Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
As it turns out the telco was not routing the calls to us, a little misktake they said after 3 days of being with no service. The line was not CAS, it was CCS, no need to compile unicall. Whatever they meant with your card has to be configured with DSS1 will remain in mystery. Maybe someone here can tell me what they mean. The configuration I previously listed is valid for lines in Panama City, Panama. With the telco being Cable Wireless Panama and the asterisk with a sangoma A102. If there's any Cable wireless tech reading this. Guys, your support s*cks big time. Thanks to all for your kind and prompt help. On 7/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: If you do not have any alarms and PRI debug span 1 still gives you nothing then you need to call your telco and say I'm not getting any Q.931 messages on the D-Channel. Stephen Bosch wrote: Erick Perez wrote: Yes I do. I even did a pri debug span 1 and when I call the asterisk box, it sees nothing. Hmn, well, that's telling. Are you using the correct cable? Is the cable plugged into the correct port on the card? The 102 is a two-port. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
Erick Perez wrote: Yes I do. I even did a pri debug span 1 and when I call the asterisk box, it sees nothing. Hmn, well, that's telling. Are you using the correct cable? Is the cable plugged into the correct port on the card? The 102 is a two-port. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
If you do not have any alarms and PRI debug span 1 still gives you nothing then you need to call your telco and say I'm not getting any Q.931 messages on the D-Channel. Stephen Bosch wrote: Erick Perez wrote: Yes I do. I even did a pri debug span 1 and when I call the asterisk box, it sees nothing. Hmn, well, that's telling. Are you using the correct cable? Is the cable plugged into the correct port on the card? The 102 is a two-port. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
Yes I do. I even did a pri debug span 1 and when I call the asterisk box, it sees nothing. On 7/26/07, Idris AVCI [EMAIL PROTECTED] wrote: Do you have any extension in default context of your extensions.conf file to accept incoming calls ? It must be something like; exten = 12345678,1,Answer() exten = 12345678,2,Playback(Welcome) ... 12345678 = The DID number you are calling to reach E1 Idris -Original Message- From: Erick Perez [mailto:[EMAIL PROTECTED] Sent: Thursday, July 26, 2007 7:03 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 My /etc/asterisk/zapata.conf is: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no #include zapata-auto.conf Zapata-auto.conf has: callerid=asreceived ;Sangoma A102 port 1 [slot:1 bus:4 span: 1] switchtype=euroisdn context=from-pstn group=0 signalling=pri_cpe channel = 1-15,17-31 Note: According to the tech support in the local telco, my E1 should be: E1 PRI, CAS, HDB3, NCRC4, DSS1 However if I configure the card for CAS, it will never connect. My card is currently configured (and makes only outgoing calls) as: E1 PRI, CCS, HDB3,NCRC4 (i have no idea what dss1 is or where it goes) My /etc/wanpipe/wanpipe1.conf is: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 4 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES thanks for your help. -- Erick Perez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
Do you have any extension in default context of your extensions.conf file to accept incoming calls ? It must be something like; exten = 12345678,1,Answer() exten = 12345678,2,Playback(Welcome) ... 12345678 = The DID number you are calling to reach E1 Idris -Original Message- From: Erick Perez [mailto:[EMAIL PROTECTED] Sent: Thursday, July 26, 2007 7:03 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 My /etc/asterisk/zapata.conf is: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no #include zapata-auto.conf Zapata-auto.conf has: callerid=asreceived ;Sangoma A102 port 1 [slot:1 bus:4 span: 1] switchtype=euroisdn context=from-pstn group=0 signalling=pri_cpe channel = 1-15,17-31 Note: According to the tech support in the local telco, my E1 should be: E1 PRI, CAS, HDB3, NCRC4, DSS1 However if I configure the card for CAS, it will never connect. My card is currently configured (and makes only outgoing calls) as: E1 PRI, CCS, HDB3,NCRC4 (i have no idea what dss1 is or where it goes) My /etc/wanpipe/wanpipe1.conf is: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 4 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES thanks for your help. -- Erick Perez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
