Re: [asterisk-users] Choosing codecs
Zitat von A J Stiles asterisk_l...@earthshod.co.uk: It will be in the /etc/asterisk/*.conf file for the appropriate calling technology. So if the calls are going over a SIP trunk, it will be in sip.conf . You want disallow=all allow=alaw There probably will be some other allow= lines; just stick a semicolon in front of the ones you do *not* want, to comment them out. Then issue core reload in Asterisk CLI, and all your calls should be A-law from now on. OK, thanks a lot! Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Unisteam not showing callerid
hi list can U help me caller id in USTM if now working -- Starting switch on '4211@4211-1' to 4203 -- Executing [4203@office:1] DumpChan(USTM/4211@4211-0x7f7ba4228fd0, ) in new stack Dumping Info For Channel: USTM/4211@4211-0x7f7ba4228fd0: Info: Name= USTM/4211@4211-0x7f7ba4228fd0 Type= USTM UniqueID= 1436177628.8051 LinkedID= 1436177628.8051 CallerIDNum=(N/A) CallerIDName= (N/A) ConnectedLineIDNum= (N/A) ConnectedLineIDName=(N/A) DNIDDigits= (N/A) RDNIS= (N/A) Parkinglot= Language= en State= Ring (4) Rings= 0 NativeFormat= (ulaw|alaw) WriteFormat=ulaw ReadFormat= ulaw RawWriteFormat= ulaw RawReadFormat= ulaw WriteTranscode= No ReadTranscode= No 1stFileDescriptor= 1652 Framesin= 0 Framesout= 0 TimetoHangup= 0 ElapsedTime=0h0m0s BridgeID= (Not bridged) Context=office Extension= 4203 Priority= 1 CallGroup= 4 PickupGroup=4 Application=DumpChan Data= (Empty) Blocking_in=(Not Blocking) Variables: -- Executing [4203@office:2] Dial(USTM/4211@4211-0x7f7ba4228fd0, USTM/4203@4203) in new stack my CONF general] port=5000 [unistim-phones](!) bookmark=Support@123;Softkey to speed dial context=office ; extension=line rtp_method=1 maintext0=GREETING ; default = Welcome, 24 characters max maintext1=have a great day ; default = the name of the device, 24 characters max dateformat=1; 0 = month/day, 1 (default) = day/month timeformat=2; 0 = 0:00am ; 1 (default) = 0h00, 2 = 0:00 callhistory=0 ; 0 = disable, 1 = enable call history, default = 1 [group3] callgroup = 3 pickupgroup = 3 [group4] callgroup = 4 pickupgroup = 4 [group5] callgroup = 3,5 pickupgroup = 3,5 [group6] callgroup = 6 pickupgroup = 6 [4294](unistim-phones,group3) device=0016caf460f5 line= 4294 callerid=Victoriya Mukan 4294 [4211](unistim-phones,group4) device=000ae475faed line= 4211 callerid=Gomenyuk tatyana 4211 -- Best regards Antony моб (066) 919-75-33 моб (063) 656-43-40 satski...@gmail.com mail%3asatski...@gmail.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] Choosing codecs
On Monday 06 Jul 2015, Luca Bertoncello wrote: Zitat von A J Stiles asterisk_l...@earthshod.co.uk: Yes. You should definitely be using A-law for calls to the Outside World. Well, I wanted to change these settings, but I'm not sure, where I have to do that... I think in the users.conf, but I think, the allow keywords is for the network... How can I change this setting? It will be in the /etc/asterisk/*.conf file for the appropriate calling technology. So if the calls are going over a SIP trunk, it will be in sip.conf . You want disallow=all allow=alaw There probably will be some other allow= lines; just stick a semicolon in front of the ones you do *not* want, to comment them out. Then issue core reload in Asterisk CLI, and all your calls should be A-law from now on. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Choosing codecs
On Monday 06 Jul 2015, Luca Bertoncello wrote: Well, but for voice quality, which codec is better? alaw or gsm? A-law is better for voice quality (sorry, thought my original explanation was obvious). But note that if the destination is a mobile phone, GSM will be used anyway, at least for the link between the final cell tower and the handset. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Choosing codecs
Zitat von A J Stiles asterisk_l...@earthshod.co.uk: Yes. You should definitely be using A-law for calls to the Outside World. If you use a different codec, then your telephone company will either transcode it for you (if it is one they understand) or just block the call (if not). Even if you are trying to use A-law to call a mobile phone, the transcoding to GSM for the final leg to and from the handset will be taken care of by the mobile company's equipment. OK, I'll change the settings! Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Choosing codecs
On Sunday 05 Jul 2015, Luca Bertoncello wrote: Hi list! I noticed that when the phone of my wife calls the gsm codec will be used, but if someone calls the phone, alaw will be used: Could someone explain me why? Second question: I think, ulaw/alaw are better then gsm, isn't it? If so, how can I change it? GSM is the native codec used for calls to mobile phones; it uses lossy compression to achieve a low bit rate. A-law is the native codec used by physical exchanges on the land line network (PSTN and ISDN). It is non-lossy. It works by arranging the steps closer together near the zero line, and further apart away from it; so the difference between the actual signal and the nearest digital representation is small in proportion to the signal. To force the use of a-law, you need something like disallow=all allow=alaw at the top of the configuration file for the calling technology in question (sip.conf for SIP, chan_dahdi.conf for DAHDI, c.). If you want to force a specific device to use a specific codec, then put an allow in the section for that device. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Choosing codecs
Zitat von A J Stiles asterisk_l...@earthshod.co.uk: On Monday 06 Jul 2015, Luca Bertoncello wrote: Well, but for voice quality, which codec is better? alaw or gsm? A-law is better for voice quality (sorry, thought my original explanation was obvious). But note that if the destination is a mobile phone, GSM will be used anyway, at least for the link between the final cell tower and the handset. OK, thank you... Maybe will be your explanation other day but mondays obvious... :D So, I think, I should try to force the using of alaw for this phone, is it right? Usually we don't call mobile phones from our landline... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Choosing codecs
Zitat von A J Stiles asterisk_l...@earthshod.co.uk: Yes. You should definitely be using A-law for calls to the Outside World. Well, I wanted to change these settings, but I'm not sure, where I have to do that... I think in the users.conf, but I think, the allow keywords is for the network... How can I change this setting? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Choosing codecs
Zitat von A J Stiles asterisk_l...@earthshod.co.uk: Hi, GSM is the native codec used for calls to mobile phones; it uses lossy compression to achieve a low bit rate. A-law is the native codec used by physical exchanges on the land line network (PSTN and ISDN). It is non-lossy. It works by arranging the steps closer together near the zero line, and further apart away from it; so the difference between the actual signal and the nearest digital representation is small in proportion to the signal. Well, but for voice quality, which codec is better? alaw or gsm? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Choosing codecs
On Monday 06 Jul 2015, Luca Bertoncello wrote: So, I think, I should try to force the using of alaw for this phone, is it right? Usually we don't call mobile phones from our landline... Yes. You should definitely be using A-law for calls to the Outside World. If you use a different codec, then your telephone company will either transcode it for you (if it is one they understand) or just block the call (if not). Even if you are trying to use A-law to call a mobile phone, the transcoding to GSM for the final leg to and from the handset will be taken care of by the mobile company's equipment. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How may SIP 183 messages a caller receives when many callee rings?
Hi. I have a beginner conceptual question about Asterisk: Let's suppose that there are 4 softphones registered in my Asterisk and all of them are currently online. In addiction , there is no call. Suddenly, one of these softphones sends a SIP message to the Asterisk. In this case the dialplan will execute the instruction ' exten = 2005,1,Dial(SIP/2000SIP/2001SIP/2002, 30) ' All softphones (2000, 2001 and 2002) will ring. These are proprietary softphones and all of then will reply with SIP 183 message. SIP 183 will contain SDP with media information. The question is: Will the caller receive SIP 183 from each callee? That is, will it receive 3 SIP 183 messages? It is important to the caller receives a SIP 183 message from each callee, because this caller needs to send early media (video) to every callee. Or, will Asterisk send just one message SIP 183 to the caller, with some kind of generic SDP message? Any hint will be very helpful! Thanks a lot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) -- _ -- 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 how to setup alarm too many outgoing calls from same user
I don't think you can do this natively within Asterisk, but take a look at SecAst (from http://www.telium.cahttp://www.telium.ca/ ). There is a free edition you can download right from the web site. SecAst will monitor the rate at which a user/device places calls to detect potential fraud. (I assume that is what you are trying to achieve). It also checks for suspicious dialed digits/patterns, geographic location of the caller based on IP, etc...it may be overkill if this is just a small home system though. Forwarded Message Subject:[asterisk-users] Asterisk how to setup alarm too many outgoing calls from same user Date: Mon, 06 Jul 2015 07:27:43 -0700 From: Motty Cruz motty.c...@gmail.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hello, I would like to setup a mechanism to trigger an alarm if user is deal too many numbers within a very short period of time. Safeguard against users hacked accounts. can someone help? 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 -- _ -- 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: saycid without prefix
The easiest solution may be to strip the leading zero's off your caller ID
before your caller enters the Voicemail app to leave you a message.
ExecIf(REGEX(^[0][0].
${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2}))
On Fri, Jul 3, 2015 at 10:53 PM, Luca Bertoncello lucab...@lucabert.de
wrote:
Hi list!
Yesterday I set up a voicemail on my Asterisk 1.8.
It works as expected, but I'd like to have the CID without unnecessary
prefix...
Right now, if I call from my mobile phone I hear the complete prefix for my
mobile number, indeed without 00.
So I hear message from 49177
How can I set Asterisk to just read the prefix if it's necessary (so that
calls from german numbers will not have 0049)?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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Re: [asterisk-users] CDR in an MySQL-Database
Hi list!
