[asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to new
Hi
I am using following say.conf file. Its a default file, which comes with
Asterisk installation.
When I call SAY DATETIME AGI function, it simply returns without playing
date time. Where as if I use mode=old setting, it works. Is this a bug
or mode=new is not implemented for SAY DATETIME AGI function?
[general]
mode=new; method for playing numbers and dates
; old - using asterisk core function
; new - using this configuration file
[digit-base](!) ; base rule for digit strings
; XXX incomplete yet
_digit:[0-9] = digits/${SAY}
_digit:[-] = letters/dash
_digit:[*] = letters/star
_digit:[@] = letters/at
_digit:[0-9]. = digit:${SAY:0:1}, digit:${SAY:1}
[date-base](!) ; base rules for dates and times
; the 'SAY' variable contains MMDDHHmm.ss-dow-doy
; these rule map the strftime attributes.
_date:Y:. = num:${SAY:0:4} ; year, 19xx
_date:[Bbh]:. = digits/mon-$[${SAY:4:2}-1] ; month name, 0..11
_date:[Aa]:. = digits/day-${SAY:16:1} ; day of week
_date:[de]:. = num:${SAY:6:2} ; day of month
_date:[H]:. = digits/oh, num:${SAY:8:2} ; hour (oh one, oh two,
..., oh nine, ten, eleven, ..., twenty-three)
_date:[I]:. = num:$[${SAY:8:2} % 12] ; hour 0-12
_date:[M]:. = num:${SAY:10:2} ; minute
; XXX too bad the '?' function does not remove the quotes
; _date:[pP]:. = digits/$[ ${SAY:10:2} 12 ? p-m :: a-m]
; am pm
_date:[pP]:. = digits/p-m ; am pm
_date:[S]:. = num:${SAY:13:2} ; seconds
_date:[Ii]:. = num:$[${SAY:8:2} % 12] ; hour 0-12
_date:[k]:. = num:${SAY:8:2} ; hour (one, two. three, ...,twenty three
_date:[m]:. = num:${SAY:4:2} ; month number
_date:[Q]:. = date:dby ;incompleat ; today, yesterday or ABdY
_date:[q]:. = date:dby ;incompleat ; (for today), yesterday,
weekday, or ABdY
_date:[R]:. = date:HM${SAY} ; 24 hour time, including minute
_date:[T]:. = date:HMS${SAY} ; 24 hour, minure, seconds
[en-base](!)
_[n]um:0. = num:${SAY:1}
_[n]um:X = digits/${SAY}
_[n]um:1X = digits/${SAY}
_[n]um:[2-9]0 = digits/${SAY}
_[n]um:[2-9][1-9] = digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:X00 = num:${SAY:0:1}, digits/hundred
_[n]um:XXX = num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:X000 = num:${SAY:0:1}, digits/thousand
_[n]um: = num:${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XX000 = num:${SAY:0:2}, digits/thousand
_[n]um:X = num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:XXX000 = num:${SAY:0:3}, digits/thousand
_[n]um:XX = num:${SAY:0:3}, digits/thousand, num:${SAY:3}
_[n]um:X00 = num:${SAY:0:1}, digits/million
_[n]um:XXX = num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:XX00 = num:${SAY:0:2}, digits/million
_[n]um: = num:${SAY:0:2}, digits/million, num:${SAY:2}
_[n]um:XXX00 = num:${SAY:0:3}, digits/million
_[n]um:X = num:${SAY:0:3}, digits/million, num:${SAY:3}
_[n]um:X0 = num:${SAY:0:1}, digits/billion
_[n]um:XX = num:${SAY:0:1}, digits/billion, num:${SAY:1}
_[n]um:XX0 = num:${SAY:0:2}, digits/billion
_[n]um:XXX = num:${SAY:0:2}, digits/billion, num:${SAY:2}
_[n]um:XXX0 = num:${SAY:0:3}, digits/billion
_[n]um: = num:${SAY:0:3}, digits/billion, num:${SAY:3}
; enumeration
_e[n]um:X = digits/h-${SAY}
_e[n]um:1X = digits/h-${SAY}
_e[n]um:[2-9]0 = digits/h-${SAY}
_e[n]um:[2-9][1-9] = num:${SAY:0:1}0, digits/h-${SAY:1}
_e[n]um:[1-9]00 = num:${SAY:0:1}, digits/h-hundred
_e[n]um:[1-9]XX = num:${SAY:0:1}, digits/h-hundred, enum:${SAY:1}
[en_GB](date-base,digit-base,en-base)
_[n]um:XXX = num:${SAY:0:1}, digits/hundred, vm-and, num:${SAY:1}
Thanks Regards,
Amit Patkar
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Re: [asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to new
