Re: [asterisk-users] Digium G100 and CID Dropping First Digit.
On Mon, Jan 15, 2018 at 10:41:27PM +, David Klaverstyn wrote: > port1 < Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: > ISDN/Telephony Numbering Plan (E.164/E.163) (1) > port1 < Presentation: Presentation allowed of > network provided number (3) '21xx' ] > port1 < [70 0a c1 30 34 39 31 34 31 32 31 34] > port1 < Called Number (len=12) [ Ext: 1 TON: Subscriber Number (4) > NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '049xx' ] > > -- Accepting call from '21xx' to '049xx' on channel 0/5, span > 1 Don't know anything about the card you are using, but seeing ISDN signaling that the type of number (TON) is national and that means overhere that leading zeros are stripped, I see nothing wrong with it. Looking at my old chan_dadhi configs there are options to prefix something based on TON. So over here I have configured: nationalprefix = 0 to prefix the leading 0 for national numbers that callees expect. The G100 manual contains the phrase "national prefix", but no info about it, so look into those prefix options. -- I feel stupid now. I should have figured that one out. On the Digium gateway: Configuration > T1/E1 > Advanced Signalling > PRI Options : International, Nation and Local prefix. Once I entered the correct digits International 00 and National 0, CID worked as expected. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium G100 and CID Dropping First Digit.
On Mon, Jan 15, 2018 at 10:41:27PM +, David Klaverstyn wrote: > port1 < Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: > ISDN/Telephony Numbering Plan (E.164/E.163) (1) > port1 < Presentation: Presentation allowed of > network provided number (3) '21xx' ] > port1 < [70 0a c1 30 34 39 31 34 31 32 31 34] > port1 < Called Number (len=12) [ Ext: 1 TON: Subscriber Number (4) NPI: > ISDN/Telephony Numbering Plan (E.164/E.163) (1) '049xx' ] > > -- Accepting call from '21xx' to '049xx' on channel 0/5, span 1 Don't know anything about the card you are using, but seeing ISDN signaling that the type of number (TON) is national and that means overhere that leading zeros are stripped, I see nothing wrong with it. Looking at my old chan_dadhi configs there are options to prefix something based on TON. So over here I have configured: nationalprefix = 0 to prefix the leading 0 for national numbers that callees expect. The G100 manual contains the phrase "national prefix", but no info about it, so look into those prefix options. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium G100 and CID Dropping First Digit.
Hi All, I have installed a number of Digium G100 devices in many countries like South Korea, Japan, Singapore and Australia. I have just installed two in New Zealand and both sites are having a problem with Caller ID. Incoming calls are dropping the first digit 0 from the caller ID. I was previously using DAHDI and a TE121 device which may have been adding the 0, I'm not too sure about that. Anyhow, is it possible the Digium G100 is causing the problem or would it be the Telco not passing the full CID number? I have the latest firmware on the G100, the same with all my other locations. I am using the latest Asterisk 13.19.0. The Telco is blaming the PBX for the problem so I'm hoping someone here can shed some light. Below is a debug extract which I hope will help. The 21 number should be 021. Jan 16 10:53:16 G100-59-c4-ca asterisk[4674]: VERBOSE[4786]: chan_gtw/chan.c:4803 in dgm_pri_message: port1 < Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Jan 16 10:53:16 G100-59-c4-ca asterisk[4674]: VERBOSE[4786]: chan_gtw/chan.c:4803 in dgm_pri_message: port1 < Presentation: Presentation allowed of network provided number (3) '21xx' ] Jan 16 10:53:16 G100-59-c4-ca asterisk[4674]: VERBOSE[4786]: chan_gtw/chan.c:4803 in dgm_pri_message: port1 < [70 0a c1 30 34 39 31 34 31 32 31 34] Jan 16 10:53:16 G100-59-c4-ca asterisk[4674]: VERBOSE[4786]: chan_gtw/chan.c:4803 in dgm_pri_message: port1 < Called Number (len=12) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '049xx' ] Jan 16 10:53:16 G100-59-c4-ca asterisk[4674]: VERBOSE[4786]: chan_gtw/sig_pri_new.c:5383 in pri_dchannel: -- Accepting call from '21xx' to '049xx' on channel 0/5, span 1 Thanks David. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Duplicate CDR's in Mysql
On 1/14/18 4:22 PM, Mike Diehl wrote: Hi all, I have a problem I've not seen before. My Asterisk server stores CDR's via mysql, and I'm getting duplicate records. For example: mysql> select uniqueid,count(*) from cdr group by uniqueid having count(*)>1; +--+--+ | uniqueid | count(*) | +--+--+ | server12-1515090905.2182 | 5 | | server12-1515091190.2215 | 3 | +--+--+ 2 rows in set (0.68 sec) If I query for each uniqueid, I see that the records are identical. I have a Perl script that goes through and removes the duplicates. Otherwise, EVERY CDR would be duplicated. Now, my Asterisk server was configured with multiple CDR backends, but I unloaded those modules. Here is what I have configured during run-time: *CLI> cdr show status Call Detail Record (CDR) settings -- Logging: Enabled Mode: Simple Log unanswered calls: No Log congestion: No * Registered Backends --- cdr-custom Adaptive ODBC I have exactly the same problem. All CDRs get duplicated. Using ODBC and Mariadb on Asterisk 13.8 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] email when certain numbers are called
On Mon, 2018-01-15 at 14:26 +0200, Atux Atux wrote: > [DefaultPlan] exten => _XX,1,System(echo "Dialed number ${EXTEN} on Asterisk from ${CALLERID(num)}" | mail -s "Dialed number ${EXTEN} on Asterisk from ${CALLERID(num)}" -a "From: Asterisk PBX " yo urem...@address.com) exten => _XX,2,Dial(SIP/VoipGate/${EXTEN},120,Tt) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] email when certain numbers are called
Hi. I have an installation of asterisk 11 and i have ssmtp in the system to send emails. I would like to get informed by email when someone dials a set of numbers eg international calls or premium numbers with the country. my dialplan is simple enough and it is the following: [DefaultPlan]exten => _XX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten => _XX,1,Busy() exten => _4XX,2,Answer()exten => _4XX,3,VoiceMail(${EXTEN}@BranchA,su)exten => _4XX,4,HangUp()exten => _4XX,102,Answer()exten => _4XX,103,VoiceMail(${EXTEN}@BranchA,sb)exten => _4XX,104,HangUp() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users