I'd like to save all information about calls (CDR) in a MySQL-Database.
I created the DB and a user for Asterisk on a separate server, then I
configured my cdr_mysql.conf so:
[global]
hostname=192.168.10.3
dbname=asterisk
table=cdr
password=MYSECRET
user=asterisk
port=3306
and my cdr.conf so:
[general]
enable=yes
unanswered = yes
safeshutdown=yes
[mysql]
usegmtime=no
loguniqueid=yes
loguserfield=yes
accountlogs=yes
I created the table in the DB so:
CREATE TABLE IF NOT EXISTS `cdr` (
`id` int(11) unsigned NOT NULL AUTO_INCREMENT,
`calldate` datetime NOT NULL DEFAULT '-00-00 00:00:00',
`clid` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`src` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`dst` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`dcontext` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`lastapp` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
`lastdata` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
`duration` float unsigned DEFAULT NULL,
`billsec` float unsigned DEFAULT NULL,
`disposition` enum('ANSWERED','BUSY','FAILED','NO ANSWER','CONGESTION')
COLLATE utf8_bin DEFAULT NULL, `channel` varchar(50) COLLATE utf8_bin DEFAULT
NULL, `dstchannel` varchar(50) COLLATE utf8_bin DEFAULT NULL,
`amaflags` varchar(50) COLLATE utf8_bin DEFAULT NULL,
`accountcode` varchar(20) COLLATE utf8_bin DEFAULT NULL,
`uniqueid` varchar(32) COLLATE utf8_bin NOT NULL DEFAULT '',
`userfield` float unsigned DEFAULT NULL,
`answer` datetime NOT NULL,
`end` datetime NOT NULL,
PRIMARY KEY (`id`),
KEY `calldate` (`calldate`),
KEY `dst` (`dst`),
KEY `src` (`src`),
KEY `dcontext` (`dcontext`),
KEY `clid` (`clid`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8 COLLATE=utf8_bin AUTO_INCREMENT=1 ;
Then I restarted Asterisk (core restart now).
Unfortunately it does not work, since I get on boot:
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1468 load_mysql_config:
MySQL RealTime: No database user found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1474 load_mysql_config:
MySQL RealTime: No database password found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1480 load_mysql_config:
MySQL RealTime: No database host found, using localhost via socket.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1486 load_mysql_config:
MySQL RealTime: No database name found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1492 load_mysql_config:
MySQL RealTime: No database port found, using 3306 as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1509 load_mysql_config:
MySQL RealTime: No database socket found (and unable to detect a suitable path).
And of course:
OpenWrt*CLI cdr show status
Call Detail Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: Yes
* Registered Backends
---
cdr-custom
Asterisk 1.8 runs on an OpenWRT-Switch.
Any idea?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
Did you study this: http://www.asteriskdocs.org/ ?
jg
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Re: [asterisk-users] CDR in an MySQL-Database
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan, Travis
Sent: Monday, July 06, 2015 4:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] CDR in an MySQL-Database
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
Sent: Monday, July 06, 2015 4:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR in an MySQL-Database
Hi list!
I'd like to save all information about calls (CDR) in a MySQL-Database.
I created the DB and a user for Asterisk on a separate server, then I
configured my cdr_mysql.conf so:
[global]
hostname=192.168.10.3
dbname=asterisk
table=cdr
password=MYSECRET
user=asterisk
port=3306
and my cdr.conf so:
[general]
enable=yes
unanswered = yes
safeshutdown=yes
[mysql]
usegmtime=no
loguniqueid=yes
loguserfield=yes
accountlogs=yes
I created the table in the DB so:
CREATE TABLE IF NOT EXISTS `cdr` (
`id` int(11) unsigned NOT NULL AUTO_INCREMENT,
`calldate` datetime NOT NULL DEFAULT '-00-00 00:00:00',
`clid` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`src` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`dst` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`dcontext` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`lastapp` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
`lastdata` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
`duration` float unsigned DEFAULT NULL,
`billsec` float unsigned DEFAULT NULL,
`disposition` enum('ANSWERED','BUSY','FAILED','NO ANSWER','CONGESTION')
COLLATE utf8_bin DEFAULT NULL, `channel` varchar(50) COLLATE utf8_bin DEFAULT
NULL, `dstchannel` varchar(50) COLLATE utf8_bin DEFAULT NULL,
`amaflags` varchar(50) COLLATE utf8_bin DEFAULT NULL,
`accountcode` varchar(20) COLLATE utf8_bin DEFAULT NULL,
`uniqueid` varchar(32) COLLATE utf8_bin NOT NULL DEFAULT '',
`userfield` float unsigned DEFAULT NULL,
`answer` datetime NOT NULL,
`end` datetime NOT NULL,
PRIMARY KEY (`id`),
KEY `calldate` (`calldate`),
KEY `dst` (`dst`),
KEY `src` (`src`),
KEY `dcontext` (`dcontext`),
KEY `clid` (`clid`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8 COLLATE=utf8_bin AUTO_INCREMENT=1 ;
Then I restarted Asterisk (core restart now).
