Martin Kronstad
2005-Aug-02 01:48 UTC
[Asterisk-Users] Config HFC-card in asterisk.(Config the phone and asterisk)
Hi! I am trying to get my ISDN phone to work with my asterisk box. Now my asterisk won't start. Current situation: I have a cable from my Billion ISDN (Bipac V1.0) to my old NT1. The cable is crossed like this: 1 2 3 -> 4 4 -> 3 5 -> 6 6 -> 5 7 8 Then I have a cable from the NT1 to the ISDNphone(not crossed cable). Both cables are connected in the ISDN ports of the NT1 (no cables in the Line port of the NT1). There is no dialtone in my phone, and when I lift the headset off the phone I get, after a few seconds a message of "No line" in the display. The ISDN card should be correctly set up in NT-mode. Result from [root@asterisk1 root]# ztcfg -vv ------------------ START RESULT ---------------------- SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ------------------ END RESULT ---------------------- Result from running ztool : Only one item on the list : OK HFC-S PCI A ISDN card 0 [NT] layer 1 AC Here are the last lines of my log (disregard the date and time, date and time on my server is wrong) ----------------- START LOG ------------------------ Aug 2 04:03:46 VERBOSE[1552]: [chan_phone.so]Aug 2 04:03:46 VERBOSE[1552]: [chan_phone.so] => (Linux Telephony API Support) Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/phone.conf': Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/phone.conf': Found Aug 2 04:03:46 VERBOSE[1552]: == Registered channel type 'Phone' (Standard Linux Telephony API Driver) Aug 2 04:03:46 VERBOSE[1552]: [chan_zap.so]Aug 2 04:03:46 DEBUG[1552]: Setting NAT on RTP to 0 Aug 2 04:03:46 DEBUG[1552]: Stopping retransmission on '4054139b6d809de158552f97216c60f0@10.0.0.10' of Request 102: Found Aug 2 04:03:46 VERBOSE[1552]: [chan_zap.so] => (Zapata Telephony w/PRI) Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/zapata.conf': Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/zapata.conf': Found Aug 2 04:03:46 WARNING[1552]: No '=' (equal sign) in line 5 of zapata.conf Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/zapata_additional.conf': Aug 2 04:03:46 VERBOSE[1552]: =Parsing '/etc/asterisk/zapata_additional.conf': Found Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/zapata-auto.conf': Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/zapata-auto.conf': Found Aug 2 04:03:46 DEBUG[1552]: Updated conferencing on 1, with 0 conference users Aug 2 04:03:46 VERBOSE[1552]: -- Registered channel 1, PRI Signalling signalling Aug 2 04:03:46 DEBUG[1552]: Updated conferencing on 2, with 0 conference users Aug 2 04:03:46 VERBOSE[1552]: -- Registered channel 2, PRI Signalling signalling Aug 2 04:03:46 WARNING[1552]: Ignoring record_out Aug 2 04:03:46 WARNING[1552]: Ignoring record_in Aug 2 04:03:46 WARNING[1552]: Ignoring echocancelwhenbridge Aug 2 04:03:46 ERROR[1552]: Signalling requested is FXO Kewlstart but line is in PRI Signalling signalling Aug 2 04:03:46 ERROR[1552]: Unable to register channel '1' Aug 2 04:03:46 WARNING[1552]: chan_zap.so: load_module failed, returning -1 Aug 2 04:03:46 VERBOSE[1552]: == Unregistered channel type 'Tor' Aug 2 04:03:46 VERBOSE[1552]: == Unregistered channel type 'Zap' Aug 2 04:03:46 VERBOSE[1552]: -- Unregistered channel 1 Aug 2 04:03:46 VERBOSE[1552]: -- Unregistered channel 2 Aug 2 04:03:46 WARNING[1552]: Loading module chan_zap.so failed! Aug 2 04:03:46 WARNING[1552]: Loading module chan_zap.so failed! --------------------- END LOG ---------------------------------- Martin Kronstad Siteman DA www.siteman.no Tlf:. 32 87 56 10 Mobil: 951 70 230 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/efa9a23d/attachment.htm
Zoa
2005-Aug-02 03:30 UTC
[Asterisk-Users] Config HFC-card in asterisk.(Config the phone and asterisk)
