a no call pickup
possible message as soon as I press the '8':
[Feb 12 18:17:39] -- Starting simple switch on 'DAHDI/2-1'
[Feb 12 18:17:39] DEBUG[2339]: devicestate.c:442 devstate_event: device
'DAHDI/2' state '2'
[Feb 12 18:17:42] DEBUG[5898][C-000e]: sig_analog.c:1600
analog_handle_dtmf: Begin DTMF digit: 0x2A '*' on DAHDI/2-1
[Feb 12 18:17:42] DEBUG[5898][C-000e]: chan_dahdi.c:2145
my_handle_dtmf: Begin DTMF digit: 0x2A '*' on DAHDI/2-1
[Feb 12 18:17:42] DEBUG[5898][C-000e]: sig_analog.c:1600
analog_handle_dtmf: End DTMF digit: 0x2A '*' on DAHDI/2-1
[Feb 12 18:17:42] DEBUG[5898][C-000e]: chan_dahdi.c:2145
my_handle_dtmf: End DTMF digit: 0x2A '*' on DAHDI/2-1
[Feb 12 18:17:42] DEBUG[5898][C-000e]: sig_analog.c:2121
__analog_ss_thread: waitfordigit returned '*' (42), timeout = 0
[Feb 12 18:17:44] DEBUG[5898][C-000e]: sig_analog.c:1600
analog_handle_dtmf: Begin DTMF digit: 0x38 '8' on DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: chan_dahdi.c:2145
my_handle_dtmf: Begin DTMF digit: 0x38 '8' on DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: sig_analog.c:1600
analog_handle_dtmf: End DTMF digit: 0x38 '8' on DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: chan_dahdi.c:2145
my_handle_dtmf: End DTMF digit: 0x38 '8' on DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: sig_analog.c:2121
__analog_ss_thread: waitfordigit returned '8' (56), timeout = 0
[Feb 12 18:17:44] DEBUG[5898][C-000e]: chan_dahdi.c:5075
dahdi_enable_ec: Enabled echo cancellation on channel 2
[Feb 12 18:17:44] DEBUG[5898][C-000e]: features.c:7880
ast_pickup_call: pickup attempt by DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: features.c:7900
ast_pickup_call: No call pickup possible... for DAHDI/2-1
[Feb 12 18:17:44] DEBUG[5898][C-000e]: sig_analog.c:2211
__analog_ss_thread: No call pickup possible...
and then a busy signal.
I didn't pay much attention to the differences between Asterisk 1.8
and 11.7 since everything seemed to still work ... Can someone point
me in the right direction?
--
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Re: [asterisk-users] how to selectively disable callerid block?
Thanks for the feedback. Best. 2014-02-13 13:16 GMT-02:00 Eric Cooper e...@cmu.edu: On Thu, Feb 13, 2014 at 10:09:04AM -0200, Jairo wrote: Take a look at the call pickup code, maybe you need to change it to not conflict with your dialplan: localhost*CLI features show Builtin Feature Default Current --- --- --- Pickup *8 *8 Blind Transfer # #1 Attended Transfer #2 One Touch Monitor Disconnect Call * * Park Call #3 One Touch MixMonitor Thank you, that was indeed the issue. -- Eric Cooper e c c @ c m u . e d u -- _ -- 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] CEL for attented transfer
Hi Jean. I am not sure but believe the sequence of CEL events should be considered for a more detailed understanding of a call. Replicating the case you described I got this CEL event and fields: *** 17. row *** id: 27984 eventtype: ATTENDEDTRANSFER eventtime: 2013-11-20 15:05:33 userdeftype: cid_name: Jairo desktop cid_num: 311 cid_ani: 311 cid_rdnis: cid_dnid: 310 exten: 310 context: entrada-canal channame: SIP/311-351096-048d appname: Dial appdata: SIP/310-777940,40,kKtT amaflags: 3 accountcode: peeraccount: uniqueid: 1384967109.1184 linkedid: 1384967109.1184 userfield: peer: Local/321@entrada-canal-0001;1 The 3 extensions can be found in the event. Does it help? 2013/11/19 Jean-Denis Girard jd.gir...@sysnux.pf -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jairo, Le 19/11/2013 01:36, Jairo a écrit : https://wiki.asterisk.org/wiki/display/AST/CEL+Events+and+Fields Thanks for your reply, but I have read this page of the wiki, I know what the fields mean. What I don't understand is how the events in my example can be used to determine 107 was attended transferred to 103 by 100. Or I do know that Local/103@100-0042;1 and Local/103@100-0042;2 were created by asterisk when SIP/100-0275 asked for atxfer? How does the event ATTENDEDTRANSFER/ SIP/107-0274/ Local/103@100-0042;1 show that 107 is transferred to 103? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlKLi74ACgkQuu7Rv+oOo/hbXQCfbznLU4gcYBlz2OTATRQxrZlv 0DoAoLLYF+ykKuQtRPusTsVOrk/BcQQI =eO3S -END PGP SIGNATURE- -- _ -- 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] CEL for attented transfer
Hi Jean, you mean what each event indicates? As this link explain? https://wiki.asterisk.org/wiki/display/AST/CEL+Events+and+Fields 2013/11/19 Jean-Denis Girard jd.gir...@sysnux.pf -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nobody, really? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 Le 17/11/2013 19:03, Jean-Denis Girard a écrit : Hi list, I'm trying to use CEL to display channel information in real time. It works fine for simple calls, blind transfers, or SIP attended transfers (initiated from the SIP phone). My problem is for Asterisk attended transfers (atxfer as configured in features.conf). The scenario is: . phone 107 calls phone 100, . 100 dials the atxfer code, . 107 is on hold, and 100 hears the transfer message, . 100 dials phone 103, . 103 answers, . 100 hangups, . 107 and 103 are connected, . 107 hangups. CEL is configured with apps=all and events=ALL, and events are stored in a database via cel_pgsql. This is the list of events in the database for this call: eventtype |channame| peer - -++--- CHAN_START | SIP/107-0274 | CHAN_START | SIP/100-0275 | ANSWER | SIP/100-0275 | ANSWER | SIP/107-0274 | BRIDGE_START | SIP/107-0274 | SIP/100-0275 CHAN_START | Local/103@100-0042;1 | CHAN_START | Local/103@100-0042;2 | CHAN_START | SIP/103-0276 | ANSWER | SIP/103-0276 | ANSWER | Local/103@100-0042;2 | BRIDGE_START | Local/103@100-0042;2 | SIP/103-0276 ANSWER | Local/103@100-0042;1 | BRIDGE_START | SIP/100-0275 | Local/103@100-0042;1 BRIDGE_END | SIP/100-0275 | Local/103@100-0042;1 ATTENDEDTRANSFER | SIP/107-0274 | Local/103@100-0042;1 CHAN_START | Transfered/SIP/107-0274| BRIDGE_END | Transfered/SIP/107-0274ZOMBIE| SIP/100-0275 BRIDGE_START | SIP/107-0274 | Local/103@100-0042;1 HANGUP | SIP/100-0275 | CHAN_END | SIP/100-0275 | HANGUP | Transfered/SIP/107-0274ZOMBIE| CHAN_END | Transfered/SIP/107-0274ZOMBIE| BRIDGE_END | SIP/107-0274 | Local/103@100-0042;1 HANGUP | Local/103@100-0042;1 | CHAN_END | Local/103@100-0042;1 | HANGUP | SIP/107-0274 | CHAN_END | SIP/107-0274 | BRIDGE_END | Local/103@100-0042;2 | SIP/103-0276 HANGUP | SIP/103-0276 | CHAN_END | SIP/103-0276 | HANGUP | Local/103@100-0042;2 | CHAN_END | Local/103@100-0042;2 | LINKEDID_END | Local/103@100-0042;2 | (33 lignes) How should these events be interpreted? Asterisk version is 11.6.0. Thanks, -BEGIN PGP SIGNATURE- iEYEARECAAYFAlKK6p0ACgkQuu7Rv+oOo/heAACeN0eMR1qwRLcdV+Tsgn9fA+6c RKcAn246hmNUU2dxivPFEziueHYRTWcS =q196 -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CEL events
Dear list. This is probably a complex subject but is that right to consider: a) each distinct linkedid field value in a mysql CEL table as a unique call? b) the duration of a call as the period (eventtime fields) between BRIDGE_END and BRIDGE_START events of the same linkedid sequence? (not considering transfers) Just a start ... Tks. -- _ -- 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 fax in debian
Maybe this can help: http://ofps.oreilly.com/titles/9781449332426/asterisk-Fax.html Best. 2013/6/13 vortex binary.vor...@gmail.com Hello. i am running debian 6 with asterisk 11.4. The system has exim4 to send to email the voicemails. i would like to get rid of the analog fax machine and use asterisk to send/receive faxes. I do have a PSTN line with a SPA3102 adapter to interface it to asterisk. The number of the PSTN line is dedicated to faxing only. So i would like to: -receive faxes to asterisk and then send it as PDFs to an email address -Send from my PC a fax directly. is there any guide on how to do that since i got lost with all of it? -- __**__**_ -- 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-**usershttp://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] Is uniqueid/sequence a safe CDR table primary key ?