Unfortunately it does not work, since I get on boot:
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1468 load_mysql_config:
MySQL RealTime: No database user found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1474 load_mysql_config:
MySQL RealTime: No database password found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1480 load_mysql_config:
MySQL RealTime: No database host found, using localhost via socket.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1486 load_mysql_config:
MySQL RealTime: No database name found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1492 load_mysql_config:
MySQL RealTime: No database port found, using 3306 as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1509 load_mysql_config:
MySQL RealTime: No database socket found (and unable to detect a suitable
path).
And of course:
OpenWrt*CLI cdr show status
Call Detail Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: Yes
* Registered Backends
---
cdr-custom
Asterisk 1.8 runs on an OpenWRT-Switch.
Any idea?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
Did you study this: http://www.asteriskdocs.org/ ?
jg
--
I was only able to make it work with ODBC. Looks like you're trying to do
realtime, and I'm not familiar. I have it working just fine with ODBC for the
CDR and realtime for everything else. There are more steps to getting odbc to
work, but apparently Digium only really supports the ODBC version of things,
and not the community supported realtime/native MySQL setup.
I think I just used the following to guide me.
http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc
Travis
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Re: [asterisk-users] Voicemail: saycid without prefix
John Kiniston johnkinis...@gmail.com schrieb:
The easiest solution may be to strip the leading zero's off your caller ID
before your caller enters the Voicemail app to leave you a message.
ExecIf(REGEX(^[0][0].
${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2}))
Thanks!
I already had this idea and implemented it.
It works...
Regards
Luca Bertoncello
(lucab...@lucabert.de)
--
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[asterisk-users] SIP/2.0 401 Unauthorized when calling from one SIP extension to another
Hello everyone, A few days ago I had a problem with a couple of extensions. I have about 12 Aastra 6731i phones, 6 are at our main office and 6 more on remote branches. We use VPN to communicate to our branches so there's no NAT involved any where. The problem I had was that I couldn't call from two extensions located at two branch offices. But I could call to them just fine. On any call placed from those phones I got the following error: SIP/2.0 401 Unauthorized This is the console output of a call placed from one of those phones: --- SIP read from UDP:192.168.96.141:5060 --- INVITE sip:85004@192.168.10.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.96.141:5060 ;branch=z9hG4bKd4511cd84a4f22669.bbb2635a0516602e8 Max-Forwards: 70 From: sip:85014@192.168.10.227:5060;tag=5dde10fb77 To: 85004 sip:85004@192.168.10.227:5060 Call-ID: 169216acc663493c CSeq: 28267 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: sip:85014@192.168.96.141:5060 ;transport=udp;+sip.instance=urn:uuid:--1000-8000-00085D2B85C3 Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 6731i/2.6.0.1007 Content-Type: application/sdp Content-Length: 698 v=0 o=MxSIP 0 0 IN IP4 192.168.96.141 s=SIP Call c=IN IP4 192.168.96.141 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 4 4 98 97 115 96 9 108 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:4 G723/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:108 G7221/16000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv - --- (14 headers 29 lines) --- Sending to 192.168.96.141:5060 (no NAT) Sending to 192.168.96.141:5060 (no NAT) Using INVITE request as basis request - 169216acc663493c Found peer '85014' for '85014' from 192.168.96.141:5060 --- Reliably Transmitting (NAT) to 192.168.96.141:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.96.141:5060 ;branch=z9hG4bKd4511cd84a4f22669.bbb2635a0516602e8;received=192.168.96.141;rport=5060 From: sip:85014@192.168.10.227:5060;tag=5dde10fb77 To: 85004 sip:85004@192.168.10.227:5060;tag=as52309181 Call-ID: 169216acc663493c CSeq: 28267 INVITE Server: Asterisk PBX 11.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=03eab1fd Content-Length: 0 Scheduling destruction of SIP dialog '169216acc663493c' in 32000 ms (Method: INVITE) --- SIP read from UDP:192.168.96.141:5060 --- ACK sip:85004@192.168.10.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.96.141:5060 ;branch=z9hG4bKd4511cd84a4f22669.bbb2635a0516602e8 Max-Forwards: 70 From: sip:85014@192.168.10.227:5060;tag=5dde10fb77 To: 85004 sip:85004@192.168.10.227:5060;tag=as52309181 Call-ID: 169216acc663493c CSeq: 28267 ACK User-Agent: Aastra 6731i/2.6.0.1007 Content-Length: 0 And that just keep repeating and repeating but the call never actually takes place. The contents of my sip.conf file: [general] context=unauthenticated allowguest=no srvlookup=no udpbindaddr=0.0.0.0 tcpenable=no shrinkcallerid=no [office-phone](!) type=peer context=LocalSets host=dynamic nat=force_rport,comedia dtmfmode=auto disallow=all allow=g729 [85004](office-phone) defaultuser=85004 secret=securepass callerid=Phone 4 85004 [85014](office-phone) defaultuser=85014 secret=securepass callerid=Phone 14 85014 host=192.168.96.141 transport=udp,tcp Originally I had not have the defaultuser option on any of the extensions, nor the host and transport on the [85014] one, but the problem was the same with or without those options. Note that I'm including only two extensions to simplify things up and that the extension with the problem is 85014. Also, I said there's no NAT involved here but I'm using the option nat=force_rport,comedia as suggested by Asterisk The Definitive Guide 4th edition. I've also switched that option to nat=no and the result was been the same. My dialplan is also really simple. extensions.conf file: [LocalSets] exten = 85004,1,Dial(SIP/85004) exten = 85014,1,NoOp() same =
[asterisk-users] CDR in an MySQL-Database
Hi list!
