[asterisk-users] Problem with Asterisk 1.4 and Linksys Spa941/962

[Stefan Schmidt]

Hello,

Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1
with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2.
Libpri and dahdi is only for dahdi dummy cause of the meetme function.

After the upgrade we had the problem that some Linksys spa941 phone at
one location could not dial out. incoming calls to the phones works
without any problem, but outbound the phone hangs up the call after its
connected to the other side. Sip debug shows me the following scenario:

-> invite
<- 407 proxy authorisation
-> ACK
-> Invite with auth header
<- 100 trying
<- 183 session progress with sdp header (correct c and m header)
<- 200 OK with same sdp header
-> ACK
-> BYE
<- 200 OK

this also happens on music on hold or playback and when trying to bridge
2 channels.

this only happens on one location and several (>1000) clients with the
same phone had no problem.

also a snom360 and xlite could dial out without any problem in the same
network.

After we had downgrade to 1.2.32 everything works fine again on these
phones.

my question is, had there been a big change in sip.conf or codec
handling which cause this problem, cause i used the same sip.conf just
adding notifyringing=yes, limitonpeers=yes and allowsubscribe=yes.


Here is my sip.conf with one client:

[general]
context=incoming
realm=softpbx
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
useclientcode=yes
defaultexpirey=3600

vmexten=voicemail
disallow=all
allow=alaw
allow=ulaw
allow=gsm

;qualify=no
;canreinvite=no

musicclass=default
language=de
useragent=ipanlage
callevents=yes
nat=yes
rtcachefriends=no
rtupdate=no
rtautoclear=no
ignoreregexpire=yes
amaflags=omit
canreinvite=no
subscribecontext=outcust
limitonpeers=yes
allowsubscribe=yes
notifyringing=yes

[xxx]
type=friend
context=outcust
nat=yes
qualify=yes
secret=yyyy
username=xxx
callerid="bla bla"
accountcode=xxx
disallow=all
allow=alaw
allow=ulaw
allow=gsm
host=dynamic

best regards

steve smith


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Mit freundlichen Grüssen
-- 
Stefan Schmidt
Sysadmin/VOIP // voip at sil.at // Tel 059944-2440//
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