ghannelais
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Posts posted by ghannelais
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In my case, it's a Trixbox install.
The Asterisk is a Trixbox 2.2 installation, no zaptel devices (ie, for timing).
Dave
Hi Dave
Trixbox here too (v. 1.2.3) with Zaptel board and no problem with choppy sound.
Asterisk on a Linux server
Uplink on a Windows server
SIP phones are Linksys SPA942 on all work stations
This is my Trixbox trunk settings in case it could help:
allow=ulaw
canreinvite=yes
context=from-trunk
disallow=all
host=dynamic
nat=yes ;very important. uplink won't work without this
secret=xxxx ;define your own password as in uplink
type=friend
username=xxxx ;make sure this is same to trunk name
Good luck,
Georges
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It seems the DTMF does not work.
Same problem here.
Uplink 1.30 (registered) works fine with our Asterisk server but we are unable to use the Digital Recepcionist, nor the Voice Mail as DTMF does not pass-thru.
For a SIP-SKYPE -> SKYPE-SIP call, it seems DTMF tone is sent back to the caller who pressed the key and not forwarded to the called party.
Please NCH try to solve this point asap or at least to explain the situation (many users have already mentionned the problem but so far no feedback).
The Uplink software is pretty good and it is really disapointing that the basic DTMF is not functioning!
Georges
Mexico City
Ringing time
in Uplink Skype to SIP Adapter
Posted
Trixbox + Linksys SPA-942 phones + Uplink driver for Skype
1- Skype to Skype call (direct, Uplink desactivated):
If called party does not answer, ring lasts 60 seconds and then Skype Voice Mail is activated (which is normal as I set the Advanced Voicemail settiing to 60 seconds).
2- Skype to Skype call (through Uplink)
If called party does not answer, ring lasts 30 seconds and then I get the Skype busy signal.
3- Normal call from Extension A to Extension B on the Asterisk system (no Skype, neither Uplink involved):
If called party does not answer, ring lasts as long as programmed in the Asterisk server (120 seconds in my case)
Can somebody confirm if the Uplink driver forces the disconnection after 30 seconds when no answer ?
These 30 seconds are not long enough for permitting the Asterisk server to hunt correctly the extensions to ring.
Any idea how to extend this time limitation?