I'd like to save all information about calls (CDR) in a MySQL-Database.
I created the DB and a user for Asterisk on a separate server, then I
configured my cdr_mysql.conf so:
[global]
hostname=192.168.10.3
dbname=asterisk
table=cdr
password=MYSECRET
user=asterisk
port=3306
and my cdr.conf so:
[general]
enable=yes
unanswered = yes
safeshutdown=yes
[mysql]
usegmtime=no
loguniqueid=yes
loguserfield=yes
accountlogs=yes
I created the table in the DB so:
CREATE TABLE IF NOT EXISTS `cdr` (
`id` int(11) unsigned NOT NULL AUTO_INCREMENT,
`calldate` datetime NOT NULL DEFAULT '-00-00 00:00:00',
`clid` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`src` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`dst` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`dcontext` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`lastapp` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
`lastdata` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
`duration` float unsigned DEFAULT NULL,
`billsec` float unsigned DEFAULT NULL,
`disposition` enum('ANSWERED','BUSY','FAILED','NO ANSWER','CONGESTION')
COLLATE utf8_bin DEFAULT NULL, `channel` varchar(50) COLLATE utf8_bin DEFAULT
NULL, `dstchannel` varchar(50) COLLATE utf8_bin DEFAULT NULL,
`amaflags` varchar(50) COLLATE utf8_bin DEFAULT NULL,
`accountcode` varchar(20) COLLATE utf8_bin DEFAULT NULL,
`uniqueid` varchar(32) COLLATE utf8_bin NOT NULL DEFAULT '',
`userfield` float unsigned DEFAULT NULL,
`answer` datetime NOT NULL,
`end` datetime NOT NULL,
PRIMARY KEY (`id`),
KEY `calldate` (`calldate`),
KEY `dst` (`dst`),
KEY `src` (`src`),
KEY `dcontext` (`dcontext`),
KEY `clid` (`clid`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8 COLLATE=utf8_bin AUTO_INCREMENT=1 ;
Then I restarted Asterisk (core restart now).
Unfortunately it does not work, since I get on boot:
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1468 load_mysql_config:
MySQL RealTime: No database user found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1474 load_mysql_config:
MySQL RealTime: No database password found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1480 load_mysql_config:
MySQL RealTime: No database host found, using localhost via socket.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1486 load_mysql_config:
MySQL RealTime: No database name found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1492 load_mysql_config:
MySQL RealTime: No database port found, using 3306 as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1509 load_mysql_config:
MySQL RealTime: No database socket found (and unable to detect a suitable path).
And of course:
OpenWrt*CLI cdr show status
Call Detail Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: Yes
* Registered Backends
---
cdr-custom
Asterisk 1.8 runs on an OpenWRT-Switch.
Any idea?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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Re: [asterisk-users] DTMF issue
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees Sent: Monday, July 06, 2015 5:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF issue Hello folks, We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance. I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility, but this hasn't helped at all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals do not clearly state which setting this is on the handsets. I have enabled DTMF logging and spoken to the SIP provider, but they couldn't really help much. I presume the issue is local to our phone system but other than the logs below, have nothing to go on: [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2' received on SIP/sip-out-00021c6d [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin passthrough '2' on SIP/sip-out-00021c6d [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2' received on SIP/sip-out-00021c6d, duration 200 ms [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted with begin '2' on SIP/sip-out-00021c6d [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end passthrough '2' on SIP/sip-out-00021c6d [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3' received on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin passthrough '3' on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' received on SIP/209-00021cac, duration 90 ms [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end accepted with begin '3' on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' detected to have actual duration 78 on the wire, emulation will be triggered on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' has duration 78 but want minimum 80, emulating on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end emulation of '3' queued on SIP/209-00021cac Can someone please provide any tips? Thanks, Jamie This doesn't help, but It DOES sound familiar. I've not seen this for a long time. If I can remember I'll write back. Just thought I'd let you know you're not crazy. :) -- _ -- 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 pin code for out-going international calls (safeguard against fraud)
The Authenticate application will do this for you. https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate You can either give it a single PIN to use for all calls, Authenticate using a value in the Asterisk Database, Or use a plain text file for the PIN's On Mon, Jul 6, 2015 at 2:43 PM, Motty Cruz motty.c...@gmail.com wrote: Hello All, I will like to configure Asterisk to use PIN Code for all outgoing international calls. Also, any suggestions as to when should I prompt users for code prior to dialing the number or after dialing the number? can someone provide with a example on how to accomplish this goal? I am a bit confuse by this : http://forums.digium.com/viewtopic.php?p=130936sid=707f657f7a61dfed55e4922304925091 Thanks for your help. -- _ -- 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 -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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 pin code for out-going international calls (safeguard against fraud)