Aug 2 04:03:46 ERROR[1552]: Signalling requested is FXO Kewlstart but line is in PRI Signalling signalling This is your problem, probably in one configuration file you have fxo kewlstart as signalling, and in the other one you have pri signalling. Greetz, Zoa --- http://www.asteriskguru.com Martin Kronstad wrote:> Hi! > > I am trying to get my ISDN phone to work with my asterisk box. > > Now my asterisk won?t start? > > Current situation: > > I have a cable from my Billion ISDN (Bipac V1.0) to my old NT1. > > The cable is crossed like this: > > 1 > > 2 > > 3 -> 4 > > 4 -> 3 > > 5 -> 6 > > 6 -> 5 > > 7 > > 8 > > Then I have a cable from the NT1 to the ISDNphone(not crossed cable). > Both cables are connected in the ISDN ports of the NT1 (no cables in > the Line port of the NT1). > > There is no dialtone in my phone, and when I lift the headset off the > phone I get, after a few seconds a message of ?No line? in the display. > > The ISDN card should be correctly set up in NT-mode. > > Result from [root@asterisk1 root]# ztcfg ?vv > > ------------------ START RESULT ---------------------- > > SPAN 1: CCS/ AMI Build-out: 399- 533 feet (DSX-1) > > Channel map: > > Channel 01: Individual Clear channel (Default) (Slaves: 01) > > Channel 02: Individual Clear channel (Default) (Slaves: 02) > > Channel 03: D-channel (Default) (Slaves: 03) > > 3 channels configured. > > ------------------ END RESULT ---------------------- > > Result from running ztool : Only one item on the list : OK HFC-S PCI A > ISDN card 0 [NT] layer 1 AC > > Here are the last lines of my log (disregard the date and time, date > and time on my server is wrong) > > ----------------- START LOG ------------------------ > > Aug 2 04:03:46 VERBOSE[1552]: [chan_phone.so]Aug 2 04:03:46 > VERBOSE[1552]: [chan_phone.so] => (Linux Telephony API Support) > Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/phone.conf': > Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/phone.conf': Found > Aug 2 04:03:46 VERBOSE[1552]: == Registered channel type 'Phone' > (Standard Linux Telephony API Driver) > Aug 2 04:03:46 VERBOSE[1552]: [chan_zap.so]Aug 2 04:03:46 DEBUG[1552]: > Setting NAT on RTP to 0 > Aug 2 04:03:46 DEBUG[1552]: Stopping retransmission on > '4054139b6d809de158552f97216c60f0@10.0.0.10' of Request 102: Found > Aug 2 04:03:46 VERBOSE[1552]: [chan_zap.so] => (Zapata Telephony w/PRI) > Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/zapata.conf': > Aug 2 04:03:46 VERBOSE[1552]: == Parsing '/etc/asterisk/zapata.conf': > Found > Aug 2 04:03:46 WARNING[1552]: No '=' (equal sign) in line 5 of zapata.conf > Aug 2 04:03:46 VERBOSE[1552]: == Parsing > '/etc/asterisk/zapata_additional.conf': Aug 2 04:03:46 VERBOSE[1552]: > == Parsing '/etc/asterisk/zapata_additional.conf': Found > Aug 2 04:03:46 VERBOSE[1552]: == Parsing > '/etc/asterisk/zapata-auto.conf': Aug 2 04:03:46 VERBOSE[1552]: => Parsing '/etc/asterisk/zapata-auto.conf': Found > Aug 2 04:03:46 DEBUG[1552]: Updated conferencing on 1, with 0 > conference users > Aug 2 04:03:46 VERBOSE[1552]: -- Registered channel 1, PRI Signalling > signalling > Aug 2 04:03:46 DEBUG[1552]: Updated conferencing on 2, with 0 > conference users > Aug 2 04:03:46 VERBOSE[1552]: -- Registered channel 2, PRI Signalling > signalling > Aug 2 04:03:46 WARNING[1552]: Ignoring record_out > Aug 2 04:03:46 WARNING[1552]: Ignoring record_in > Aug 2 04:03:46 WARNING[1552]: Ignoring echocancelwhenbridge > Aug 2 04:03:46 ERROR[1552]: Signalling requested is FXO Kewlstart but > line is in PRI Signalling signalling > Aug 2 04:03:46 ERROR[1552]: Unable to register channel '1' > Aug 2 04:03:46 WARNING[1552]: chan_zap.so: load_module failed, > returning -1 > Aug 2 04:03:46 VERBOSE[1552]: == Unregistered channel type 'Tor' > Aug 2 04:03:46 VERBOSE[1552]: == Unregistered channel type 'Zap' > Aug 2 04:03:46 VERBOSE[1552]: -- Unregistered channel 1 > Aug 2 04:03:46 VERBOSE[1552]: -- Unregistered channel 2 > Aug 2 04:03:46 WARNING[1552]: Loading module chan_zap.so failed! > Aug 2 04:03:46 WARNING[1552]: Loading module chan_zap.so failed! > > --------------------- END LOG ---------------------------------- > > Martin Kronstad > > Siteman DA > > www.siteman.no > > Tlf:. 32 87 56 10 > > Mobil: 951 70 230 > >------------------------------------------------------------------------ > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 254 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/b2b69495/signature.pgp