>>> On 3/25/2009 at 11:19 AM, in message <[email protected]>, "wtuben" <[email protected]> wrote: > Hello, all: > the detail: > 1. sipxezPhone side : ILBC 30ms > 2. eyebeam : ILBC 30ms > 3. they donnot have a firewall > > and when sipxezphone call eyebeam, the call setup, and sipxezphone here > the eyebeam, but eyebeam cannot hear sipxezphone! > Will the codec iLBC used by the sipxezphone and eyebeam is NOT the same?? > And I hope somebody will do the same operation to watch the "BUG" as above? > > > Thx > > > ------------------ Original ------------------ > From: "Tony Graziano"<[email protected]>; > Date: 2009年3月24日(星期二) 晚上10:13 > To: "wtuben"<[email protected]>; "sipx-dev"<[email protected]>; > Subject: Re: [sipX-dev] iLBC FAIL! > > >>>> On 3/23/2009 at 10:55 PM, in message > <[email protected]>, > "wtuben" <[email protected]> wrote: > > Hello everyone, > > > > when I use sipxezphone place a p2p call to the eyebeam , and their codec > > > only use ILBC, > > > > and the result is: eyebeam CANNOT hear sipxezphone, but the sipxezphone CAN > > > hear the eyebeam!! > > > > AND when all use g711a/u ,they hear with each other!! > > > > thx > > > > ___________________________________________ > > sipx-dev mailing list > > [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipx-dev > > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev > > > When you call P2P without going through the proxy, the phones both need to > support the same codecs. They also both should be on the same subnet and you > should trun your PC firewalls off on both machines for testing purposes. > > If the call is made through sipXecs (the proxy), the call should go through > assuming they both use codecs the other phone also has. > > HINT: Except for sipXbridge, sipXecs does not negotiate media to ring a > user, the only time media is negotiated via the proxy is when the > AutoAttendant or Media server (voicemail greeting, recording, playing) occur. > > I'm not sure what you are asking since you are making P2P calls and one > phone is no longer supported. For simplicity sake, please use updated and > documented phones or softphones for testing. (i.e. xlite or something
> current). > > If you are not calling via the sipxecs proxy, I'm not sure how anyone can > help you except suggest the above right now. 1. But the issue is still that the sipxezphone is no longer supported. 2. If the call isn't going through sipx its not in the way or managing the call. 3. sipx does not neotiate the media (codec). See #1 above. I'm not sure of the logic of your question, but would like to help. Try to keep the call within a "supportable" parameter. Ex: PC1 with xlite registered as line 221 calls PC2 with xlite registered as 222 by dialing it's three digit sipx line number. When you call peer-to-peer, the two pc's are not using the proxy as all. So make sure you are using a supportable softphone (see #1) and call via the sipx extension to the other phone (see #2). In either case, sipx does not negotiate the media, but by sending the call through sipx you can at least capture logging and see why it fails if it does. _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev
