Hi there, I'm setting up a Asterisk network and I ran into some problems ... as you might have guessed :)
The set up is like this: Internal Communication in the company should be handled through softphones over an asterisk server (works). Outbound Communication should be handled through a HUAWEI E169 stick, accessed by the chan_dongle project. http://code.google.com/p/asterisk-chan-dongle/ When I call internal numbers, everything works fine, but when I try to access outside, I get the following error: == Using SIP RTP CoS mark 5 -- Executing [06766770031@internal:1] Answer("SIP/1001-00000023", "") in new stack -- Executing [06766770031@internal:2] Dial("SIP/1001-00000023", "dongle0/r1/06766770031,20,r") in new stack [Sep 13 11:33:31] WARNING[9835]: channel.c:5603 ast_request: No channel type registered for 'dongle0' [Sep 13 11:33:31] WARNING[9835]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'dongle0' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [06766770031@internal:3] Hangup("SIP/1001-00000023", "") in new stack == Spawn extension (internal, 06766770031, 3) exited non-zero on 'SIP/1001-00000023' >From googling my way around, I know this type of error normally relates to a module not being loaded, but chan_dongle.so shows up when I type a "module show". I've been fiddling around with this for days and frankly I don't really know where the problem could lie. Below are excerpts from sip.conf and extensions.conf SIP.conf <code> [general] bindport = 5060 bindaddr = 192.168.61.25 tcpbindaddr = 192.168.61.25 tcpenable = yes context = internal transport = udp disallow = all allow = gsm allow = ulaw allow = alaw [dongle0] type=friend context=internal audio=/dev/ttyUSB1 data=/dev/ttyUSB2 imei=359638011610601 imsi=232018830482446 transport=udp disallow = all allow = gsm allow = ulaw allow = alaw [1000] type=friend callerid = "Benny" <1000> secret=1000 host=dynamic canreinvite=no dtmfmode=rfc2833 mailbox=1000 disallow=all allow=gsm allow=ulaw allow=alaw transport=udp context=internal [1001] type=friend callerid = "Timme" <1001> secret=1001 host=dynamic canreinvite=no dtmfmode=rfc2833 mailbox=1001 disallow=all allow=gsm allow=ulaw allow=alaw </code> Extensions.conf <code> [internal] ; for 4-digit numbers, assume it's a SIP number in our own context ; call it exten => _XXXX,1,Answer() exten => _XXXX,n,Dial(SIP/${EXTEN},20,r) exten => _XXXX,n,Hangup ; else ; for a number starting with zero try to call via Dongle exten => _0X.,1,Answer() exten => _0X.,n,Dial(dongle0/r1/${EXTEN},20,r) exten => _0x.,n,Hangup </code> Please shed some light on this ..... Kind regards, Benedikt
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users