Hi, which version of asterisk are you running? Perhaps if you post your
extensions.conf and others related files you could get more accurate help.
If you answer a ringing phone and you can't answer the call, there you
could have a network or sip config problem, that means that the SIP packet
Hi Everybody, I'm having a little problem with asterisk CLI, after the
version 1.4.19 I'm not been able to see the CLI with colors anymore. I
have a ubuntu box with asterisk 1.4.21 installed and I don't know how to
enable the colors again. Of course I have the variable $TERM set to
: Ending VLDTMF digit '3'
The problem is that the application mapped in feature.conf it isn't been
triggered. I would appreciate any help, I have already googled to death
and I couldn't find anything. Thanks in advance.
Lucas Alvarez
I'm using asterisk 1.2.10
David Gagnon wrote:
Are you having this problem with the trunk?
-Message d'origine-
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À : Asterisk Developers Mailing List; Asterisk Users Mailing List
the functionality of user and peer types. Is this
documented somewhere? How can I know the status of an extension of type
friend? I hope someone could bring me some light about this issue.
Thanks in advance.
Lucas Alvarez
___
--Bandwidth and Colocation
Does someone know where I can get the last sip version? My Polycom
reseller doesn't have it and I need to enable the buddy for 14 contacts.
Thanks in advance.
Lucas Alvarez
Douglas Garstang wrote:
I've never seen that problem, and I've only ever used 1.2+ with
Polycom and buddies
of the firewall in the sipura LAN and forward it to the sipura device.
Let me know if this works.
Lucas Alvarez
Joseph wrote:
So far I have gathered that on my NAT (where asterisk server is) I have
to port forward UDP ports: 5060 and range 1 - 2 to my asterisk
server
But I'm still stuck how
I'm using www.widevoip.com with asterisk, it works great. They provide
us a VPN connection and a SIP user.
Regards.
Lucas Alvarez
Andre Courchesne - Consultant wrote:
Hi,
Anyone has good/bad experience with SIP providers in upstate NY? Any
recommendations of such provider who works great
Hi, you have to forward port 5060 udp and all the others udp ports that
asterisk uses for RTP, 1 to 2 by default, see rtp.conf.
At the sipura device in the server configuration you have to put the LAN
ip of your Asterisk box as sip proxy server and firewall's ip as
out-bound sip proxy.
