[Asterisk-Users] Need help on * anf HFC.
Hi, I'm a newbie on * trying to setup an HFC card. I'm locked for many days getting the all-circuits-busy. And no idea what else to look for/how to diagnose. I'm in Spain, I've tried changing many parameters on zapata/zaptelcong with no luck, also NT TE modes (honsetly, I've no idea what is). Any clue will be very much appreciated! I've installed [EMAIL PROTECTED] on my RH9, and on top of that, bristuff-0.2.0-RC7f (that reinstalls asterisk). Here you have what I've: lspci output: 01:02.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) When loading zaptel drivers: Mar 6 21:29:13 linux-1 kernel: Zapata Telephony Interface Registered on major 196 Mar 6 21:29:13 linux-1 kernel: PCI: Enabling device 01:02.0 ( - 0003) Mar 6 21:29:13 linux-1 kernel: zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xf91c5e00 fifo 0xf7598000(0x37598000) IRQ 5 HZ 100 Mar 6 21:29:13 linux-1 kernel: zaphfc: Card 0 configured for NT mode Mar 6 21:29:13 linux-1 kernel: zaphfc: 1 hfc-pci card(s) in this box. Mar 6 21:29:13 linux-1 kernel: Registered tone zone 3 (Netherlands) mar 6 21:29:13 linux-1 zaptel: Loading zaptel framework: succeeded Mar 6 21:29:15 linux-1 kernel: Specify address with base=0xN Mar 6 21:29:15 linux-1 kernel: Registered Tormenta2 PCI Mar 6 21:29:17 linux-1 kernel: Registered tone zone 3 (Netherlands) mar 6 21:29:17 linux-1 zaptel: Running ztcfg: succeeded mar 6 21:29:34 linux-1 su(pam_unix)[21409]: session opened for user asterisk by (uid=0) mar 6 21:29:34 linux-1 su(pam_unix)[21409]: session closed for user asterisk mar 6 21:29:40 linux-1 su(pam_unix)[21484]: session opened for user asterisk by (uid=0) mar 6 21:29:40 linux-1 su(pam_unix)[21484]: session closed for user asterisk Mar 6 21:30:01 linux-1 kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun. Mar 6 21:30:49 linux-1 kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun. Mar 6 21:32:26 linux-1 last message repeated 2 times Mar 6 21:34:03 linux-1 last message repeated 2 times Mar 6 21:35:40 linux-1 last message repeated 2 times My /etc/zaptel.conf: # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 My zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode ;signalling = bri_cpe_ptmp ; p2p TE mode ;signalling = bri_cpe ; p2mp NT mode ;signalling = bri_net_ptmp ; p2p NT mode signalling = bri_net pridialplan = local ;prilocaldialplan = local ; nationalprefix = 0 ;internationalprefix = 00 ; trust user provided callerid (clip no screening)? ;pritrustusercid = no echocancel=yes ;echotraining = 100 ;echocancelwhenbridged=yes immediate=yes group = 1 context=outbound-trunks channel = 1-2 Asterisk console while trying to use the dial out trunk: Mar 6 21:40:01 DEBUG[21452]: Setting NAT on RTP to 0 Mar 6 21:40:01 DEBUG[21452]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Found Mar 6 21:40:01 DEBUG[21452]: Setting NAT on RTP to 0 Mar 6 21:40:01 DEBUG[21452]: Check for res for 200 Mar 6 21:40:01 DEBUG[21452]: Call from user '200' is 1 out of 0 Mar 6 21:40:01 DEBUG[21452]: build_route: Contact hop: Roser Roca sip:[EMAIL PROTECTED]:5061 Mar 6 21:40:01 VERBOSE[21452]: -- Executing Macro(SIP/200-1cf6, dialout-default|9639712471) in new stack Mar 6 21:40:01 WARNING[21452]: ast_yyerror(): syntax error: parse error; Input: fooEl Serrat = foo ^ ^ Mar 6 21:40:01 DEBUG[21452]: Expression is 'fooEl' Mar 6 21:40:01 VERBOSE[21452]: -- Executing GotoIf(SIP/200-1cf6, fooEl?4) in new stack Mar 6 21:40:01 DEBUG[21452]: Not taking any branch Mar 6 21:40:01 VERBOSE[21452]: -- Executing SetCallerID(SIP/200-1cf6, El Serrat) in new stack Mar 6 21:40:01 VERBOSE[21452]: -- Executing Goto(SIP/200-1cf6, 6) in new stack Mar 6 21:40:01 VERBOSE[21452]: -- Goto (macro-dialout-default,s,6) Mar 6 21:40:01 VERBOSE[21452]: -- Executing Dial(SIP/200-1cf6, ZAP/g0/9639712471) in new stack Mar 6 21:40:01 NOTICE[21452]: Unable to create channel of type 'ZAP' Mar 6 21:40:01 VERBOSE[21452]: == Everyone is busy/congested at this time Mar 6 21:40:01 DEBUG[21452]: Exiting with DIALSTATUS=CHANUNAVAIL. Mar 6 21:40:01 VERBOSE[21452]: -- Executing Macro(SIP/200-1cf6, outisbusy) in new stack Mar 6 21:40:01 VERBOSE[21452]: -- Executing Playback(SIP/200-1cf6, allison7/all-circuits-busy-now) in new stack Mar 6 21:40:01 DEBUG[21452]: Ooh, format changed from unknown to ulaw Mar 6 21:40:01 DEBUG[21452]: Scheduling timer at 160 sample intervals Mar 6 21:40:01 VERBOSE[21452]: -- Playing 'allison7/all-circuits-busy-now' (language 'en') Mar 6 21:40:01 DEBUG[21452]: Stopping
Re: [Asterisk-Users] Need help on * anf HFC.