I've seen this before. It can be done by calling an AGI script when placing the outgoing call. You'd then prompt and make sure the code matches and do your billing logic, etc there. Then place the call if it's valid. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motty Cruz Sent: Monday, July 06, 2015 5:44 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud) Hello All, I will like to configure Asterisk to use PIN Code for all outgoing international calls. Also, any suggestions as to when should I prompt users for code prior to dialing the number or after dialing the number? can someone provide with a example on how to accomplish this goal? I am a bit confuse by this : http://forums.digium.com/viewtopic.php?p=130936sid=707f657f7a61dfed55e4922304925091 Thanks for your help. -- _ -- 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] DTMF issue
Hello folks, We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance. I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility, but this hasn't helped at all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals do not clearly state which setting this is on the handsets. I have enabled DTMF logging and spoken to the SIP provider, but they couldn't really help much. I presume the issue is local to our phone system but other than the logs below, have nothing to go on: [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2' received on SIP/sip-out-00021c6d [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin passthrough '2' on SIP/sip-out-00021c6d [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2' received on SIP/sip-out-00021c6d, duration 200 ms [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted with begin '2' on SIP/sip-out-00021c6d [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end passthrough '2' on SIP/sip-out-00021c6d [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3' received on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin passthrough '3' on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' received on SIP/209-00021cac, duration 90 ms [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end accepted with begin '3' on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' detected to have actual duration 78 on the wire, emulation will be triggered on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' has duration 78 but want minimum 80, emulating on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end emulation of '3' queued on SIP/209-00021cac Can someone please provide any tips? Thanks, Jamie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)
Hello All, I will like to configure Asterisk to use PIN Code for all outgoing international calls. Also, any suggestions as to when should I prompt users for code prior to dialing the number or after dialing the number? can someone provide with a example on how to accomplish this goal? I am a bit confuse by this : http://forums.digium.com/viewtopic.php?p=130936sid=707f657f7a61dfed55e4922304925091 Thanks for your help. -- _ -- 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 in an MySQL-Database
Luca Bertoncello wrote:
Hi list!
I'd like to save all information about calls (CDR) in a MySQL-Database.
I created the DB and a user for Asterisk on a separate server, then I
configured my cdr_mysql.conf so:
[global]
hostname=192.168.10.3
dbname=asterisk
table=cdr
password=MYSECRET
user=asterisk
port=3306
and my cdr.conf so:
[general]
enable=yes
unanswered = yes
safeshutdown=yes
[mysql]
usegmtime=no
loguniqueid=yes
loguserfield=yes
accountlogs=yes
I created the table in the DB so:
CREATE TABLE IF NOT EXISTS `cdr` (
`id` int(11) unsigned NOT NULL AUTO_INCREMENT,
`calldate` datetime NOT NULL DEFAULT '-00-00 00:00:00',
`clid` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`src` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`dst` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`dcontext` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`lastapp` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
`lastdata` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
`duration` float unsigned DEFAULT NULL,
`billsec` float unsigned DEFAULT NULL,
`disposition` enum('ANSWERED','BUSY','FAILED','NO ANSWER','CONGESTION')
COLLATE utf8_bin DEFAULT NULL, `channel` varchar(50) COLLATE utf8_bin DEFAULT
NULL, `dstchannel` varchar(50) COLLATE utf8_bin DEFAULT NULL,
`amaflags` varchar(50) COLLATE utf8_bin DEFAULT NULL,
`accountcode` varchar(20) COLLATE utf8_bin DEFAULT NULL,
`uniqueid` varchar(32) COLLATE utf8_bin NOT NULL DEFAULT '',
`userfield` float unsigned DEFAULT NULL,
`answer` datetime NOT NULL,
`end` datetime NOT NULL,
PRIMARY KEY (`id`),
KEY `calldate` (`calldate`),
KEY `dst` (`dst`),
KEY `src` (`src`),
KEY `dcontext` (`dcontext`),
KEY `clid` (`clid`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8 COLLATE=utf8_bin AUTO_INCREMENT=1 ;
Then I restarted Asterisk (core restart now).
Unfortunately it does not work, since I get on boot:
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1468 load_mysql_config:
MySQL RealTime: No database user found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1474 load_mysql_config:
MySQL RealTime: No database password found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1480 load_mysql_config:
MySQL RealTime: No database host found, using localhost via socket.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1486 load_mysql_config:
MySQL RealTime: No database name found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1492 load_mysql_config:
MySQL RealTime: No database port found, using 3306 as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1509 load_mysql_config:
MySQL RealTime: No database socket found (and unable to detect a suitable path).
