Tihomir, Thanks for the reply. I was quite surprised to see 30ms packetization time. and first thought it's a typo. But now I see (if I'm not mistaken) that FS really sends open logical channel with Alaw:30ms when opening logical channel during fast start (see attached file). This parameter is "display only" on my IPO. Is it possible to change it on the FS side? I think it should be somewhere in mod_h323 configuration. But I did not found any mod_h323 configuration parameters except ip addr. and port. By the way, I know that one can use different packetization times for the same codec, but I've never heard, that somebody really uses 30 ms for G711Alaw. Always 20ms. Thanks ans regards, Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Thursday, February 11, 2010 9:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working On Thu, Feb 11, 2010 at 4:18 PM, Nikolay Kondratyev <kond@nstel.ru> wrote: Tihomir, Thanks for help. I enabled fast start on IPO and I can hear voice now. But the ringback tone and voice appears to be wheezy, but I will investigate that tomorrow. set framing time for your codec to 30ms in IPO, also play with PI in alerting... set it to 2. Thanks again. Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Thursday, February 11, 2010 2:34 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working Nikolay, you are sending slow start with tunneling=true ?!?! It is not gong to work :) Please can you set fast start instead? Your call failed because there was no mediaControll channel negotiated at all... actually the call had to be aborted because wrong signaling .. but anyhow. Please on your IPO use FastStart with h245Tunneling=true... also, same settings on FS side as well (exclude h245 in setup as well). Frame 13 (277 bytes on wire, 277 bytes captured) Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst: Vmware_67:33:a7 (00:0c:29:67:33:a7) Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49 (172.23.22.49) Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port: h323hostcall (1720), Seq: 1, Ack: 1, Len: 223 TPKT, Version: 3, Length: 223 Q.931 Protocol discriminator: Q.931 Call reference value length: 2 Call reference flag: Message sent from originating side Call reference value: 0012 Message type: SETUP (0x05) Bearer capability Display 'Gornak Alexandr>2853' Calling party number: '5840' Called party number: '2853' User-user H.225.0 CS H323-UserInformation h323-uu-pdu h323-message-body: setup (0) setup h4501SupplementaryService: 1 item 1... .... h245Tunneling: True On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond@nstel.ru> wrote:
But I do recommenf to use SIP, since it's much better supported by FS.
/Peter
But SIP is poorly supported by IPO. Thanks and regards, Nikolay. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org