Hello, I don't know if your zaptel.conf and zapata.conf setup regarding your isdn is correct, but if you use the default AMP setup, you need to assign your channels to group 0 for dialing out, and assign it to context from-pstn if you want to receive calls. group = 0 context=from-pstn channel = 1-2 BTW, i'm from Spaintoo, and I'm really interested in knowing if your setup works ;) On Sun, 06 Mar 2005 21:41:17 +0100, Ramon Roca [EMAIL PROTECTED] wrote: [channels] group = 1 context=outbound-trunks channel = 1-2 Mar 6 21:40:01 VERBOSE[21452]: -- Executing Dial(SIP/200-1cf6, ZAP/g0/9639712471) in new stack g0 means channel group 0, and you had group 1 Julian. ___ 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] Need help on * anf HFC.
Hey Julian, thanks! It really make a difference. Thanks for pointing me to this. Stupid newbie mistake. Yes, I'm using AMP, it was bundled with [EMAIL PROTECTED] Now I'm not longer getting the all-the-circuits-are-busy-now, but still doesn't dial out, now I'm getting the congestion tone. Maybe I'm loading the zaphfc with wrong parameters for Spanish ISDN? I'm just using a regular ISDN at home, and plugged the RJ45 cable at the same port where was the Euromix RDSI phone. Here it is the current * console while dialing out: Mar 6 22:44:58 DEBUG[3700]: Setting NAT on RTP to 0 Mar 6 22:44:58 DEBUG[3700]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Found Mar 6 22:44:58 DEBUG[3700]: Setting NAT on RTP to 0 Mar 6 22:44:58 DEBUG[3700]: Check for res for 200 Mar 6 22:44:58 DEBUG[3700]: Call from user '200' is 1 out of 0 Mar 6 22:44:58 DEBUG[3700]: build_route: Contact hop: Roser Roca sip:[EMAIL PROTECTED]:5061 Mar 6 22:44:58 VERBOSE[3700]: -- Executing Macro(SIP/200-bd90, dialout-default|9639712471) in new stack Mar 6 22:44:58 WARNING[3700]: ast_yyerror(): syntax error: parse error; Input: fooEl Serrat = foo ^ ^ Mar 6 22:44:58 DEBUG[3700]: Expression is 'fooEl' Mar 6 22:44:58 VERBOSE[3700]: -- Executing GotoIf(SIP/200-bd90, fooEl?4) in new stack Mar 6 22:44:58 DEBUG[3700]: Not taking any branch Mar 6 22:44:58 VERBOSE[3700]: -- Executing SetCallerID(SIP/200-bd90, El Serrat) in new stack Mar 6 22:44:58 VERBOSE[3700]: -- Executing Goto(SIP/200-bd90, 6) in new stack Mar 6 22:44:58 VERBOSE[3700]: -- Goto (macro-dialout-default,s,6) Mar 6 22:44:58 VERBOSE[3700]: -- Executing Dial(SIP/200-bd90, ZAP/g0/9639712471) in new stack Mar 6 22:44:58 VERBOSE[3700]: -- Called g0/9639712471 Mar 6 22:45:02 VERBOSE[3700]: -- Channel 0/1, span 1 got hangup Mar 6 22:45:02 DEBUG[3700]: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Mar 6 22:45:02 DEBUG[3700]: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1 Mar 6 22:45:02 DEBUG[3700]: Already hungup... Calling hangup once, and clearing call Mar 6 22:45:02 DEBUG[3700]: disabled echo cancellation on channel 1 Mar 6 22:45:02 DEBUG[3700]: Set option TDD MODE, value: OFF(0) on Zap/1-1 Mar 6 22:45:02 DEBUG[3700]: Updated conferencing on 1, with 0 conference users Mar 6 22:45:02 DEBUG[3700]: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 Mar 6 22:45:02 DEBUG[3700]: disabled echo cancellation on channel 1 Mar 6 22:45:02 VERBOSE[3700]: -- Hungup 'Zap/1-1' Mar 6 22:45:02 VERBOSE[3700]: == No one is available to answer at this time Mar 6 22:45:02 DEBUG[3700]: Exiting with DIALSTATUS=NOANSWER. Mar 6 22:45:02 VERBOSE[3700]: -- Executing Congestion(SIP/200-bd90, ) in new stack Mar 6 22:45:02 VERBOSE[3700]: == Spawn extension (macro-dialout-default, s, 7) exited non-zero on 'SIP/200-bd90' in macro 'dialout-default' Mar 6 22:45:02 VERBOSE[3700]: == Spawn extension (from-internal, 9639712471, 1) exited non-zero on 'SIP/200-bd90' Mar 6 22:45:02 VERBOSE[3700]: -- Executing Macro(SIP/200-bd90, hangupcall) in new stack En/na Julian J. M. ha escrit: Hello, I don't know if your zaptel.conf and zapata.conf setup regarding your isdn is correct, but if you use the default AMP setup, you need to assign your channels to group 0 for dialing out, and assign it to context from-pstn if you want to receive calls. group = 0 context=from-pstn channel = 1-2 BTW, i'm from Spaintoo, and I'm really interested in knowing if your setup works ;) On Sun, 06 Mar 2005 21:41:17 +0100, Ramon Roca [EMAIL PROTECTED] wrote: [channels] group = 1 context=outbound-trunks channel = 1-2 Mar 6 21:40:01 VERBOSE[21452]: -- Executing Dial(SIP/200-1cf6, ZAP/g0/9639712471) in new stack g0 means channel group 0, and you had group 1 Julian. ___ 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
