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Old May 10th, 2008, 01:12 AM
wycowboy911 wycowboy911 is offline
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Unhappy No DTMF or Voice on call in to asterisk.

I've tried looking through the forums and verified open ports on my router, but I still can't get voice or DTMF to come across when dialing into my box using my IPKall number.
Voice and DTMF DO work between extensions.


My server is at 192.168.1.6 and I've set my router's DMZ to the same IP.
I've also opened the following ports for TCP AND UDP traffic (also tried separating the TCP/UDP settings like their supposed to be) and forwarded them to the same IP: 5060, 5004, 10000, 3478, 16348-32768

Here's a copy of my sip.conf file.

[general]
context=default
port=5060
srvlookup=yes

register => user:XXXX@us.voxalot.com
register => user:XXXX@us.voxalot.com
register => user:XXXX@proxy01.sipphone.com

[INipkall]
type=peer ; have tried user and friend and it still doesn't work
;dtmfmode= ; have tried all settings and I can't get it to work - gave up and took it out
canreinvite=yes ; have tried it without this
quailfy=yes ; have tried it without this
host=66.54.140.46 ; have tried using voiper.ipkall.com also
nat=yes ; have tried it both as yes and as no

[jay]
user=friend
username=jay
secret=XXXXX
host=dynamic
mailbox=1001
nat=yes


Still there's no DTMF tones *OR* voice registered on my end. I've used the console (asterisk -rvvvvv) to monitor and within 3 seconds of the caller speaking, the SIP channel says the caller hungup.

I'm really at a loss here. I can verify that voice and dtmf are working when my number is forwarded to one of my other providers, just not my home server, so I *KNOW* the
problem's on my end.

cc: IPKall Forum, IPKall Support (via email)
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