[Asterisk-Users] Sipura line 1 outgoing voice problem?

[Matt McIntyre]

I am experiencing this same problem and was wondering if anyone has come
to a resolution.  I have contacted Sipura but have not heard any
response yet and am having trouble determining for sure whether the
problem resides with Asterisk or the Sipura.  As I have noticed that
there are many users on the list who use the Sipura unit without this
problem (and even a fellow with one unit that worked and one that did) I
think the Sipura must be suspect.
 
Thanks,
 
Matt
 
 
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
cveazey at blackhillsfiber.com
Sent: Wednesday, March 17, 2004 11:18 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Sipura line 1 outgoing voice problem?
 

__________________ 
Back in January I started having a problem with my Sipura (and there was
at least one other on the list with the same problem) that if I answer
an incoming call (via X100P) on line 1 of my Sipura, the caller cannot
hear any voice from the internal extension.  If the internal user puts
the external user on hold (via flash hook) and returns, both directions
of audio are fine.

Line 2 never has had this problem.  For the meantime, I switched the
internal phones so that my wife's favorite phone is line 2 and I told
her to not pick up with line 1.  Not a very permanent solution :)

NAT is not an issue as the Sipura and * are on the same network.  Is
anyone else having this problem?  It looks like other people are using
Sipura (I saw one user with 30 of them ?!) and am surprised that nobody
else is complaining about this problem.  I am willing to step through
some sip debug if anyone is interested in the output.

* version: Asterisk CVS-02/08/04-22:22:57
Sipura firmware: 1.0.31 (just upgraded tonight to see if the problem
would go away)

Relevent config sections:

--8<--  sip.conf  --8<--

[cordless1]
type=friend
username=cordless1
secret=xxx
host=dynamic
context=cordless1
dtmfmode=info
mailbox=1234
canreinvite=no
disallow=all
allow=alaw

[cordless2]
type=friend
username=cordless2
secret=xxx
host=dynamic
context=cordless2
dtmfmode=info
mailbox=1234
canreinvite=no
disallow=all
allow=alaw


-- Chris
_______________________________________________ 





I had the exact same problem with a Mediatrix 1102....doing a flash hook
brought both sides of the conversation together.  I found out that my
sip.conf file had GSM as the first priority codec and the 1102 doesn't
support GSM.  I kept that the same but put a "disallow = gsm" statement
in my sip entry for the 1102 so g.711ulaw would be the first negotiated
codec.  That fixed the problem. 

VZ 
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