On Mon, Jul 1, 2013 at 6:24 AM, Amit Patkar | ATPL a...@avhan.com wrote: Hi I am using following say.conf file. Its a default file, which comes with Asterisk installation. When I call SAY DATETIME AGI function, it simply returns without playing date time. Where as if I use mode=old setting, it works. Is this a bug or mode=new is not implemented for SAY DATETIME AGI function? Processing of say.conf is performed by the app_playback module, not by the Asterisk core itself. Do you have the app_playback module loaded? If you run the CLI command say load, what does it report? Matt -- _ -- 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 behind NAT and Kamailio -- Internal IP in SDP and not externip
Hello, On 6/28/13 4:29 PM, Johan Wilfer wrote: Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the external. This is the setup: Teleco - Kamailio - Asterisk SIP -- 1.2.3.4 10.0.0.1 -- 10.0.0.2 externip=1.2.3.5 localnet=10.0.0.0/255.255.255.0 RTP 1.2.3.5 (NAT:ed to 10.0.0.2) On an incomming call from the teleco - to kamailio (public addr) - to asterisk in the private net. Asterisk responds with the following SDP: v=0 o=root 1889 1889 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 23344 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Asterisk seems to think that because the proxy is on the localnet, the media is too, so it doesn't use the externip as the RTP-ip. This is a incomming call and the RTP ip of the other leg is another public address. So the RTP-ip should the public address (externip). If I connect to the teleco directly from the pbx (bypassing kamailio) Asterisk correctly uses the externip as the rtp-ip in the SDP. I know this is an old and unsupported version of Asterisk, but any input on the topic is welcome. If this is supported in later versions we can maybe work around until we migrate later. what I did when I had similar scenario was to let asterisk completely behind NAT, using only the local IP. I used rtpproxy running on the same host as kamailio to bridge the rtp between external and internal networks. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda -- _ -- 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 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to new
Thanks Matt. app_playback module is loaded. I am able to play numbers.Issue is only with date time and only with SAY DATETIME function. If I use SAY DATE, date is getting played. Do I need to check some other settings? Thanks Regards, Amit Patkar On 7/1/2013 6:03 PM, Matthew Jordan wrote: On Mon, Jul 1, 2013 at 6:24 AM, Amit Patkar | ATPL a...@avhan.com mailto:a...@avhan.com wrote: Hi I am using following say.conf file. Its a default file, which comes with Asterisk installation. When I call SAY DATETIME AGI function, it simply returns without playing date time. Where as if I use mode=old setting, it works. Is this a bug or mode=new is not implemented for SAY DATETIME AGI function? Processing of say.conf is performed by the app_playback module, not by the Asterisk core itself. Do you have the app_playback module loaded? If you run the CLI command say load, what does it report? Matt -- _ -- 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] Send event/notification from one channel driver ot another
On Sun, Jun 30, 2013 at 11:03 PM, Meadows Hoa meadows_...@yahoo.com wrote: Is there a way to send an event notification from one Channel Driver to another Channel Driver? If I wanted to have one channel driver do something custom depending on the call state of the other channel driver involved in the call, how could you send some kind of event or state notification directly to this other channel driver? Yes, although in general this happens within the Asterisk code itself and is not exposed to the dialplan. Typically, if anything (channel driver, resource module, etc.) wants to notify a channel that something has occurred, it queues a control frame on that channel. Control frames include things like media source changes/media updates, indications that signalling actions should take place, etc. What specific use case are you looking at? Matt -- 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
Re: [asterisk-users] CEL logging and queue APP_START/END, maybe an issue?
On Sun, Jun 30, 2013 at 4:24 PM, Fabio Moretti fmore...@tecytal.com wrote: Hi, I'm still doing experiment with CEL and recently I've noticed that if: 1- call enter in the pbx 2- call enter a queue 3- call get answered 4- the operators call directly another queue, using the queue number 5- the call get answered 6- the call end the point 4 does not generate a correct logging. In my case I have the extension 1006 answering the call with linkedid 1371568201.486360, then call queue number 21. I get only a BRIDGE_UPDATE event: '1812621', 'BRIDGE_UPDATE', '2013-06-18 12:11:31', 'ISS23071100', '23071100', '23071100', '24872939', '', '21', 'from-internal', 'Local/1006@from-queue-00034a00;2', '', '', '', '', 'Queue', '21,tr,,', '3', '', '1371568269.486380', '1371568201.486360', 'Local/714126@from-queue-00034a04;1', '', '', '' here you can se the extension called (21), the source channel (Local/1006@from-queue-00034a00), the peer (Local/714126@from-queue-00034a04) and especially you can see that the app name is queue. After this event I have only an APP_END for queue number 21, but no CHAN_START/END for the peer channel, no queue events, nothing. This is if I follow the call with linkedid 1371568201.486360. If I search the uniqueid of the BRIDGE_UPDATE as linkedid (1371568269.486380) I get the whole queue 21 call history, as I expected to find in the original call linkedid. I think that because the call is the same the linkedid shouldn't change at all, but in this case the cel logging is generating a sub cel, is this by design? I'm supposed to check if the current event have generate a sub-cel to reconstruct the call histoery completely? And if it is by design, isn't a bug put the queue APP_START event in the sub-cel linkedid and the APP_END in the original linkedid? If someone can have a look I've attached the CSV for the two linkedids.. Nope, this is entirely expected. A BRIDGE_UPDATE CEL event occurs when a masquerade happens and the participants in a bridge have been altered - that is, a channel came in and pushed one of the channels in the bridge out. In this particular case, Local/714126@from-queue-00034a04;1 has replaced the channel Local/1006@from-queue-00034a00;2 was bridged with, SIP/1006-7705. You can see that SIP/1006-7705 is disposed of immediately following this event. When a BRIDGE_UPDATE happens, you have to start pulling the records from the new channel in with whoever is still in the bridge. This usually means that someone's linkedid changed (as the participants changed). linkedids absolutely do change on a channel in this scenario - when two channels are bridged (which is what has happened when the BRIDGE_UPDATE occurs - the Local channel pushed the SIP channel out), the linkedids are updated on the participants based on who had the older linkedid. In this case, Local/714126@from-queue-00034a04;1's linkedid was updated. So if you want to know everything that happened with that Local channel, you have to tie together both the current linkedid as well as what was its previous linkedid. When you're dealing with CEL, you're operating on a level much closer to what Asterisk is actually doing with its channels. This means having to deal with Local channel pairs and - more importantly - masquerades. This is a whole lot more powerful than CDRs, but does mean that you have to do some bookkeeping to keep track of the channel states. On a side note, the fact that masquerades are hard and tend to require people to do lots of updates was a driving factor in the development efforts that went on in 12. Masquerades are now an implementation detail, so in the future, you won't have to deal with BRIDGE_UPDATE. Matt -- 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
[asterisk-users] AudioCodes MP-112
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Re: [asterisk-users] AudioCodes MP-112
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