search for: pri_net

.... hth -----Original Message----- From: Steve Totaro [mailto:asterisk@totarotechnologies.com] Sent: Wednesday, September 28, 2005 12:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net' Any ideas? 51] logger.c: [chan_zap.so] => (Zapata Telephony) Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found Sep 28 00:42:35 VERBOSE[5151] logger.c...

...h file or directory) Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata_additional.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata_additional.conf': Found Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net' Sep 28 00:42:35 ERROR[5151] chan_zap.c: Signalling must be specified before any channels are. Sep 28 00:42:35 WARNING[5151] loader.c: chan_zap.so: load_module failed, returning -1 Sep 28 00:42:35 WARNING[5151] loader.c: Loading module chan_zap.so failed! zaptel.conf span=1,0,0,esf,b8zs bchan...

If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. Imparticular, I would like to know that it works before I invest in the extra hardware. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.co...

...hocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=-6.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=yes callerid=asreceived group=1 context=from-pstn signalling=pri_cpe channel => 1-15 ;,17-31 => only 15 first channels on PRI group=2 context=from-ivr signalling=pri_net channel => 32-46,48-62 group=3 context=from-ivr-bis signalling=pri_net channel => 63-77,79-93 group=4 signalling=pri_net channel => 94-108,110-124 ============================================ Any ideas ? Regards Jerome

Hey, Thanks for the input Andrew. I did all you suggested but noticed that when I did the loopback test, the output *was not* there as you mentioned ("I'm set to pri_net, but the other side thinks it is pri_net!"). In fact, the same message as before kept repeating every second or so: >> Unnumbered frame: >> SAPI: 00 C/R: 0 EA: 0 >> TEI: 000 EA: 1 >> M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode e...

...ure if the span line is correct. Any help would be greatly appreciated. /etc/zaptel.conf loadzone=us defaultzone=us span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=24 /etc/asterisk/Zapata.conf [trunkgroups] [channels] language=us context=sasktel switchtype=national signalling=pri_net ; I have also tried pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=...

...-> pin3 pin5 <--> pin8 pin8 <--> pin7 I'm not really sure this is correct, since I've found only how to connect pins 1,2,4 and 5. The other pins are connected as I supposed it should be. This is my zapata.conf: Server#1 zapata.conf (Server#2 has the same zapata.conf but pri_net is pri_cpe) -------------------------------------- [channels] context=inicio switchtype=euroisdn signalling=pri_net rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=no transfer=no cancallf...

Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to "pri_net"...this cant be all? And the cable > pin1 <--> pin4> pin2 <--> pin5> pin3 <--> pin6> pin4 <--> pin1> pin5 <--> pin2> pin6 <--> pin3> pin5 <--> pin8> pin8 <--> pin7 Please Help and advice Thanks Altus

Hi guys, Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server? The pbx doesn't have sip and I want to come in off of a sip trunk and interface with the older system. I know I can use a pri card to hook in to the phone network, but can I use this same card to feed back the signaling as if I were the phone company to the older system? Thanks, Tom

...-- Registered channel 1, FXO Kewlstart signalling -- Registered channel 2, FXO Kewlstart signalling -- Registered channel 3, FXS Kewlstart signalling -- Registered channel 4, FXS Kewlstart signalling Dec 11 13:27:48 ERROR[1555]: chan_zap.c:9422 setup_zap: Unknown signalling method 'pri_net' Dec 11 13:27:48 ERROR[1555]: chan_zap.c:6209 mkintf: Signalling requested is FXS Kewlstart but line is in PRI Signalling signalling Dec 11 13:27:48 ERROR[1555]: chan_zap.c:9141 setup_zap: Unable to register channel '5-19' Dec 11 13:27:48 WARNING[1555]: loader.c:345 ast_load_resource:...

...my Intertel PBX. Has anyone had experience with troubleshooting this? I have call routing tables set up on the Intertel that *should* be taking care of this, but it's not working. Here are a few snippets of my config files - are there any glaring errors? zapata.conf: [channels] signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=1 context=logic;...

...gcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 callgroup=1 pickupgroup=1 immediate=no context=pri-external group = 1 signalling=pri_cpe channel => 1-15,17-31 context=pri-ericsson group = 2 signalling=pri_net channel => 32-46,48-62 context=pri-loopin group = 3 signalling=pri_cpe channel => 63-77,79-93 context=pri-loopout group = 4 signalling=pri_net channel => 94-108,110-124 #### /etc/asterisk/extensions.conf (just the part of it that matters) [pri-external] ; calls from the telco include =&g...

...plan = unknown prilocaldialplan = unknown echocancel = yes context=incoming group = 1 ; S/T port 1 channel => 1-2 group = 2 ; S/T port 2 channel => 4-5 group = 3 ; S/T port 3 channel => 7-8 group = 4 ; S/T port 4 channel => 10-11 ; E1 - for output to external Dialogic only ; we are pri_net in this case group = 9 pridialplan = unknown signalling=pri_net channel => 13-27,29-43 ------------------------------------------ STARTUP MODPROBES, ETC: #following for Quad BRI system: cd /usr/src/bri/bri-stuff.0.1.0-RC2g/qozap modprobe -v zaptel >>/var/log/asterisk/modprobe.log sleep...

hi all, i have one (teles) pbx with a BRI telephone and an outgoing E1 port. The outgoing E1 is connected to an pri_net port from my *. The incoming call will dail out to a h323 soft phone like openphone or sjphone or just netmeeting. The call will be conneted, but i don't hear any sound, from no one of the both sides. Can somebody help me? Thanks, Thomas.

I have a pri comming into a t100p in my asterisk box. I have a second t100p configured as pri_net connected to a nas server. I can route modem calls to the NAS with no problem, but I am concerned about isdn data connections. Will asterisk route 64k isdn calls between 2 t100p's? Also, should I expect worse modem connections than normal? Chad Sawyer -------------- next part -------------...

...25-47 dchan=48 --------------------------------------------------------------------- ;; /etc/asterisk/zapata.conf context = default switchtype = national signalling = pri_cpe group = 1 channel => 1-23 ;; The problematic card context = default switchtype = national overlapdial = yes signalling = pri_net group = 2 channel => 25-47 ------------------------------------------------------------------------ I tried with different options for the switchtype too. The signalling id definitely pri_net Would really be grateful for any help in this regard. Thanks and regards, - Ashish

...ridged=yes echotraining=yes rxgain=0.0 txgain=-6.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no busydetect=no callerid=asreceived group=1 context=from-pstn signalling=pri_cpe channel => 1-15 ;,17-31 => only 15 first channels on PRI group=2 context=from-ivr signalling=pri_net channel => 32-46,48-62 group=3 context=from-ivr-bis signalling=pri_net channel => 63-77,79-93 group=4 signalling=pri_net channel => 94-108,110-124 ============================================ Regards Jerome

...-------------- span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,2,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 loadzone = us defaultzone = us /etc/asterisk/zapata.conf ------------------------------------- [trunkgroups] [channels] context=from-pstn ;signalling=fxs_ks signalling=pri_net rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no faxdetect=incoming echotr...

...ion on a server by conection another asterisk server to it. I have the T-1 cards up and both set to bchan=1-23 and dchan=24 in zaptel.conf I am getting lost on the signalling in zapata.conf. using pri_cpe makes sense for my target box, but I am getting the following errors on my source box if I use pri_net or pri_cpe signalling. May 31 17:15:06 NOTICE[3790]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 May 31 17:15:06 NOTICE[3790]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 May 31 17:15:06 NOTICE[3790]: ch...

...i required to put please guide me. I will be greately appreciated your feedback on this regard. *configuration details* */etc/zaptel.conf* # Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 */etc/asterisk/zapata.conf* signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=euroisdn ;switchtype=national echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived ov...