[Asterisk-Users] Zap failed
I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Asterisk 1.0.9, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <[EMAIL PROTECTED]> [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Sep 16 20:36:51 ERROR[6750]: chan_zap.c:6246 mkintf: Unable to get parameters Sep 16 20:36:51 ERROR[6750]: chan_zap.c:9191 setup_zap: Unable to register channel '1-15' Sep 16 20:36:51 WARNING[6750]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Sep 16 20:36:51 WARNING[6750]: loader.c:440 load_modules: Loading module chan_zap.so failed! sip:/etc # Ouch ... error while writing audio data: : Broken pipe Here is the /proc/zaptel/1 file(seems to be correct, and log Messages too indicates the initialization is correct.): cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard E100P E1/PRA Card 0" HDB3/CCS ClockSource 1 WCT1/0/1 Clear 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear 9 WCT1/0/9 Clear 10 WCT1/0/10 Clear 11 WCT1/0/11 Clear 12 WCT1/0/12 Clear 13 WCT1/0/13 Clear 14 WCT1/0/14 Clear 15 WCT1/0/15 Clear 16 WCT1/0/16 HDLCFCS 17 WCT1/0/17 Clear 18 WCT1/0/18 Clear 19 WCT1/0/19 Clear 20 WCT1/0/20 Clear 21 WCT1/0/21 Clear 22 WCT1/0/22 Clear 23 WCT1/0/23 Clear 24 WCT1/0/24 Clear 25 WCT1/0/25 Clear 26 WCT1/0/26 Clear 27 WCT1/0/27 Clear 28 WCT1/0/28 Clear 29 WCT1/0/29 Clear 30 WCT1/0/30 Clear 31 WCT1/0/31 Clear Ztcfg -vv Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. /etc/zaptel file: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=us defaultzone=us /etc/asterisk/zapata.conf [channels] switchtype = euroisdn signalling = pri_net pridialplan = local language=en context=ivr-in overlapdial=yes usecallerid=yes hidecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default group=1 channel => 1-15 channel => 17-31 smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Asterisk and Zyxel Prestige 2000W_v2
Here is conf example [51] type=friend username=Test secret=testpassword host=dynamic canreinvite=no context=sip disallow=all allow=alaw dtmfmode=rfc2833 And you have to make phone conf. Like this Username = 51 Password = testpassword Phone number 51 [EMAIL PROTECTED] Iyi Calismalar. Ugur GUNCER System Administrator TeleBizz Telecommunication, Billing, Internet & Satellite Solutions Akinci Bayiri Sokak No:13 Cevre Apt. B Blok Kat:3 Daire:5 Mecidiyekoy-Sisli/ISTANBUL Mobile:+90 544 535 97 37 Tel :+90 212 347 69 59 Fax :+90 212 347 69 49 Personal E-mail: [EMAIL PROTECTED] General E-mail: [EMAIL PROTECTED] Internet URL: http://www.telebizz.com.tr The information contained in this message is confidential and is intended for the addressee only. Any unauthorized dissemination or copying or use or disclosure of information contained herein is strictly prohibited and may be illegal. If you are not the named or intended recipient please notify us immediately by telephone, fax or return e-mail. TeleBizz has installed active virus software but does not accept liability or responsibility for the security or reliability of transmission or for any virus transmitted. The contents of this e-mail and any attachments are not intended as and do not constitute advice and TeleBizz disclaims any liability or responsibility for the accuracy thereof. Bu e-posta ve onunla iletilen bütün dosyalar sadece göndericisi tarafından alması amaçlanan yetkili gerçek ya da tüzel kişinin kullanımı içindir. Eğer söz konusu yetkili alıcı değilseniz bu elektronik postanın içeriğini açıklamanız, kopyalamanız, yönlendirmeniz ve kullanmanız kesinlikle yasaktır ve bu elektronik postayı derhal silmeniz gerekmektedir. TeleBizz bu mesajın içerdiği bilgilerin doğruluğu veya eksiksiz olduğu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından ve saklanmasından sorumlu değildir. Bu mesajdaki görüşler yalnızca gönderen kişiye aittir ve TeleBizz'in görüşlerini yansıtmayabilir. Bu e-posta bilinen bütün bilgisayar virüslerine karşı taranmıştır. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Klaus Sonnenleiter > Sent: Friday, September 16, 2005 5:01 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Asterisk and Zyxel Prestige 2000W_v2 > > Hi > > Has anybody tried to use the Zyxel Prestige 2000W_v2 with > Asterisk? I have the latest firmware and can now make > outbound calls. So it looks like Asterisk does accept the > configuration in sip.conf. However, I cannot receive any > calls. Also, I keep getting this message on the > console: > > chan_sip.c:7733 handle_request: Registration from > '' failed for '10.99.1.151'. > > TIA > > Klaus > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 BEGIN:VCARD VERSION:2.1 N:GUNCER;Ugur FN:Ugur GUNCER ([EMAIL PROTECTED]) ORG:Telebizz Telekomunikasyon ve Internet Hizmetleri Ltd. Sti.;IT TITLE:Sistem Yoneticisi TEL;WORK;VOICE:+90 (212) 347 69 59 TEL;CELL;VOICE:+90 (544) 535 97 37 TEL;WORK;FAX:+90 (212) 347 69 49 ADR;WORK;ENCODING=QUOTED-PRINTABLE:;Telebizz Istanbul;Akinci Bayiri=0D=0ASokak No:13 Cevre Apt. =0D=0AB Blok Da= ire: 5;Mecidiyekoy-Sisli, Istanbul;;80290;T=FCrkiye LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Telebizz Istanbul=0D=0AAkinci Bayiri=0D=0ASokak No:13 Cevre Apt. =0D=0AB Blo= k Daire: 5=0D=0AMecidiyekoy-Sisli, Istanbul 80290=0D=0AT=FCrkiye EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20050606T102426Z END:VCARD smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Grandstream HandyTone 386
Hello all I have a question about Grandstream HandyTone 386 can Grandstream HandyTone 386 make 2 sim. calls with g729 codec in same time Iyi Calismalar. Ugur GUNCER smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Conferance DialPlan
I'd like to make a dial plan but couldn't work it out. I'd be appreciated if
you can help me.
The client reaches asterisk by PRI and starts conferance by the SIP agent
dedicated to his number. besides, I want to add another second client who
dialed the same number to the first client's conferance by the SIP agent.
the point is this: I call from PRI with SIP agent by the dial but they start
the conferance without entering the conferance room. when 2 call come enters
the conferance room being aware of that the SIP is busy. I need to meet the
calls and SIP in the same conferance room.
Here is my current Conferance Dial Plan
[conferance]
exten => _XX,1,Ringing(10)
exten => _XX,2,Answer
exten => _XX,3,SetGlobalVar(numara=${EXTEN})
exten => _XX,4,Dial(SIP/${EXTEN},30,m)
exten => _XX,5,Goto(${numara}-${DIALSTATUS},1)
exten => _XX,6,Meetme(${numara})
exten => _XX-BUSY,1,Meetme(${numara})
exten => _XX-ANSWER,1,Meetme(${numara})
exten => _XX-NOANSWER,1,Playback(jingle)
exten => _XX-NOANSWER,2,Hangup
exten => _XX-CHANUNAVAIL,1,Playback(jingle)
exten => _XX-CHANUNAVAIL,2,hangup
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[Asterisk-Users] How To conferance
How Can i make conferance like this Call came from PRI And joining Called Number Conferance Room (211) While joining progress. I want to make Asterisk call sip agent for 2nd conferance person When sip agent answer then SIP agent join to room(211) 1st.Conferance Person (PRI) 2nd. Conferance Person (SIP) If sip agent busy 1st person join conf room 211 directly How can i wrote exte. For this plan ___ 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
[Asterisk-Users] Dialing With Backgound Music
How can play music when is clients phone ringing in dial progress. ___ 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
RE: [Asterisk-Users] Answering without ringing from PRI
I made patch But when i wrote make im taking errors . ./gentone ringtone 440 480 Wavelength 1 (in samples): 18.18182 Minimum samples (1): 200 (11.00.3 wavelengths) Wavelength 1 (in samples): 16.7 Minimum samples (1): 50 (3.00.3 wavelengths) Need 200 samples Wrote ringtone.h gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-v1-0-03/10/05-14:53:33\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC-c -o chan_oss.o chan_oss.c gcc -shared -Xlinker -x -o chan_oss.so chan_oss.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-v1-0-03/10/05-14:53:33\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC-c -o chan_phone.o chan_phone.c gcc -shared -Xlinker -x -o chan_phone.so chan_phone.o gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-v1-0-03/10/05-14:53:33\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -o chan_zap.o chan_zap.c chan_zap.c: In function `pri_dchannel': chan_zap.c:7733: error: structure has no member named `proceeding' make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Steve Hanselman > Sent: Friday, April 08, 2005 12:05 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Answering without ringing from PRI > > Have you tried the latest CVS, there was a bug relating to > ALERTING which was fixed yesterday... > > -Original Message- > From: Ugur GUNCER [mailto:[EMAIL PROTECTED] > Sent: 08 April 2005 04:54 > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Answering without ringing from PRI > > I made that but still same no ringing for pri coming calls > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf > Of Mathew > > McKernan > > Sent: Friday, April 08, 2005 5:02 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [Asterisk-Users] Answering without ringing from PRI > > > > Hi, > > > > Where you have your 1st priority, I suspect you have it set to > > "Answer". > > Try changing this to Wait(1). Then on priority 2 put answer. i.e. > > > > Exten => s,1,Wait(1) > > Exten => s,2,Answer > > Exten => blah blah > > > > Hope that covers it, > > > > Thanks > > > > Mathew > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Ugur > > GUNCER > > Sent: Friday, 8 April 2005 11:39 AM > > To: 'Asterisk Users Mailing List - Non-
RE: [Asterisk-Users] Answering without ringing from PRI
I made that but still same no ringing for pri coming calls > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Mathew McKernan > Sent: Friday, April 08, 2005 5:02 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Answering without ringing from PRI > > Hi, > > Where you have your 1st priority, I suspect you have it set > to "Answer". > Try changing this to Wait(1). Then on priority 2 put answer. i.e. > > Exten => s,1,Wait(1) > Exten => s,2,Answer > Exten => blah blah > > Hope that covers it, > > Thanks > > Mathew > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Ugur GUNCER > Sent: Friday, 8 April 2005 11:39 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [Asterisk-Users] Answering without ringing from PRI > Importance: High > > > > How can i set asterisk for when call came from pri ring once > then answer pri call. > > In now call cames from pri then asterisk directly answering > pri call without ringing. Then my carries hangup call because > they said your box is answer without ringing > > > Iyi Calismalar > Saygilarimla > > > > Ugur GUNCER > Sistem Yoneticisi > Telebizz Tel. ve Int. Hizm. > > Office= +90 212 347 6959 > Gsm = +90 544 535 9737 > Fax = +90 212 347 6949 > > ___ > 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 > BEGIN:VCARD VERSION:2.1 N:Guncer;Ugur;;Bay FN:Ugur Guncer ([EMAIL PROTECTED]) ORG:Telebizz Telekomunikasyon ve Internet Hizmetleri Ltd. Sti.;IT TITLE:Sistem Yoneticisi TEL;WORK;VOICE:+90 (212) 347 69 59 TEL;CELL;VOICE:+90 (544) 535 97 37 TEL;WORK;FAX:+90 (212) 347 69 49 ADR;WORK;ENCODING=QUOTED-PRINTABLE:;Telebizz Istanbul;Akinci Bayiri=0D=0ASokak No:13 Cevre Apt. =0D=0AB Blok Da= ire: 5;Mecidiyekoy-Sisli, Istanbul;;80290;T=FCrkiye LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Telebizz Istanbul=0D=0AAkinci Bayiri=0D=0ASokak No:13 Cevre Apt. =0D=0AB Blo= k Daire: 5=0D=0AMecidiyekoy-Sisli, Istanbul 80290=0D=0AT=FCrkiye URL;WORK:http://www.telebizz.org.uk EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20041008T130547Z END:VCARD ___ 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
RE: [Asterisk-Users] Re: Answering without ringing from PRI
They are still telling not ringing Here is the carrier log part 08.04.2005 06:35:10 Connected without receiving ringing, getting call details on channel 239. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of LJ > Sent: Friday, April 08, 2005 5:48 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Re: Answering without ringing from PRI > > I had a similar problem where the PBX I was connecting to > would not recognize the answer until I set Ringing() before > the answer. I do not recall if I used a wait in between. It > was something like: > > Exten => 2688,1,Ringing() > Exten => 2688,2,Wait,1 > Exten => 2688,3,Answer > > Hope that helps. > > --LJ > > - Original Message - > From: "Ugur GUNCER" <[EMAIL PROTECTED]> > Newsgroups: gmane.comp.telephony.pbx.asterisk.user > Sent: Thursday, April 07, 2005 8:38 PM > Subject: Answering without ringing from PRI > > > > > > > > How can i set asterisk for when call came from pri ring > once then answer > > pri > > call. > > > > In now call cames from pri then asterisk directly answering > pri call > > without > > ringing. Then my carries hangup call because they said your > box is answer > > without ringing > > > > > > Iyi Calismalar > > Saygilarimla > > > > > > > > Ugur GUNCER > > Sistem Yoneticisi > > Telebizz Tel. ve Int. Hizm. > > > > Office = +90 212 347 6959 > > Gsm = +90 544 535 9737 > > Fax = +90 212 347 6949 > > > > > > > -- > -- > > > > ___ > > 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 > > ___ > 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 > ___ 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
[Asterisk-Users] Answering without ringing from PRI
How can i set asterisk for when call came from pri ring once then answer pri call. In now call cames from pri then asterisk directly answering pri call without ringing. Then my carries hangup call because they said your box is answer without ringing Iyi Calismalar Saygilarimla Ugur GUNCER Sistem Yoneticisi Telebizz Tel. ve Int. Hizm. Office = +90 212 347 6959 Gsm = +90 544 535 9737 Fax = +90 212 347 6949 BEGIN:VCARD VERSION:2.1 N:Guncer;Ugur;;Bay FN:Ugur Guncer ([EMAIL PROTECTED]) ORG:Telebizz Telekomunikasyon ve Internet Hizmetleri Ltd. Sti.;IT TITLE:Sistem Yoneticisi TEL;WORK;VOICE:+90 (212) 347 69 59 TEL;CELL;VOICE:+90 (544) 535 97 37 TEL;WORK;FAX:+90 (212) 347 69 49 ADR;WORK;ENCODING=QUOTED-PRINTABLE:;Telebizz Istanbul;Akinci Bayiri=0D=0ASokak No:13 Cevre Apt. =0D=0AB Blok Da= ire: 5;Mecidiyekoy-Sisli, Istanbul;;80290;T=FCrkiye LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Telebizz Istanbul=0D=0AAkinci Bayiri=0D=0ASokak No:13 Cevre Apt. =0D=0AB Blo= k Daire: 5=0D=0AMecidiyekoy-Sisli, Istanbul 80290=0D=0AT=FCrkiye URL;WORK:http://www.telebizz.org.uk EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20041008T130547Z END:VCARD ___ 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
[Asterisk-Users] Conferancing with different interface
Hi All How can i made conference first person coming from PRI and second person dialed from asterisk with SIP. How will be my extension conf I wrote extension for first person exten => _5463XX,1,Answer exten => _5463XX,2,MeetMe(1234|a); exten => _5463XX,3,Hangup But i dont know how can call the second person for invite to conferance with Can some body give me a example ___ 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
[Asterisk-Users] Cant Hear Any Sounds
I connect pri to asterisk with e100p card when i call from pri i cant hear any sound And when i call ip phone icant hear any sound. Does any one have idea ___ 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
[Asterisk-Users] Cant Hear Any Sound
I connect pri to asterisk with e100p card when i call from pri i cant hear any sound And when i call ip phone icant hear any sound. Does any one have idea ___ 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
RE: [Asterisk-Users] Best Performance