And of course:
OpenWrt*CLI cdr show status
Call Detail Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: Yes
* Registered Backends
---
cdr-custom
Asterisk 1.8 runs on an OpenWRT-Switch.
Any idea?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
Been doing this with MySql for the last 10 years, though not on an openWrt
machine
MySql is on the Asterisk machine.
Also have additional database tables to block by callerId and name
Did have some issues with the dialplan syntax when moving from 1.4 to 11, but
it just works
I assume OpenWRT is a pre compiled Asterisk package?
You may not have the proper configuration to use MySql
Your error message(s) seem to say it expects to find the MySql server on
localhost but you say it is on a different machine!!
perhaps you need to fix that first?
John Novack
--
Dog is my Co-pilot
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] DTMF issue
On 7/6/15 5:53 PM, Jamie Rees wrote: Hello folks, We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance. I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility, but this hasn't helped at all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals do not clearly state which setting this is on the handsets. I have enabled DTMF logging and spoken to the SIP provider, but they couldn't really help much. I presume the issue is local to our phone system but other than the logs below, have nothing to go on: [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2' received on SIP/sip-out-00021c6d [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin passthrough '2' on SIP/sip-out-00021c6d [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2' received on SIP/sip-out-00021c6d, duration 200 ms [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted with begin '2' on SIP/sip-out-00021c6d [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end passthrough '2' on SIP/sip-out-00021c6d [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3' received on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin passthrough '3' on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' received on SIP/209-00021cac, duration 90 ms [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end accepted with begin '3' on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' detected to have actual duration 78 on the wire, emulation will be triggered on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' has duration 78 but want minimum 80, emulating on SIP/209-00021cac [2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end emulation of '3' queued on SIP/209-00021cac Can someone please provide any tips? Yes, I have had this annoyance happen to me before. It is very frustrating. In order to rule out the SIP Provider, I suggest you record the call. If the beep is not heard in the recording but only by the end user on the Cisco Phone, then its a phone issue. The phone is confusing audio with the specific frequencies of DTMF. There is little you can do to fix this except for firmware upgrades (and I remember there were some that addressed this specific issue, at least on Cisco ATAs). Thanks, Jamie -- Technical Support http://www.cellroute.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to enable IM over the asterisk server
I am currently, I create the VOIP server which enable the user to make the call over the asterisk server, Additionally now I want the user to be able to chat to each other too. I found some suggestion of using the openfire with asterisk but not much said on it, Anyway could you please share me how can I config the IM server over asterisk? I am waiting for your reply, Thyda -- _ -- 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 in an MySQL-Database
John Novack jnov...@stromberg-carlson.org schrieb: Been doing this with MySql for the last 10 years, though not on an openWrt machine MySql is on the Asterisk machine. Also have additional database tables to block by callerId and name Did have some issues with the dialplan syntax when moving from 1.4 to 11, but it just works I assume OpenWRT is a pre compiled Asterisk package? You may not have the proper configuration to use MySql Your error message(s) seem to say it expects to find the MySql server on localhost but you say it is on a different machine!! perhaps you need to fix that first? I think, the team of OpenWRT did NOT prepare the CDR-MySQL-Module, since I could not find cdr_addon_mysql.so... I resolved writing the data in a CSV, and then importing the data in the MySQL-DB with a script... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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 call quality: Asterisk call quality on trunks
