Hi guys has anyone tried using a Gaitronics SIP extension on the 3800. I have of course got Openstage IP phones working OK but this is giving me a sore head. I build the extension as a sip client & it does register OK according to the hg1500 logs. I can ring it from a system phone & it does ring but no voice paths. I cannot ring a system phone from the sip extension it just times out.I have all ports open on the network but am even having problems getting an Xlite soft phone working on it from my laptop. The Siemens how to guide makes it look simple but I am really stuggling here. New switch on latest release UK based.
Tek-Tips is the largest IT community on the Internet today!
Members share and learn making Tek-Tips Forums the best source of peer-reviewed technical information on the Internet!
-
Congratulations gkittelson on being selected by the Tek-Tips community for having the most helpful posts in the forums last week. Way to Go!
Hi Path 3800 & Gai-Tronics SIP extension
- Thread starter telbisjon
- Start date
Iamnothere
Technical User
Is the SIP phone configured in AMO SBCSU to use SBDSS1 and OPTTBL=10?
If so, then type this command: CHA-ZAND:OPTLOAD;
This command forces the system to reload all used OPT Tables. If you just recently added SIP Subscribers, chances arew that the OPT Table was not loaded into memory, causing a call to/from a SIP phone to have no speech path.
That should resolve the problem, IF everything else is configured correctly!
If so, then type this command: CHA-ZAND:OPTLOAD;
This command forces the system to reload all used OPT Tables. If you just recently added SIP Subscribers, chances arew that the OPT Table was not loaded into memory, causing a call to/from a SIP phone to have no speech path.
That should resolve the problem, IF everything else is configured correctly!
I know you are talking GaiTronics, but I had this problem when my IT department made me figure out how to set Cisco SIP phones working on my 4000 V5. In my case I believe there was a parameter called "Register Gateway" or something like that which needed to be turned on. I could get it to dial out and be dialed, but no speech path. It had something to do in my case with the outgoing gateway settings I think. There is a thread on here that's about a year old or so that talks about Cisco SIP phones on a 4000 and I closed it out giving out the info I found to work in that case.
Who knows - maybe you will see something in there this helps with!
Who knows - maybe you will see something in there this helps with!
I looked up the answer in the other thread. If any of this looks familiar maybe it will help.
Since it seems nobody here has a clue I'll share what I eventually found out using the OpenStage 40 SIP Administrators Guide - seems the parameter names are fairly standardized except under SIP settings - what Siemens refers to as the "Voice Gateway" IP is what Cisco calls the "Outgoing Dial Proxy" and the "Incoming Proxy". That will get you dial tone, but you must also put a tick mark in the "Register with proxy" box to be able to use the phone.
Since it seems nobody here has a clue I'll share what I eventually found out using the OpenStage 40 SIP Administrators Guide - seems the parameter names are fairly standardized except under SIP settings - what Siemens refers to as the "Voice Gateway" IP is what Cisco calls the "Outgoing Dial Proxy" and the "Incoming Proxy". That will get you dial tone, but you must also put a tick mark in the "Register with proxy" box to be able to use the phone.
- Status