[Asterisk-Dev] final mgcp patch
Karl Putland
karl at putland.linux-site.net
Fri May 9 06:49:04 MST 2003
On Fri, 2003-05-09 at 07:19, Tycho Schenkeveld wrote:
> Hi Karl,
>
> Good to hear that you have the DG-104S as well! At least I know now that it
> should work with the hardware and that I'm just doing something wrong.
>
> I'll have a look at the dtmfmode for the boxes, maybe there's something
> going wrong there. The dialtone doesn't even disappear when I dial a digit,
> though. I tried finding you on #asterisk, but I'm in Holland so we live in
> different time zones.
>
> Could you tell me which firmware you're using with the DG-104S? I have
> Firmware 3.0B35-C,
I believe that is the same version I have.
> according to my colleague that's the latest he found on
> the D-link website. Boot PROM FW is 2.10-B22, we don't seem to be able to
> boot that one. I'll have a look at the hook flash settings, that might solve
> the problems, sometimes * and the DG-104 go out of sync and the 104 just
> keeps transmitting voice at 64kbps even though it has hung up! If you could
> tell me your timeout settings that would be really great too!
>
Attached is the reference that I used to reset the various timeout
values. You must use the telnet interface to do the provisioning of the
dilink. I had to do it MANY time before the dlink finally decided to
keep the settings that I was entering.
> The most important problem we have is that everything works great locally,
> but when we try to use it over the internet the speech channels don't open,
> or they only open one-way (eg. I could hear you but you could not hear me).
> But I believe that those issues are related to the routers, most sites have
> routers before the boxes and I think that even though they are in a DMZ
> configuration, they report their internal IP address to *, which passes it
> to the other box.
>
dlink <--mgcp--> * <--IAX over the Internet--> * <--mgcp--> dlink
This works for me here
> Anyway, thank you for your great work on the MGCP stack! I'll keep messing
> with it here. If you have any clues I'd really appreciate it!
>
Great. BTW you just missed me in #asterisk.
I'm in the US, Denver, CO and you were just a little early for me.
> Regards,
>
> Tycho Schenkeveld
>
> -----Oorspronkelijk bericht-----
> Van: asterisk-dev-admin at lists.digium.com
> [mailto:asterisk-dev-admin at lists.digium.com]Namens Karl Putland
> Verzonden: donderdag 8 mei 2003 18:46
> Aan: tycho at connectingcrew.nl
> CC: asterisk-dev at lists.digium.com
> Onderwerp: RE: [Asterisk-Dev] final mgcp patch
>
>
> On Thu, 2003-05-08 at 10:14, Tycho Schenkeveld wrote:
> > Hi Karl,
> >
> > I just got the latest version of * from CVS, and I noticed a lot of
> changes
> > in the MGCP channel, which we use here with DG-104S D-link voip gateways,
>
> I also have the DG-104S and used that for testing while making the
> changes.
>
> > those only do MGCP. However, with the latest version it doesn't recognise
> > dialled DTMF digits any more!
>
> That I don't understand. It works for me.
>
> > I don't know if this is caused by the patch,
> > but I just wanted to let you know.
>
> Thanks. Find me in #asterisk. I'll try to get issues resolved quickly.
>
> >
> > By the way, we do have lots of other problems with the release version,
> like
> > hangups that are identified as hookflash and not handled by the mgcp
> > channel, so that's improved for sure! I did see that it recognises them
> now!
> >
>
> Most likely the hu/hf issue is a provisioning issue in the dlink. When
> I updated the firmware on the one I have, it stopped recognizing hf.
> Changing the various timeouts for disconnect and debounce fixed my
> problems.
>
>
> --Karl
>
>
> > Regards,
> >
> > Tycho Schenkeveld
> >
> > -----Oorspronkelijk bericht-----
> > Van: asterisk-dev-admin at lists.digium.com
> > [mailto:asterisk-dev-admin at lists.digium.com]Namens Karl Putland
> > Verzonden: dinsdag 6 mei 2003 5:35
> > Aan: asterisk-dev at lists.digium.com
> > Onderwerp: [Asterisk-Dev] final mgcp patch
> >
> >
> > Well, with few complaints (all addressed I believe) and not seeming to
> > create any new issues with chan_mgcp, I hereby submit this as my final
> > patch (for now) and declare that it is ready for cvs (assuming Mark has
> > no objections).
> >
> > --Karl
> >
> > --
> > Karl Putland <karl at putland.linux-site.net>
> >
> > _______________________________________________
> > Asterisk-Dev mailing list
> > Asterisk-Dev at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> --
> Karl Putland <karl at putland.linux-site.net>
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
--
Karl Putland <karl at putland.linux-site.net>
-------------- next part --------------
show coding
Configuration for coding profile id 1:
Tx Coding = G.711 MU
Rx Coding = G.711 MU
Coding profile for voice
Tx VIF size = 1920 (bits)
Rx VIF size = 1920 (bits)
VAD = ENABLED
VAD threshold = Adaptive VAD
Playout nominal delay = 20 (msec)
Playout maximum delay = 120 (msec)
Adaptive Playout = ENABLED
Rate = 14400
DTMF Relay = ENABLED
Tone detect = ENABLED
Call Progress Tone detect = DISABLED
V.18 Tone detect = DISABLED
SS7 COT Tone detect = DISABLED
SF Sig Tone detect = DISABLED
EC = ENABLED
EC NL = ENABLED
EC NL Sens = 327
EC Tail = 16 (msec)
EC Freeze = UPDATE
EC Coeffs = NORMAL
Modem TX level = 0 (dB)
Modem CD threshold = 0
Modem no activity timeout = 0 (sec)
Silence detection time = 60 (msec)
Silence detection level = -50 (dB)
Fax debug level = 0
Caller ID Support = ENABLED
Resampling = DISABLED
EC Config = NLP_FIXED
NLP Confort Noise = 65486
Encapsulation = RTP
ggdbg>show tcid 0
show tcid 0
Mode: Switched xGCP
Pref Voice coding profile: 1
Telephony Interface Configuration:
Companding = Mu-Law
Gain (RX,TX) = (5,0)
Idle noise level = -6500 x .01 dB
Signaling Protocol: FXS Loop Start
FXS Loop Start Parameters:
Offhook Debounce: 50 msec
Onhook Debounce: 50 msec
Seize Detect: 100 msec
Originator Clear Detect: 1200 msec
Answering Party Clear Detect: 150 msec
CPC Wait: 200 msec
CPC Duration: 850 msec
Ring Id: Default (-1)
Caller Id Generation: ON
Dial Out Parameters:
Out Wait: 250 msec
Out Type: tone
Tone Out Off Time: 70 msec
Tone Out On Time: 70 msec
Tone Out Power: -130 x 0.1 dB (not configurable)
Echo Cancellation Parameters:
Mode: LINE
Output: HANDS FREE
Handsfree Spkr Gain: 0
Handset Speaker Gain: 0
V.18A Tone Detect Parameters:
Hangover: 0
Threshold: 0
Fraction: 0
Other Parameters
Call Limit: 120 sec
Endpoint id : aaln/1
Partial Digit Timer: 16 sec
Critical Digit Timer : 10 sec
Service Class : 5
Default Digit Map: 0T
Default Event List: hu,hd,ft,mt
Entity to be notified: dlinkgw@[172.17.0.253]:2427
ggdbg>
More information about the asterisk-dev
mailing list