01-24-2004 09:41 AM - edited 03-13-2019 03:30 AM
Hi,
I am working with SIPH323 protocol converter. I have cisco SIP phones in my SIP network and cisco SKinny phones (H323) with CallManager in H323 network.
When I call from SIP to Skinny phone, it rings and gets connected also. But I dont hear anything (voice/audio) in skinny phone what I speak from SIP phone.
Similarly when I call from Skinny to SIP phone, I dont hear the ringing tone of SIP phone in Skinny phone, but both phones get connected and I dont hear any voice.
I would appreciate if anyone could tell me the possible solutions for this problem.
Thanks,
Balaji
01-29-2004 08:54 AM
I doubt whether the CCM supports SIP protocol , try with the links provided below
http://cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml
01-30-2004 10:54 AM
This is probably a routing problem. A common problem with H.323 gateways is the source address. H.323 advertise a source address that the far end is to send the RTP stream to. However, this address is not neccessarily the address of the closest interface and can in fact be an address that the remote site cannot reach. And if that is the case, the only error is a lack of voice. Calls will still ring and connect, because that uses TCP, but the RTP stream will be lost in the ether.
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide