With the former router it could (but we have all the ports opened!)Īny advice please? Don't you think there is something wrong at the PBXnSIP side? However it doesn't matter if you click any number, nothing happens, the auto attendant keeps talking and talking.ĥ) The PBXnSIP can not register with CallWithUS and other providers we have. Now with the CISO ASA box, with all the ports required correctly open, no MEDIA is transfered between PBXnSIP server and rest of extensions.Ĥ) If you call from outside (using a regular phone line) to our number, our PBXnSIP auto attendant answers correctly and you can hear all the different options you can dial. I can understand a FIREWALL problem if this was happening from an outsider extension trying to register to our internal PBXnSIP server, but the issue is internally!!!!Īnother thing that I don't understand is that when the PBXnSIP was under DMZ with the former router everything was working great. I would understand ports problems with outgoing and incoming calls, but what about internal calls? Why when one internal extension calls to another one, there is no sound? The phones ring, but no sound.ġ) All the SIP phones (internal extensions) register correctly to the PBX server (a computer in the same LAN)Ģ) All SIP phones (internal extensions) can ring to others (you can hear the ringing sound)ģ) When an extension calls to another one and this one answers, you can see the time ticking (so the call is taking place), BUT NO SOUND (the media ports?) We opened and forwarded the ports to the PBX IP as shown above. The PBX server is connected to the ASA box, and we are not setting any DMZ to it. Now we have installed an ADSL bridge modem and a CISCO ASA 5500 box hooked to it. We have a public static IP where we used to set the PBX under DMZ before we switched to the CISCO ASA (we used to use a WAG Linksys regular router). Is the PBX still running on a public IP address (routable, in the DMZ)? Maybe there something went wrong during the migration. This would mean that you dont have to explicitly set them up IMHO. AFAIK the ATA monitors the traffic and then makes a decision which ports should be opened for RTP. I first guess is that the ATA has problems detecting what ports should be open. There are no errors on the Call Logs in the PBXnSIP software.īefore we used to have the server under DMZ, but now we can not, so we have to set the ports manually.ĭo you think this is a PORTS issue? Any recommendation? In the CISCO box we already created the following Access Rules:Īnd the same were used to create NAT rules too. Also if you call from outside, you can hear the auto attendant, but when you dial any of the options offered, nothing happens, it keeps talking. All the extensions are successfully registered on PBXnSIP domain panel, and when dialing you can hear the destination extension ringing, but when somebody picks up the phone, no sound, however the call is counting the time as if we were talking. Everything is working except the fact that we have no sound when we dial to any internal extension our outside destination. We have replaced the router by a CISCO ASA 5500 box. Our PBXnSIP server has been working pretty good for more than one year so far, installed in a very stable Linux server and under DMZ in a home use Linksys WAG router.
0 Comments
Leave a Reply. |
Details
AuthorWrite something about yourself. No need to be fancy, just an overview. ArchivesCategories |