Actually I can start and receive SIP calls (PC client, iphone client) but?I have an issue with calling external number throught PSTN (certified-asterisk-1.8.11-cert2). I'm having this ?error when making a call: *CLI> ? == Using SIP RTP CoS mark 5 ? ? -- Executing [9000 at local:1] Dial("SIP/3000-00000006", "DAHDI/1/4384019357,10") in new stack [Jun 23 16:18:09] WARNING[28781]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) ? == Everyone is busy/congested at this time (1:0/0/1) ? ? -- Executing [9000 at local:2] Hangup("SIP/3000-00000006", "") in new stack ? == Spawn extension (local, 9000, 2) exited non-zero on 'SIP/3000-00000006' My configs : *CLI> dahdi show channels ? ?Chan Extension ?Context ? ? ? ? Language ? MOH Interpret Blocked ? ?State ?pseudo ? ? ? ? ? ?default ? ? ? ? ? ? ? ? ? ?default ? ? ? In Service *CLI> dahdi show status Description ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?Alarms ?IRQ ? ?bpviol CRC Fra Codi Options ?LBO Wildcard TDM410P ? ? ? ? ? ? ? ? ? ? ? ? OK ? ? ?0 ? ? ?0 ? ? ?0 CAS Unk ? ? ? ? ? 0 db (CSU)/0-133 feet (DSX-1) *CLI> root at my-PC:/usr/src/certified-asterisk-1.8.11-cert2# lsmod | grep dahdi dahdi_echocan_mg2 ? ? ?12998 ?4 dahdi_voicebus ? ? ? ? 58608 ?1 wctdm24xxp dahdi ? ? ? ? ? ? ? ? 220595 ?3 dahdi_echocan_mg2,wctdm24xxp,dahdi_voicebus crc_ccitt ? ? ? ? ? ? ?12667 ?2 wctdm24xxp,dah extensions.conf [local] exten => 100,1,Dial(gtalk/asterisk/Myaccount.voip at gmail.com) exten => 2000,1,Dial(SIP/2000,10) exten => 3000,1,Dial(SIP/3000,10) exten => 9000,1,Dial(DAHDI/1/MyCellPhoneNumber,10) exten => 9000,2,hangup() root at My-PC:/usr/src/certified-asterisk-1.8.11-cert2# dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM410P name=WCTDM/0 manufacturer=Digium devicetype=Wildcard TDM410P location=PCI Bus 04 Slot 01 basechan=1 totchans=4 irq=0 type=analog port=1,FXO port=2,FXO port=3,FXS port=4,FXS root at My-PC:/usr/src/certified-asterisk-1.8.11-cert2# dahdi_hardware pci:0000:04:00.0 wctdm24xxp+ d161:8005 Wildcard TDM410P
On Sat, Jun 23, 2012 at 10:32 AM, neo haux <neo.haux at gmx.com> wrote:> Actually I can start and receive SIP calls (PC client, iphone client) > but?I have an issue with calling external number throught PSTN > (certified-asterisk-1.8.11-cert2). > > I'm having this ?error when making a call: > > *CLI> ? == Using SIP RTP CoS mark 5 > ? ? -- Executing [9000 at local:1] Dial("SIP/3000-00000006", > "DAHDI/1/4384019357,10") in new stack > [Jun 23 16:18:09] WARNING[28781]: app_dial.c:2218 dial_exec_full: > Unable to create channel of type 'DAHDI' (cause 0 - Unknown) > ? == Everyone is busy/congested at this time (1:0/0/1) > ? ? -- Executing [9000 at local:2] Hangup("SIP/3000-00000006", "") in new stack > ? == Spawn extension (local, 9000, 2) exited non-zero on 'SIP/3000-00000006' > > > My configs : > *CLI> dahdi show channels > ? ?Chan Extension ?Context ? ? ? ? Language ? MOH Interpret > Blocked ? ?State > ?pseudo ? ? ? ? ? ?default ? ? ? ? ? ? ? ? ? ?default > ?? ? ? In ServiceWhere are your channels? That's why you are receiving the error "Unable to create channel of type 'DAHDI'". Define your channel groups in /etc/asterisk/chan_dahdi.conf Then is should look like this: # asterisk -rx "dahdi show channels" Chan Extension Context Language MOH Interpret Blocked State pseudo default default In Service 1 from-external en default In Service 2 from-external en default In Service 3 from-external en default In Service 4 from-external en default In Service 5 from-external en default In Service 6 from-external en default In Service ... - Julian
On Saturday 23 June 2012, neo haux wrote:> Actually I can start and receive SIP calls (PC client, iphone client) > but I have an issue with calling external number throught PSTN > (certified-asterisk-1.8.11-cert2).I notice the number you Dial()led didn't start with a zero. Check with your telco about this if you like but you almost certainly need to include the initial 0 of the STD code when using an analogue exchange line, because a TDM410P simply emulates standard subscriber's apparatus. Basically, your Asterisk box is just a subscriber dialling out on a POTS phone; and it has to do exactly what a person with a cheap phone would do. That is, the STD code (including initial 0) and subscriber's number for someone in a different town; or just the number for a call to someone in the same town (though dialling the code for a local call won't break anything). For IDD you need to dial 00, wait awhile, then the code for the destination country, STD code *without* the initial 0 (note, some small countries don't use STD codes) then subscriber's number. -- AJS Answers come *after* questions.