Hello, Still about CDR and MySQL table, should the calldate field be inserted by Asterisk? This is the table structure we are using, based on Asterisk wiki: mysql describe cdr; +-+---+--+-+-++ | Field | Type | Null | Key | Default | Extra | +-+---+--+-+-++ | id | mediumint(8) unsigned | NO | PRI | NULL| auto_increment | | calldate| datetime | NO | | -00-00 00:00:00 || | clid| varchar(80) | NO | | || | src | varchar(80) | NO | | || | dst | varchar(80) | NO | | || | dcontext| varchar(80) | NO | | || | channel | varchar(80) | NO | | || | dstchannel | varchar(80) | NO | | || | lastapp | varchar(80) | NO | | || | lastdata| varchar(80) | NO | | || | duration| int(11) | NO | | 0 || | billsec | int(11) | NO | | 0 || | disposition | varchar(45) | NO | | || | amaflags| int(11) | NO | | 0 || | accountcode | varchar(20) | NO | | || | uniqueid| varchar(32) | NO | | || | userfield | varchar(255) | NO | | || | peeraccount | varchar(20) | NO | | || | linkedid| varchar(32) | NO | | || | sequence| int(11) | NO | | 0 || +-+---+--+-+-++ 20 rows in set (0.01 sec) Thank you! 2013/6/4 Olivier oza_4...@yahoo.fr OK, then I'll go with linkedid, uniqueid and sequence number. Thanks for sharing this on this list 2013/6/3 Matthew Jordan mjor...@digium.com On 06/03/2013 11:20 AM, Olivier wrote: Hi, When dealing with CDR SQL tables, I always added an auto-incremented cdr_id key as a primary key, just in case provided uniqueid key went wrong. Now I'm facing a situation where I need to insert into a database's table and from the dialplan, a reference to the CDR record which is currently processed. So my questions are: 1. Can uniqueid/sequence (or uniqueid/sequence/calldate) bundle be safely used as CDR's table primary key (ie I cannot have any uniqueid/sequence combination from one CDR record to match a past uniqueid/sequence combination) ? Possibly. Things to keep in mind: * You can run into uniqueid collisions across multiple systems if you do not specify a system name in asterisk.conf or do not specify a unique system name in asterisk.conf. * You can run into uniqueid collisions if your system clock goes backwards for any reason (the uniqueid for a channel happens to use a timestamp for its uniqueness) Whether or not this is unique enough will be completely dependent on your overall system configuration. In general, the recommended combination that *should* uniquely specify a CDR (when configured correctly) is linkedid (which should be enabled and added to your schema), uniqueid, and sequence number, with the asterisk system name specified. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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 -- Jairo Molina Jr∴ http://www.intermol.com.br -- _ -- 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
Re: [asterisk-users] Is uniqueid/sequence a safe CDR table primary key ?
Yes, using cdr_adaptive_odbc.conf. As it is a new table, just changed the name from calldate to start and now it is inserting the field ok. Thank you very much for your help. Best. 2013/6/11 Kevin Larsen kevin.lar...@pioneerballoon.com Are you using cdr_adaptive_odbc.conf to populate it? If so, there is no Asterisk analog to calldate. You would need an alias set up. Mine looks like: alias start = calldate so that the start of my call is what gets logged to the database as the calldate. Kevin Larsen From:Jairo ja...@intermol.com.br To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date:06/11/2013 08:28 AM Subject:Re: [asterisk-users] Is uniqueid/sequence a safe CDR table primary key ? Sent by:asterisk-users-boun...