I try ulaw - alow but I cant hear any sound came from PRI to sip lines But i hear sip to sip lines Does any one have idea > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > tim panton > Sent: Wednesday, April 06, 2005 12:00 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Best Performance > > > On 5 Apr 2005, at 15:33, Ugur GUNCER wrote: > > > Hi > > > > Does anyone know what isthe best codec for sound syncr. And quality > > with > > asterisk+zyxel p200w > > I found 2 that work acceptably. > > If you have a good WiFi signal and are not using WEP then > (a/u)law work ok with the P2000W > > If your Wifi signal is less than perfect, or you want to use > WEP then you'll have to go for 729a - but that costs $10 / > channel from Digium. > > Tim. > > http://www.westhawk.co.uk/ > > ___ > 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 > ___ 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
[Asterisk-Users] Best Performance
Hi Does anyone know what isthe best codec for sound syncr. And quality with asterisk+zyxel p200w ___ 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
[Asterisk-Users] E100p zapata errors
Hi everyb. I was installed e100p card on my suse and made conf. Files when i start the asterisk i take some errors i try to many config files but i still take errors anyone have idea for this errors Errors [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 5 10:45:00 ERROR[5028]: chan_zap.c:6215 mkintf: Unable to get parameters Apr 5 10:45:00 ERROR[5028]: chan_zap.c:9155 setup_zap: Unable to register channel '1-15' Apr 5 10:45:00 WARNING[5028]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Apr 5 10:45:00 WARNING[5028]: loader.c:440 load_modules: Loading module chan_zap.so failed! Here is zaptel.conf asterisk:~ # less /etc/zaptel.conf #Configuration for EuroISDN (E1) span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone = us defaultzone=us Here is zapata.conf [trunkgroups] [channels] language=en context=default switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_net rxwink=150 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel => 1-15 channel => 17-31 context=default Here is lsmod output asterisk:/etc/asterisk # lsmod Module Size Used by nvram 13832 0 speedstep_lib 8452 0 freq_table 8576 0 thermal21896 0 processor 30400 1 thermal fan 9348 0 button 12432 0 battery15364 0 ac 10372 0 edd14620 0 ipv6 272256 17 evdev 13184 0 joydev 13760 0 sg 42528 0 st 43164 0 sr_mod 21156 0 ide_cd 8 0 cdrom 42652 2 sr_mod,ide_cd tg386660 0 ohci_hcd 25604 0 wct1xxp19104 0 zaptel184836 1 wct1xxp crc_ccitt 6144 1 zaptel sworks_agp 13088 0 agpgart37804 1 sworks_agp subfs 12672 2 dm_mod 63104 0 usbcore 120164 3 ohci_hcd reiserfs 265680 1 aacraid48048 2 sd_mod 22144 3 scsi_mod 121412 5 sg,st,sr_mod,aacraid,sd_mod Ztcfg output asterisk:/etc # ztcfg -vv Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. ___ 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
[Asterisk-Users] Sip registration Problems With Zyxel P2000W
Hi all, I bougth zyxel wifi phone but i cant register when i want to register phone to asterisk i recieve These errors I spend 6 hours to fix regist problem but i cant find the solution [9875] type=friend username=9875 secret=5789 host=dynamic context=default callerid="Ugur Guncer" <9875> canreinvite=no dtmfmode=rfc2833 nat=no Sip read: REGISTER sip:213.139.225.82:5060 SIP/2.0 Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: ;tag=5175B05114E474A31693 To: Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone Contact: Expires: 300 Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 85.99.110.143 : 43956 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: ;tag=5175B05114E474A31693 To: ;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 85.99.110.143:43956 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: ;tag=5175B05114E474A31693 To: ;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="0f3403ce" Content-Length: ___ 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
