[asterisk-users] DTMF detection problem with analog card
Hi all. I have a DTMF detection problem by my new analog card (ATCOM 2 FXO port). When i`m playing a voice with 'GET DATA' AGI command, sometimes asterisk do not receive DTMF from caller while the voice is playing. But if user waits to the end of playing voice, there is no problem. I`m using Asterisk 10.3.1, dahdi-2.6.1 on CentOS.6.4. Could you please help me? Here is my configs: system.conf: fxsks=1 fxsks=2 loadzone = nl defaultzone = nl chan_dahdi.conf: -- [channels] ;=== ;General options ;=== usecallerid = yes hidecallerid = no busydetect=yes busycount=3 ;=== ;FXO Modules ;=== group = 1 signalling = fxs_ks context = my-context channel = 1,2-- _ -- 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] ERROR: Unknown signalling method ss7
libss7-trunk cannot be used with any released version of Asterisk.
Thank you Richard.
I could install libss7-branche 1.0 successfully.
But i faced another problem. When i`m dialling over my SS7 enabled E1,
caller-id did not presentaded correctly on the destination phone.
I`m using these two method to change screeningPresentation parameters of IAM
message:
1.
exten = _X.,1,Dial(DAHDI/g1/${EXTEN}, 10, u(allowed)f(75462541))
2.
exten = _X.,1,Set(CALLERID(num)=75462541)
exten = _X.,n,Set(CALLERID(pres)=allowed)
;exten = _X.,n,Set(CALLERID(num-pres)=allowed)
exten = _X.,n,Dial(DAHDI/g1/${EXTEN})
With both method, in the outgoing IAM message from asterisk to Telecom,
PresentationScreening are as follow but caller-id on the destination phone is
''.
[1] Calling Party Number:
[1] Nature of address: 3
[1] NI: 0
[1] Numbering plan: 1
[1] Presentation: 0
[1] Screening: 3
[1] Address signals: 75462541
Can you please guide me?
Best Regards.
--
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[asterisk-users] ERROR: Unknown signalling method ss7
Hi all I installed DAHDI Version - 2.6.1 DAHDI Tools Version - 2.6.1 libss7-trunk Asterisk 11.0.1 from source on Fedora 12 x86_64. Now i`m unable to load chan_dahdi and libss7: myserver*CLI module load chan_dahdi.so ERROR[10124]: chan_dahdi.c:17842 process_dahdi: Unknown signalling method 'ss7' at line 37. myserver*CLI module load libss7.so Unable to load module libss7.so Command 'module load libss7.so' failed. [Mar 14 22:30:05] WARNING[10124]: loader.c:423 load_dynamic_module: Module 'libss7.so' did not register itself during load [Mar 14 22:30:05] WARNING[10124]: loader.c:878 load_resource: Module 'libss7.so' could not be loaded. what is the problem? Can you please help me to solve this problem? Here is my config files: system.conf: = span=1,1,0,ccs,hdb3 bchan=1-15,17-31 mtp2=16 #dchan=16 loadzone = us defaultzone = us == chan_dahdi.conf: === [trunkgroups] [channels] callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes ;General options usecallerid = yes hidecallerid = no callwaiting = yes threewaycalling = yes transfer = yes echocancel = yes echocancelwhenbridged = yes rxgain = 0.0 txgain = 0.0 switchtype = national group = 1 signalling = ss7 ss7type = itu linkset = 1 ss7type = itu linkset = 1 pointcode = adjpointcode = defaultdpc = cicbeginswith = 1 channel = 1-15 cicbeginswith = 17 channel = 17-31 sigchan = 16 == Best Regards.-- _ -- 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
