[Asterisk-Users] Weird issues with Asterisk@home 0.9

[Henry Devito]

> Hi,
>
> I am having a few issue withs Asterisk at home 0.9.
> 1) Setting up a sip connection, with voicemail to use with (eyebeam/X-pro)
> softphone. I can receive voicemail no problem and even in this revision
> the MWI seems to work correctly, though when i try to go to the message
> center, (*98) and enter my voicemail box number, it always gives me
> invalid password.
> If i now do this from a analog phone which is also connected to the
> system via a TDM400P card, i can get into the voicemail box no problem.
> Anyone have experienced this before?
>

What do you have the DTMF type set for in your SIP.conf and on the phone? 
It should be  RTP (RFC2833)

> 2) Another problem i came accross is when using the IAX2 extentions (using
> IAX phone), and trying to make a call i get nothing but static on the
> line, even between IAX <-> IAX and IAX <-> SIP, IAX -> PSTN..  This wasn't
> an issue in the 0.8 release where it seemed quite stable and the sound
> quality was even better then from the SIP phone.
>
>
> 3) The next problem i have encountered is that the webmail feature doesn't
> seem to work right, as in accepting user input to log in. It seems as if
> the mysql tables/inputs haven't been setup correctly. I have tried the
> webmail with multiple different userid's, and always get username/password
> invalid. Any ideas?
>
>
> 4) The last problem, though might not be specific to AAH, i am still
> experiencing a sound issue when calling the PSTN, where the
> connection sounds like the rushing of an ocean in the phone line, and
> a light echo every once in a while. I live in Canada, and use Telus as my
> Telco provider, so i am not sure what specifics i may be missing.
> I have totally rewired the house for asterisk, thus new lines everywhere,
> making sure they are not to close to powerlines, etc for static concerns.
> Also standard phones don't seem to have the problem.
>
> I should mention, this problem occurs both going through SIP (eyeBeam)
> aswell as a Analog phone (Siemens Gigaset SL30 connected via FXS port).
>
> Here is my zapata.conf:
>
> ;
> ; Zapata telephony interface
> ;
> ; Configuration file
>
> [trunkgroups]
> [channels]
>
> language=en
> context=from-pstn
> signalling=fxs_ks
> rxwink=300              ; Atlas seems to use long (250ms) winks
> ;
> ; Whether or not to do distinctive ring detection on FXO lines
> ;
> ;usedistinctiveringdetection=yes
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> hanguponpolarityswitch=yes ; Added to test hangups
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes ; was no
> echotraining=800
> rxgain=0.0
> txgain=0.0
> group=0
> callgroup=1
> pickupgroup=1
> immediate=no
>
> ;faxdetect=both
> faxdetect=incoming
> ;faxdetect=outgoing
> ;faxdetect=no
> channel=3
>
> ;
> context=from-pstn
> signalling=fxs_ks
> faxdetect=incoming
> usecallerid=yes
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=800
> group=0
> channel=4
>
> ;
> context=from-internal
> signalling=fxo_ks
> usecallerid=asreceived
> echocancel=yes
> echocancelwhenbridged=yes
> echotraining=800
> callerid=2892091
> group=1
> channel=1-2
>
> ;#include zapata-channels.conf
>
>
> Anyone have any ideas? Please let me know.
> Thanks
>
>
> S.
>
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