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...ip with exten "${EXTEN}") exten => _j.,n,Set(3digitexten=${EXTEN:12:3} exten => _j.,n,NoOp("Callerid is " ${CALLERID(all)} ) exten => _j.,n,GoTo(from-outside,${3digitexten},1) [from-outside] exten => 123,1,NoOp() exten => 123,n,Answer() exten => 123,n,Dial(SIP/jnctn/1212xxxyyyy) exten => 123,n,HangUp() sip.conf: [general] externaddr=xx.yyy.zz.aa nat=yes directmedia=no ; tried nonat sip show peer jnctn: Insecure : invite Force rport : Yes ......... DirectMedia : No sip show peer teliax: Insecure : port,invite Force rport : Ye...

...kup=yes videosupport=yes echocancelwhenbridged=yes dtmfmode=rfc2833 disallow=all ; First disallow all codecs allow=ulaw allow=alaw ; Allow codecs in order of allow=ilbc ; preference allow=gsm allow=h261 register => budzhaus:SECRETHERE at ekiga.net register => obitori:SECRETHERE at jnctn.net SNIP [jnctn] type=peer host=sip.jnctn.net username=obitori secret=SECRETHERE fromdomain=jnctn.net insecure=very nat=yes qualify=yes context=incoming [101] type=friend secret=SECRETHERE qualify=yes ; Qualify peer is not more than 2000 mS away nat=no ; This phone is not natted host...

...is not recognized as "In Use". Here is the output from "queue show" prior to the call: *CLI> queue show QUEUE_3 has 0 calls (max unlimited) in 'wrandom' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Todd1 (SIP/14153436587 at jnctn) (dynamic) (Not in use) has taken no calls yet No Callers Here is the output when actually connected to the inbound caller: *CLI> queue show QUEUE_3 has 0 calls (max unlimited) in 'wrandom' strategy (1s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Todd1...

...=0 LINEOUT=Zap/1 ; Line to use in emergency #include "extensions.conf.macro" #include "extensions.conf.telx" [from-telx-atl] include => internal include => ext-external-from-atl [from-telx-NY17S] include => internal include => ext-internal [from-jnctn] include => aa-main exten => _NXXNXXXXXX,1,Goto(aa-main,s,1) [from-swifttel] exten => _NXXNXXXXXX,1,NoOp("Context is from-swifttel") exten => _NXXNXXXXXX,2,Goto(aa-main,s,1) [default] include => aa-main exten => _NXXNXXXXXX,1,NoOp("Context is default") exten...

...s, for example, in extensions.conf: [default-user-dial] ; Any North American ten-digit number. exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@my_sip_provider) In our case, we actually register with our SIP origination provider, so we have this IP trunk: [junction_networks] fromdomain=jnctn.net host=sip.jnctn.net port=5060 insecure=very username=this_user secret=this_password type=peer qualify=no canreinvite=no dtmfmode=rfc2833 But in addition, in the [general] context at the top of sip.conf, we have: register => our_user:our_password at sip.jnctn.net As you can see,...

...0.10.45 D A 5060 OK (133 ms) But pings are < 1ms: ping 10.10.10.42 ........ rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms Why are the sip latencies so high? And is it a problem? And if so, how do I fix it? FWIW, latencies to outside providers over nat are close to ping: jnctn/.... 5060 OK (7 ms) teliax/... N 5060 OK (7 ms) ping <teliax ip address> ........ rtt min/avg/max/mdev = 3.471/4.120/4.466/0.288 ms sean

Dear Alex; Thanks alot for your nice help. This is if I need to let Asterisk register with another softswitch (so I used register =>), what if I need asterisk to send call for the softswitch without register to it (directly)? If I removed the register => then how it will distiguish the IP address in the "host" at the [sip_trunk] is the IP address of the softswitch that need to

...asterisk box to any of the internal phones is < 1ms. My traceroute [v0.75] PBX Thu Oct 21 13:58:36 2010 Host 10.10.10.42 Loss% Snt Last Avg Best Wrst StDev 0.0% 65 0.6 0.6 0.5 1.3 0.2 And latencies to outside sip providers are low: jnctn/... 66.227.100.20 5060 OK (7 ms) teliax/.. 8.14.120.23 N 5060 OK (7 ms) What gives? Isn't sip show peers measuring latency? Why so different from ping? And, more importantly, do I have a problem? sean

Trying to set up Junction Networks for outgoing on 1.6.1: extensions.conf: exten => _99X.,n,Dial(IAX2/jnctn_out/${called-num}) iax.conf [jnctn_out] type=peer host=iax.jnctn.net username= secret= qualify=yes I'm not using realtime. But CLI: -- Executing [99xxxyyyy at internal:3] Dial("DAHDI/1-1", "IAX2/jnctn_out/1wwwxxxzzzz") in new stack [2009-06-27 14:21:45] WARNING[25...

I need 100 DID's in a specific rate center (916-854-xxxx). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas