search for: te_ptmp

.... PMP L2Link UP L1Link: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=de...

...ed_more_infos=no nttimeout=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 regis...

...ld=0 use_callingpres=yes 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

...16:41:18 2007: P[ 1] --> we have already send Release_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 c...

...PBX running?) -------- 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...

...------- Port 4: NT-mode BRI S/T interface port (for phones) -> Layer 2 protocol 0x02000002 is detected, but not allowed for NT lib. * Port NOT useable for PBX (maybe there is already a PBX running?) -------- I do: etc/misdn-init.conf card=1,hfcpci card=2,hfcpci card=3,hfcpci card=4,hfcpci te_ptmp=1,2 nt_ptp=3,4 --- etc/misdn.conf <mISDNconf> <module poll="128" debug="0" timer="no">hfcmulti</module> <module debug="0" options="0">mISDN_dsp</module> <devnode user="asterisk" group="a...

...; # transparent mode > # > # > card=1,0x4, dtmf > > # > # Port settings > # > # Syntax: <port_type>=<port_number>[,<port_number>...] > # > # <port_type> te_ptp - TE-Mode, PTP > # te_ptmp - TE-Mode, PTMP > # te_capi_ptp - TE-Mode (capi), PTP > # te_capi_ptmp - TE-Mode (capi), PTMP > # nt_ptp - NT-Mode, PTP > # nt_ptmp - NT-Mode, PTMP > # <p...

...shold=0 use_callingpres=yes 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...