search for: dsp_poll

...timeout=no method=standard dialplan=0 localdialplan=0 cpndialplan=0 early_bconnect=yes incoming_early_audio=no nodialtone=no presentation=-1 screen=-1 jitterbuffer=4000 jitterbuffer_upper_threshold=0 hdlc=no [isdn] ports=1 context=incoming msns=* misdn-init.conf: card=1,hfcpci te_ptmp=1 poll=128 dsp_poll=128 dsp_options=0 dtmfthreshold=100 debug=0 When I make a call the CLI tells that: *CLI> -- Executing Dial("SIP/101-081990e8", "mISDN/1/943833473|45|tTwW") in new stack Jun 19 12:32:25 WARNING[2153]: channel.c:2618 ast_request: No channel type registered for 'mISDN...

...ink:UP Blocked:0 Debug:0 but I'm not sure how to check our settings are right with Etisalat, as above then goes to: pbx*CLI> misdn show stacks BEGIN STACK_LIST: * Port 1 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 /etc/misdn-init.conf: card=1,hfcpci te_ptmp=1 poll=128 dsp_poll=128 dsp_options=0 dtmfthreshold=100 debug=0 /etc/asterisk/misdn.conf: [general] debug = 0 method=standard bridging=no stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh [default] context=misdn language=en musicclass=default senddtmf=yes f...

...presentation=allowed [default] context=numeris language=fr nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 ;dialplan=0 use_callingpres=yes presentation=allowed senddtmf=yes [numeris] context=numeris-in ports=1,2,3,4 msns=* *** /etc/misdn-init.conf card=1,0x4 te_ptmp=1,2,3,4 poll=128 dsp_poll=128 dsp_options=0 dtmfthreshold=100 debug=0 -- Olivier Fauchon Freelance System-Network Admin Phone: +33 610 493 763 http://www.aixmarseille.com

...ease_complete and the line is "BUSY". The phones on the NT line works fine, and SIP <-> mISDN (NT) calls has no problems. All comment lines has been removed to keep mail size reasonable. /etc/misdn-init.conf: card=1,0x4,dtmf te_ptmp=1,2 nt_ptmp=3,4 option=1,master_clock poll=128 dsp_poll=128 dsp_options=0 dtmfthreshold=100 debug=0 misdn.conf: [general] misdn_init=/etc/misdn-init.conf debug=1 ntdebugflags=0 ntdebugfile=/var/log/misdn-nt.log bridging=yes stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh [default] context=def...

...-- Port 2: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> childcnt: 2 -------- mISDN_close: fid(3) isize(131072) inbuf(0x9eff060) irp(0x9eff060) iend(0x9eff060) [root@gateway ~]# [root@gateway ~]# cat /etc/misdn-init.conf card=1,0x4 te_ptmp=1,2 poll=127 dsp_poll=128 dsp_options=0 dtmfthreshold=100 debug=5 [root@gateway ~]# cat /etc/asterisk/misdn.conf [general] debug = 0 method=standard bridging=no stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh [default] context=misdn language=it musicclass...

...make force does not help at all) 2) In some of our installations I'm getting stutter in the sound stream every two seconds or so (just under). I've tried to track this down to configuration but not been successful in spotting what the problem might be. Should I look for things like poll or dsp_poll values or does anyone have any suggestions that may help in pinning this? Btw, the setup is two B410P's in one machine, four ports in NT mode and four in TE mode. Feeding three ISDN BRI's into the system and three out again: PSTN <-> NT box <-> B410P port in TE mode <-> A...

...network controller [HFC-PCI] (rev 02) this is my sip general part (jb enable to get the jitter buffer working): jbenable = yes jbforce = yes jbmaxsize = 200 jbresyncthreshold = 1000 jbimpl = adaptive jblog = yes This is the relevant part of the misdn-init.conf card=1,hfcpci te_ptp=1,2 poll=128 dsp_poll=128 dsp_options=0 dtmfthreshold=100 debug=0 And this is the misdn.conf [general] misdn_init=/etc/misdn-init.conf debug=0 ntdebugflags=0 ntdebugfile=/var/log/misdn-nt.log ntkeepcalls=no bridging=no l1watcher_timeout=0 stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_pr...

...es presentation=allowed [default] context=numeris language=fr nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 ;dialplan=0 use_callingpres=yes presentation=allowed senddtmf=yes [numeris] context=numeris-in ports=1,2,3,4 msns=* * /etc/misdn-init.conf card=1,0x4 te_ptmp=1,2,3,4 poll=128 dsp_poll=128 dsp_options=0 dtmfthreshold=100 debug=0 obelix:/etc/as * SIP configuration [default] exten => 412,1,Dial(SIP/hard_sysadmins&SIP/ofauchon_soft,30) exten => 412,2,Voicemail(412) [numeris-in] exten => 0412,1,Goto(default,412,1) * Log of an incoming call (notice the activity...