@lists.digium.com -- Hello, Still about CDR and MySQL table, should the calldate field be inserted by Asterisk? This is the table structure we are using, based on Asterisk wiki: mysql describe cdr; +-+---+--+-+-++ | Field | Type | Null | Key | Default | Extra | +-+---+--+-+-++ | id | mediumint(8) unsigned | NO | PRI | NULL| auto_increment | | calldate| datetime | NO | | -00-00 00:00:00 || | clid| varchar(80) | NO | | || | src | varchar(80) | NO | | || | dst | varchar(80) | NO | | || | dcontext| varchar(80) | NO | | || | channel | varchar(80) | NO | | || | dstchannel | varchar(80) | NO | | || | lastapp | varchar(80) | NO | | || | lastdata| varchar(80) | NO | | || | duration| int(11) | NO | | 0 || | billsec | int(11) | NO | | 0 || | disposition | varchar(45) | NO | | || | amaflags| int(11) | NO | | 0 || | accountcode | varchar(20) | NO | | || | uniqueid| varchar(32) | NO | | || | userfield | varchar(255) | NO | | || | peeraccount | varchar(20) | NO | | || | linkedid| varchar(32) | NO | | || | sequence| int(11) | NO | | 0 || +-+---+--+-+-++ 20 rows in set (0.01 sec) Thank you! 2013/6/4 Olivier *oza_4...@yahoo.fr* oza_4...@yahoo.fr OK, then I'll go with linkedid, uniqueid and sequence number. Thanks for sharing this on this list 2013/6/3 Matthew Jordan *mjor...@digium.com* mjor...@digium.com On 06/03/2013 11:20 AM, Olivier wrote: Hi, When dealing with CDR SQL tables, I always added an auto-incremented cdr_id key as a primary key, just in case provided uniqueid key went wrong. Now I'm facing a situation where I need to insert into a database's table and from the dialplan, a reference to the CDR record which is currently processed. So my questions are: 1. Can uniqueid/sequence (or uniqueid/sequence/calldate) bundle be safely used as CDR's table primary key (ie I cannot have any uniqueid/sequence combination from one CDR record to match a past uniqueid/sequence combination) ? Possibly. Things to keep in mind: * You can run into uniqueid collisions across multiple systems if you do not specify a system name in asterisk.conf or do not specify a unique system name in asterisk.conf. * You can run into uniqueid collisions if your system clock goes backwards for any reason (the uniqueid for a channel happens to use a timestamp for its uniqueness) Whether or not this is unique enough will be completely dependent on your overall system configuration. In general, the recommended combination that *should* uniquely specify a CDR (when configured correctly) is linkedid (which should be enabled and added to your schema), uniqueid, and sequence number, with the asterisk system name specified. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: *http://digium.com* http://digium.com/ * http://asterisk.org* http://asterisk.org/ -- _ -- Bandwidth and Colocation Provided by *http://www.api-digital.com*http://www.api-digital.com/-- New
[Asterisk-Users] Unable to reconfigure channel
Hi all, I have a problem with the cvs head zaptel library: I cannot update my zapata.conf into asterisk when I issue the reload command from the CLI prompt; only when I stop and restart the asterisk service. The system sends the following message: Jun 20 09:49:14 WARNING[3754]: chan_zap.c:10364 setup_zap: Ignoring switchtype Jun 20 09:49:14 WARNING[3754]: chan_zap.c:10364 setup_zap: Ignoring signalling -- Reconfigured channel 1, FXS Kewlstart signalling -- Reconfigured channel 2, FXS Kewlstart signalling -- Reconfigured channel 3, FXS Kewlstart signalling Jun 20 09:49:14 ERROR[3754]: chan_zap.c:9827 setup_zap: Unable to reconfigure channel '4' Jun 20 09:49:14 WARNING[3754]: chan_zap.c:10567 reload: Reload of chan_zap.so is unsuccessful! I have the simplest configuration: [trunkgroups] [channels] language=en context=default switchtype=national signalling=fxs_ks channel = 1 channel = 2 channel = 3 I was using the cvs head version because I need the wctdm driver for the TDM04B (4*fxo modules) Does anybody know what is wrong? Thanks in advance!! Jairo Barahona Garita inCom Developer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 (Digium E100P) problem : B-channel succesfully restarted.
