New user help

General Help
1

I have version FreePBX version 2.11.0.38 Asterisk version 11.13.0 running on a Raspberry Pi. My current setup is 2 Cisco 7940G phones converted to SIP. I’m using voip.ms for sip service. The status in FreePBX shows the 2 Cisco phones as well as a softfone online. It also shows the trunk online and registered.

Now the problem. I can dial from the softpohne to either extension with no problem. I can dial from the softphone to an outside number with no problem. I can dial from Cisco phones to the softphone. I can also dial from the Cisco phones to Lenny with no problem. I can’t dial one Cisco phone from the other one. When I hit the dial button, I get the fast busy tone and the reorder message. The message in CLI shows chan_sip.c:23019 handle_response_invite: Received response: “Forbidden” from ‘“Anonymous;” :tag=asZe71186d’

Same thing if I try to dial out from the Cisco phone. I thought it may be a dialplan issue so I tried a couple with the same results. Here’s what’s currently loaded in the phone:

DIALTEMPLATE
TEMPLATE MATCH="*" Timeout=“5”/>
/DIALTEMPLATE

I’m not sure where to go with this. I have spent many evenings in front of Google and making changes but I’m not getting anywhere.

Any help would be greatly appreciated.

Thanks

2

It probably is a dialplan problem, but I don’t use Cisco phones so I can’t help there.

Try swapping the registration details from the softphone and one of the Cisco phones and see if the problem follows the phone or the extension #. If it follows the extension number, then it’s got to be a dial plan issue.

You haven’t told us your numbering plan, but if the softphone has a 4 digit extension (1001)and your Cisco phones have a 2-digit extension (44), you might try changing the length of the Cisco extension numbers to see if that makes a difference.

You should also check out potential Codec issues. Make sure that the phones support G711 and that Asterisk SIP settings lists G711 as the first supported codec. You might even make it the only supported codec just to see if that makes a difference.

3

Thank you for the reply! You can’t imagine how frustrating this is!

The basic softphone I’m testing with is Quote.com. I have assigned it a 3 digit extension as are the other two Cisco phones. I just confirmed that I can dial the softphone from either extension and I can dial either extension from the softphone. I can’t dial from the ip phones to each other or an outside call.

I checked the codec in the phone and it’s set to G711ulaw which is what asterisk is set to. I also unchecked all codecs in Asterisk other than ulaw. Same results.

I’m starting with a very basic dialplan:

<DIALTEMPLATE>
  <TEMPLATE MATCH="3.." TIMEOUT="0"/>
  <TEMPLATE MATCH="1" TIMEOUT="0"/>
  <TEMPLATE MATCH="500" TIMEOUT="0"/>
  <TEMPLATE MATCH="9..........." TIMEOUT="0"/>
</DIALTEMPLATE>

I don’t know if it makes a difference but when I dial from one extension to the other (ip phones), It rings about a half a ring before it goes to the fast busy and reorder message.