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Hi All,<br />
I have acreated an extension(5678) &nbsp;in extensions_custom.conf in order to record voice message and play it again...Everything is working fine if you dial the extension number from another extension registered on the same Asterisk server...<br />
Now I have an openser server and need to dial this extension number and record the voice in order to play it back...The call is reaching the asterisk server but I'm getting No ROute to Destination message....Please note if I dial a normal extension from openser to asterisk server, the call works fine...<br />
Please letme know if there is any additonal configuration that I should do because i created a [general] peer in sip.conf as below but still getting The error of NO ROUTE TO DESTINATION<br />
<br />
[general]<br />
type=peer<br />
nsecure=very<br />
host=domain_name<br />
dtmfmode=rfc2833<br />
disallow=all<br />
context=domain_ip<br />
allow=ulaw
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Hi All,
I have acreated an extension(5678)  in extensions_custom.conf in order to record voice message and play it again...Everything is working fine if you dial the extension number from another extension registered on the same Asterisk server...
Now I have an openser server and need to dial this extension number and record the voice in order to play it back...The call is reaching the asterisk server but I'm getting No ROute to Destination message....Please note if I dial a normal extension from openser to asterisk server, the call works fine...
Please letme know if there is any additonal configuration that I should do because i created a [general] peer in sip.conf as below but still getting The error of NO ROUTE TO DESTINATION

[general]
type=peer
nsecure=very
host=domain_name
dtmfmode=rfc2833
disallow=all
context=domain_ip
allow=ulaw

No Route to Destination

Hi All,
I have acreated an extension(5678)  in extensions_custom.conf in order to record voice message and play it again...Everything is working fine if you dial the extension number from another extension registered on the same Asterisk server...
Now I have an openser server and need to dial this extension number and record the voice in order to play it back...The call is reaching the asterisk server but I&#39;m getting No ROute to Destination message....Please note if I dial a normal extension from openser to asterisk server, the call works fine...
Please letme know if there is any additonal configuration that I should do because i created a [general] peer in sip.conf as below but still getting The error of NO ROUTE TO DESTINATION

[general]
type=peer
nsecure=very
host=domain_name
dtmfmode=rfc2833
disallow=all
context=domain_ip
allow=ulaw

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.