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 SkipVought on being selected by the Tek-Tips community for having the most helpful posts in the forums last week. Way to Go!

SIP one-way audio inbound

Status
Not open for further replies.

Telecomboy

Programmer
May 8, 2003
4,111
0
0
US
I am having an issue where the trunk is up and working, but inbound calls are getting one-way audio. Being basic edition I have not dealt with SIP on this. I have tried manipulating every setting on the IPO with the same results. The provider says they are sending an invite, but not receiving an acknowledgement back from the phone system. I have verified the following are disabled: SIP ALG is disabled, SCCP disabled (Provider told me to do this), SIP Inspect off. I am not sure where else to turn. Anyone else run into this and have a solution?
 
One way audio is 99.9% of the time the network blocking UDP ports or not routing them correctly. I would start there.

P.S. God have mercy on your soul doing SIP with Basic...

The truth is just an excuse for lack of imagination.
 
I know. Basic Mode is killing me. They claim 5060 is open all the way through. Is there other UDP ports that need to be open?
 
5060 is signalling, and I don't see anywhere in Basic mode where you can set the Audio RTP range.
Essential edition defaults to 46750 to 50750 so maybe try that range?
 
It probably depends on what version you are running as to what the RTP ports are because the default RTP ports changed at one point. What version are you running on the system?

For instance I have R9.1.12 I jumped in and the only reference to RTP ports in basic search:

Static Port Block
Use the RTP port range 49152 to 53246.

 
Oh by the way I sure hope you have a combo card in the system... only way to get VCM resources in basic mode as VCM cards are not supported and you need VCM resources for SIP trunks.

Also, if I remember right, delaying an auto attendant in basic with SIP (or PRI for that matter) is broken and does not work.
 
I do have a combo card in there. I did also see that RTP range associated with Static Port Block (it is R9.1), but I have it set to firewall blocking after running stun. I wasn't sure if that only applied if using the Static Port Block setting. I will make sure those RTP ports are open though. Thanks for the suggestions. If I get this thing working I will definitely post the results.
 
Just to follow-up: after doing a ton of testing the issue came down to the fact that the IPO was sending the internal ip address out on the invite and accept versus the external ip address. The proper fields were filled in on the IPO to send the external ip, it just didn't do it. I upgraded them to essential and it starting working immediately. Stupid basic edition.
 
Status
Not open for further replies.

Part and Inventory Search

Sponsor

Back
Top