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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: painkiller on September 16, 2007, 12:28:29 PM
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Hello S.
I connected my Qtek 9100 to Sail with voip, i can make internal call but no outbound calls.
When i use asterisk -vvvvvr :
new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script selintra completed, returning 0
-- Executing AGI("SIP/6000-09e01d08", "selintra|OutRoute|voipbuster") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (SetCallerID) Options: (iaxtrunkin)
-- AGI Script Executing Application: (Dial) Options: (SIP/9**********@iaxtrunkin)
-- Called 9**********@iaxtrunkin
-- SIP/iaxtrunkin-09de03a0 is making progress passing it to SIP/6000-09e01d08
-- AGI Script Executing Application: (Wait) Options: (1)
-- SIP/iaxtrunkin-09de03a0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- AGI Script Executing Application: (SetCallerID) Options: (0511212011)
-- AGI Script Executing Application: (Dial) Options: (SIP/9**********@*******11)
-- Called 9**********@*********
-- SIP/*********-09de03a0 is ringing
-- SIP/*********-09de03a0 answered SIP/6000-09e01d08
-- Attempting native bridge of SIP/6000-09e01d08 and SIP/********11-09de03a0
Wat i can see is that when i call from extension 6000 (qtek) the server give an extra 9 in before the telefoonnumber and complains that that is a invalid number, i need to remove that 9 but how ? What is causing this?
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Hello Painkiller,
Almost certainly, your pda is prepending the 9. SAIL/Asterisk will only modify the number if you set it to do so. As a quick fix, you can remove the 9 in the SIP channel by placing 9: in the transformation mask (in the SIP Trunk).
To fully understand what is happening, you will need to run a SIP trace...
tethereal -R sip -i eth0 -f "host address.of.your.ip-phone"
That should get you started.
Kind Regards
S
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You are right, some file in the pda i hade to modify with dailplains, thanks for de reply and for the quick workaround