Good afteroon all, First of all: thanks for everybody who is willing to think this through with me: I'm having some issues regarding call quality between some calls. Let me try to explain the situation first We have a Asterisk 11.16 server based on the Xivo distribution. There are 2 servers running in cluster (Active Passive), both virtual with the following config: Quadcore CPU 8 GB ram About 50Gb of diskspace which is used for about 15% (Let's call this Asterisk cluster 001 for clarity) The Asterisk server has a trunk to a cisco call manager which is on the same site/LAN, and 4 trunks to other Asterisk servers (same distribution but lower specs, name Asterisk cluster 002 and 003). These are all sites in our WAN but they are geographically divided and connected via MPLS links. Each affiliate has a specific number range XXXYYY where XXX stands for the affiliate and YYY is the extension of the users. (Average bandwidth = 4Mpbs which has to be shared by applications. QoS allows that VoIP is prioritized) Now, the actual problem: I've set my main codecs to G711 a-law, G7 222 (for cisco call manager) and GSM as last. The GSM is set as primary for those trunks which don't have 4 Mbps of bandwidth available. In most cases, trunk calling results in bad quality of conversations (a-law is chosen as codec) but or it is jitterish, or one party does not hear the other party (complete silence) It could be that the second time they call, everything is ok. -- So a little ASCII map about the geographical setup: Aff 1: [Asterisk cluster 001] -- LAN trunk -- Cisco call manager | MPLS connection 20Mbps | |-- MPLS Cloud--- MPLS connection 2Mbps -- [Asterisk cluster 002] | MPLS connection 4 Mbps -- [Asterisk cluster 003] Calls between Cluster 001 --- cluster 002 or 003 are potentially of bad quality (sometimes ok but most of all jiterish) Calls between Cluster 002 --- cluster 003 are good The bandwidth if cluster001 ( 20 Mbps) is used about 50% with peaks to 75%. I've aslo actived the jitter buffer with a buffer of 200ms but this didn't seem to do any good. Does anybody have some hints how I can troubleshoot this? Note: the Cisco calls to the other affiliaters over the same WAN don't have issues but these are based on SCCP protocol. Thanks in advance Kristof Privileged Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of my firm shall be understood as neither given nor endorsed by it. -- This message has been scanned by Cisa Antispam Service and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk how to setup alarm too many outgoing calls from same user
Hello, I would like to setup a mechanism to trigger an alarm if user is deal too many numbers within a very short period of time. Safeguard against users hacked accounts. can someone help? 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
[asterisk-users] Asterisk 13.4.0 - mixmonitor only records one side's perspective
Hi All
I have a problem with mixmonitor in 13.4.0 doing the following:
1. Caller phones in
2. Reception picks up
3. Talks to caller
4. Does attended transfer, talks to manager to screen the caller wanting to
speak to him
5. Complete the transfer by putting down her handset so the caller can
speak to the manager
6. Caller talks to the manager
The problem is that mixmonitor only records steps (3) and (6) - for step (4)
in the recording file, you hear the MOH the outside caller hears while the
receptionist is screeing the call.
We want to record the -entire- conversation, including the receptionist
screening with the manager.
E. g. you literally only hear the perspective of the originator of the call
- you never hear the receptionist - manager leg of the call if a transfer
occurs during a call.
Here's how an incoming call is handled - on 1.8 the below code recorded the
entire conversation flawlessly, including the screening conversation of
reception - manager
[inc]
exten=_[123]xxx,1,Macro(VCRECORD,${MACRO_CONTEXT}EXT${CALLERID(num)}ACC${CD
R(accountcode)},${ARG2})
exten=_[123]xxx,n,Set(__TRANSFER_CONTEXT=call-redirect)
exten=_[123]xxx,n(checkacc),NoOp(MY Account code is ${CDR(accountcode)})
;exten=_[123]xxx,n(checkacc),NoOp(OTHER Account code is
${CDR(accountcode)})
exten=_[123]xxx,n(dodial),Dial(Sip/${EXTEN},120,tTg)
exten=_[123]xxx,n,NoOp(Dialstatus: ${DIALSTATUS})
exten=_[123]xxx,n,GotoIf($[${DIALSTATUS}=NOANSWER]?takevoicemail:checkd
ont)
exten=_[123]xxx,n(checkdont),GotoIf($[${DIALSTATUS}=DONTCALL]?takevoice
mail:donecall)
exten=_[123]xxx,n,NoOp(Taking a voicemail...)
exten=_[123]xxx,n(takevoicemail),VoiceMail(${EXTEN}@default)
exten=_[123]xxx,n(donecall),Hangup()
[call-redirect]
include = parkedcalls
exten=_[123]xxx,1,NoOp(Transferring Call. This Channel ${CHANNEL}, Other
channel ${BLINDTRANSFER})
exten=_[123]xxx,n(dodial),Dial(Sip/${EXTEN},120,tTg)
[macro-VCRECORD] ; MACRO To setup Recording
;${ARG1} Description To Save
;${ARG2} Dialed Number
exten=s,1,NoOp(Start of MixMonitor recording)
exten=s,n,Set(IAXVAR(accountcode)=${CDR(accountcode)})
exten=s,n,GoToIf($[${MIXMONITOR_FILENAME} = ]?startrec:finrec)
exten=s,n(startrec),Set(recDir=${STRFTIME(${EPOCH},,%y%m/%d)})
exten=s,n,Set(recFile=${recDir}/${STRFTIME(${EPOCH},,%y%m%d%H%M%S)}D${ARG1}
N${ARG2}ID${UNIQUEID}.gsm)
exten=s,n(setacc),set(recFile=${recDir}/${CDR(linkedid)}.gsm)
exten=s,n(makedir),System(/bin/mkdir -p
/var/spool/asterisk/monitor/${recDir})
exten=s,n,MixMonitor(${recFile},a)
exten=s,n(finrec),Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten=s,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=s,n,Set(__chanrecording=/var/spool/asterisk/monitor/${recFile})
exten=s,n,NoOp(Recording to ${MIXMONITOR_FILENAME})
exten=s,n,UserEvent(RecordingToFile,Uniqueid: ${UNIQUEID},Channel:
${CHANNEL},FileName: ${MIXMONITOR_FILENAME})
exten=s,n,MacroExit
Can somebody help or offer a suggestion how to get MixMonitor to record an
ENTIRE conversation in 13.4.0, not just record from the perspective of the
initiator of a call?
Thanks
--
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