|
Clive, cool - winter is getting quite near ova here... Well, how would I find out what is happening - I mean how do I know what * is connecting with to net2phone. "...They have their own proprietry protocol..."I thought it was because of the G723.1 codec and passthrough - but the I must take the voice prompts way. :-) (Didn't thought that it'll cause a problem - just the warnings and notices but continue still...) Thank you for that tip. "...For G723.1 passthrough, you just allow it..."------------------------------------------------------------------- So that is in "sip.conf" [general] disallow=all; allow=G723; allow=ulaw; allow=alaw; allow=gsm; (some text later) [net2phone] (some text) canreinvite=yes; (some text) ------------------------------------------------------------------- Sources for net2phone: http://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20Net2phone http://www.voip-info.org/tiki-index.php?page=Net2phone PS - I do get a frame error about expecting 4 getting 256 when * is trying to initiate to call through to net2phone device MAX IP-10 through the net2phone network - could be that protocall you were talking about or have I completely missed the plot? Kind Regards Etienne [EMAIL PROTECTED] wrote: Etienne, howzit I am not 100% sure about this, but Net2phone do not always use standard SIP as the protocol. They have their own proprietry protocol as well, so perhaps your phone is trying to talk on the proprietry protocol. For G723.1 passthrough, you just allow it, and it should work fine, as long as you do not try playing any voice prompts to the channel. good luck. regards Clive ===================== Phone I.T. http://www.phonehome.co.za On 13 Apr 2005 at 8:52, Etienne Pretorius wrote:Hello all, I came a cross a problem yesterday that I don't quite know how to solve. I am trying to use * to connect to net2phone, and have a net2phone MAX IP-10 connect to net2phone. From the settings on http://www.voip-info.org/ it was easy to get asterisk to connect to the network - acting like a net2phone device/user. Anyway the problem arose when attempting to call the MAX IP-10 device through the net2phone network. They seem to be using the G732.1 codec. I have in my settings in sip.conf allow=G732.1 or what ever flavour of the like and still I can not talk to the two devices. I googled a bit and came across the fact of * being able to do a pass through - well I was not successful and this subject is either simple or not well documented. The devices are using SIP and there is a bridge initiated, but there is no audio and no voice being passed through... I have tried connecting as the receiving device a GrandStrem Budge Tone-100 and still no luck. So all that I am inquiring is has anyone successfully done a pass through and if so can someone please guide me through some of the settings. I have set the [net2phone] with a canreinvite=yes - that a post on a forum also suggested, and that also did not work. On a separate issue: When the Grandstream Budge Tone-100 is connected on the internal network then the audio and the voice in both directions work fine. But when the device is connected on a separate network - ie on an other ADSL line, then the device doesn't send voice packets although is receives packets. I have opened up IPTABLES, to allow udp 5060 and udp 10000:20000 in both directions on any interface and the problem still persists. (SIP phone: Grandstream Budge Tone 100 connects to * and the call is answered by a Softphone X-Lite with all the codecs enabled. As far as I can tell thy both are "speaking" with a G711 codec ULaw/ALaw). So can anyone please give me a guideline or some advise on where to look to solve the problem. -- Kind Regards Etienne _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users |
_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
