[Asterisk-Users] Codecs and * pass through...

[Etienne Pretorius]

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


clive at sentechsa.com 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
>>
>>
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