Hi! I have an Asterisk Box with one E1. This is connected with PSTN. My problem is that periodically the Asterisk console shows the following message. -- B-channel 0/1 succesfully restarted on span 1 -- B-channel 0/2 succesfully restarted on span 1 -- B-channel 0/3 succesfully restarted on span 1 -- B-channel 0/4 succesfully restarted on span 1 -- B-channel 0/5 succesfully restarted on span 1 -- B-channel 0/6 succesfully restarted on span 1 -- B-channel 0/7 succesfully restarted on span 1 -- B-channel 0/8 succesfully restarted on span 1 -- B-channel 0/9 succesfully restarted on span 1 -- B-channel 0/10 succesfully restarted on span 1 -- B-channel 0/11 succesfully restarted on span 1 -- B-channel 0/12 succesfully restarted on span 1 -- B-channel 0/13 succesfully restarted on span 1 -- B-channel 0/14 succesfully restarted on span 1 -- B-channel 0/15 succesfully restarted on span 1 -- B-channel 0/16 succesfully restarted on span 1 -- B-channel 0/17 succesfully restarted on span 1 -- B-channel 0/18 succesfully restarted on span 1 -- B-channel 0/19 succesfully restarted on span 1 -- B-channel 0/20 succesfully restarted on span 1 -- B-channel 0/21 succesfully restarted on span 1 -- B-channel 0/22 succesfully restarted on span 1 -- B-channel 0/23 succesfully restarted on span 1 -- B-channel 0/24 succesfully restarted on span 1 -- B-channel 0/25 succesfully restarted on span 1 -- B-channel 0/26 succesfully restarted on span 1 -- B-channel 0/27 succesfully restarted on span 1 -- B-channel 0/28 succesfully restarted on span 1 -- B-channel 0/29 succesfully restarted on span 1 -- B-channel 0/30 succesfully restarted on span 1 -- B-channel 0/31 succesfully restarted on span 1 I have look for in the source code to check the meaning of this message but I haven't found anything. The only thing that I could check, It's that T203=10 seg and T200=1 seg. For example, Cisco has T203=30 seg. In http://www.acacia-net.com/wwwcla/protocol/q921.htm I read the meaning of those timers. Could Asterisk restart E1 because It don't recieve anything before T203 finished? My configuration is: /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 I don't know if this behavior is correct. Can someone help me? Thanks in advance. __ Renovamos el Correo Yahoo!: ¡250 MB GRATIS! Nuevos servicios, más seguridad http://correo.yahoo.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: E1 (Digium E100P) problem : B-channel succesfully restarted
Hi! I have an Asterisk Box with one E1. This is connected with PSTN. My problem is that periodically the Asterisk console shows the following message. -- B-channel 0/1 succesfully restarted on span 1 -- B-channel 0/2 succesfully restarted on span 1 [..etc...] I don't know if this behavior is correct. Can someone help me? Perfectly normal behaviour. Once an hour, idle channels are restarted, but active channels are left undisturbed. Thanks tony, why are the idle channels restarted?, the PSTN could think that my box has some problem. Perhaps, other equipments also make the same but I don't know anything about it. I used Access Server of Cisco and I think that the channels weren't restarted. __ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stun codec
You can use Ethereal to see what your phone (stun) is sending. Of this way you can see the RTP ports and IP public that your phones are going to use. You can see that information in INVITE and OK packets. For other hand, If you use one router with symmetrical NAT then Stun won't work http://www.networkmagazine.com/shared/article/showArticle.jhtml?articleId=17602009classroom= __ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and T.38.
Hi Rafa! Hi Jairo, Try with other values for the jitter in your Gateway (H323). One customer have a scenario like this: Phone/Fax Gateways H323 -with 16/8/2/1 Port FXS- --- GNUGK --- Asterisk --- Zap (E1) We have the same configuration, and I think that our problem is the jitter in Asterisk, since we have jitterMin=0ms and jitterMax=0ms. We changed chan_oh323.c because it didn't accept jitter less than 20ms (We used Asterisk to translate SIP -- H323--Cisco, and the Cisco have jitter buffer, so we didn't need jitter buffer in Asterisk). Now, we have put: oh323.conf --- jitterMin=20ms jitterMax=20ms codec=G711A frames=20 zapata.conf --- jitterbuffers=20ms but fax still doesn't work. Anyideas? Other question. If Asterisk hasn't T.38 then Fax will need 64Kbps (G711)+ 16Kbps (IP+UDP+RTP) bandwidth, and that is much bandwidth. Is this correct?, Do you know any comercial implementation of T.38 to Asterisk? Thanks. __ Renovamos el Correo Yahoo!: ¡250 MB GRATIS! Nuevos servicios, más seguridad http://correo.yahoo.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and T.38.
Hi! I want to send fax to PSTN with Asterisk, but by now I can't. I am using the following boxs: Internet Zap E1 Phone/Fax Gateway(H323)--- Asterisk--- PSTN The gateway H323 has T38 and T30. Before I began with Asterisk, I used Cisco to connect with PSTN, and the Fax worked very well in T38 (T30 didn't work, I think the reason is the jitter). I have read that Asterisk doesn't support T.38, is this correct?, is there any comercial implementation in Asterisk?. If Asterisk doesn't support T.38, how can I use it to send fax to PSTN? Thanks in advance. __ Renovamos el Correo Yahoo!: ¡250 MB GRATIS! Nuevos servicios, más seguridad http://correo.yahoo.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Clients
Hello, I am new in Asterisk... and I didn't get to find this information in the site or in the links that appears in the support section... The one that I want to know is which are the compatible softwares with Asterisk (PC working as an extension of the PBX)... Theoretically, every H.323 client could be connected to Asterisk... am I right? With for instance, the projects of the openh323 (http://www.openh323.org)??? Thank you, Jairo Cavanus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
