International plan
Hi there,
Can anybody post proper dial plan for international calls please?
I tried obvious choices with ( 9 + 011. and 9 + 011n. ) but system cuts me off immediately after dialing second digit. I need to dial 011 + many digits after that to complete call.
Maybe problem is in numbers stripping?
Thank you,
Zack
a SIP phone collects all digits, and sends this entire number to PABX when hitting send, or not pressing a button for some seconds
Some phones have built in dialplans, that quess a number is complete after 3 digits (internal extension range) and send this number out immediately.
So check the phones dial plan.....
The Default dial plan for international calls on trixbox pro is:
9+ 011. Strip 1 digit
For international calls, set your local dial plan on your phone to this:
0|911|900X|*[13]|9905X+#|9011X+#
Also notice that you do not have any local or long distance number dial plans here. If you would like to add those use this as long as your in the US.
0|911|900X|*[13]|9905X+#|9011X+#|9[2-9]XX[2-9]XXXXXX|91[2-9]XX[2-9]XXXXXX
I tried first string and dialed out few numbers in Europe (one of them fax). Dials out normally but gives me busy signal after few seconds.
This is more complicated than I thought ... here is complete Local Dial Plan copied from Aastra phone:
0|911|900X|*[13]|9905X+#|9416X+#|85[05]5|[78]XXX|*[65]X|9[469]11|9[2-9]XXXXXXXXX|*[78][456]XXXX|91[2-9]XX[2-9]XXXXXX
Ddavidson, thank you for looking into that. I actually added string you suggested this morning:
0|911|900X|*[13]|9905X+#|9011X+#|9416X+#|85[05]5|[78]XXX|*[65]X|9[469]11|9[2-9]XXXXXXXXX|*[78][456]XXXX|91[2-9]XX[2-9]XXXXXX
(see third group of numbers, I wanted to bold it or red it but don't know how to do that in this forum).
Anyway, what happens now after this change is following: phone is not bothering me dialing out number like 9-011-385-1-619-5xxx (my mom in Europe!) but after dialing that there is nothing in line. No ringing no busy or whatsoever tone. I wait like 20-30 second like that and hang up.
When I am using Bell from home phone it rings within seconds after last digit.
I called AAstra tech support, got them on line and "they will look into that".
Zack
Hey Zack,
You also need to make sure your dial plan has:
9+ 011. Strip 1 digit
If you do not have this in your main dial plan for trixbox it will not work. You must have a dial plan both on your phone (so the phone knows what digits it can dial) then on your trixbox (so the phone system knows what digits to send to telco).
Derek,
Something in Trixbox is preventing those numbers from dialing properly. I dial out and get busy signal, always, no matter if I take VoIP line or POTS line to dial out.
Here is string in the phone suggested by Aastra tech support:
11xxxxxxxxxxx|911|900X|*[13]|9905X+#|9011X+#|9416X+#|85[05]5|[78]XXX|*[65]X|9[469]11|9[2-9]XXXXXXXXX|*[78][456]XXXX|91[2-9]XX[2-9]XXXXXX.
Trixbox dial plans (I don't know how to attach screenshot to this forum):
Showing 9 Dial Plan Entries
Delete Dial String Description Type Primary Route Strip Prepend
9 + nxxxxxx local call 1
9 + nxxnxxxxxx Standard long di... local call VoIP: voicepulse 1
9 + 11 Emergency 911 local call 0
9 + 411 local call 1
9 + 611 local call 1
9 + 1nxxnxxxxxx Standard long di... long distance VoIP: voicepulse 1
9 + 011xxxxxxxxxxx interantional calls international VoIP: voicepulse 1
9 + 416. toll free Local System 1
9 + 905. toll free Local System 1
I have never set the dial plan like this:
9 + 011xxxxxxxxxxx
If you are sure you have the correct number of digits then that should work. Make sure to strip 1 digit so you are not sending the 9 to telco.
I usually use the default which is:
9+ 011. Strip 1 digit
This has always worked for me.
Your phone dial plan includes:
9011X+#
Which will allow the phone to make international calls.
I would try change that one dial rule, then pull a call trace and post it if you continue to have issues.
Derek,
I had 9+011. plan from before (as it was suggested by others in the forum) and it didn't do any good. I can try it again.
I don't know how to do call trace. I guess I have to bring someone who is knowledgeable in Linux to do it on my machine and I'll post findings but after phone string was suggesetd by Aaatra and they made calls themselves using same thing on same kind of phone it makes me thing that there is something in the Trixbox screwing things up. Needles to say, I can take same POTS line, dial out using Bell and get call going within a seconds.
Here is copy of CLI:
---
-- SIP/voicepulse-08686408 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Dial("SIP/SOFTPHONE001-b7303e08", "Local/01138765839222@internal") in new stack
Sep 24 11:42:38 NOTICE[9490]: chan_local.c:530 local_alloc: No such extension/context 01138765839222@internal creating local channel
Sep 24 11:42:38 NOTICE[9490]: app_dial.c:1082 dial_exec_full: Unable to create channel of type 'Local' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Busy("SIP/SOFTPHONE001-b7303e08", "") in new stack
Transmitting (NAT) to 209.151.130.12:51588:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 209.151.130.12:51588;branch=z9hG4bK-d8754z-893f720a0475de7c-1---d8754z-;received=209.151.130.12;rport=51588
From: "SOFTPHONE001"
To: "901138765839222"
Call-ID: OWFjOTdjZGRkOGExYWMyYThjNjIzZjY3MjEzYWI0NmY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
== Spawn extension (internal, 901138765839222, 107) exited non-zero on 'SIP/SOFTPHONE001-b7303e08'
-- Executing ResetCDR("SIP/SOFTPHONE001-b7303e08", "w") in new stack
-- Executing NoCDR("SIP/SOFTPHONE001-b7303e08", "") in new stack
Sep 24 11:42:38 NOTICE[9490]: cdr.c:445 ast_cdr_free: CDR on channel 'SIP/SOFTPHONE001-b7303e08' lacks end
-- Executing GotoIf("SIP/SOFTPHONE001-b7303e08", "1?5") in new stack
-- Goto (internal,h,5)
-- Executing Hangup("SIP/SOFTPHONE001-b7303e08", "") in new stack
== Spawn extension (internal, h, 5) exited non-zero on 'SIP/SOFTPHONE001-b7303e08'
trixbox128168*CLI> sip no debug
<-- SIP read from 209.151.130.12:51588:
ACK sip:901138765839222@209.151.141.113:5060 SIP/2.0
Via: SIP/2.0/UDP 209.151.130.12:51588;branch=z9hG4bK-d8754z-893f720a0475de7c-1---d8754z-;rport
To: "901138765839222"
From: "SOFTPHONE001"
Call-ID: OWFjOTdjZGRkOGExYWMyYThjNjIzZjY3MjEzYWI0NmY.
CSeq: 2 ACK
Content-Length: 0
My problem is solved! Voicepulse tech support actually called me and they were observing my call in real time. After that they said to remove strip first 4 digits in Trixbox dial plan and that was it! He said that is a way their system handles the calls.
Thank you everyone who wanted to help.


Member Since:
2008-07-20