Hello,

I’m trying to send a tif file, using Fax for Asterisk and the call is
executed, but when I get the reINVITE with T.38 data, the local server
doesn’t recognize that we have this capability and sends a 488 message.
These are the logs:

<--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --->
INVITE sip:1234567@10.0.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8
From: ;tag=gK0d817deb
To: “Fax” ;tag=as0ddeacb5
Call-ID: href=”mailto:74ca1e4e3e86a1b873428773477e201f@yyy.yyy.yyy.yyy”>74ca1e4e3e86a1b873428773477e201f@yyy.yyy.yyy.yyy
CSeq: 1785 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed
Contact:
Supported: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Length: 303
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 218 7126 IN IP4 xxx.xxx.xxx.xx8
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xx7
t=0 0
m=image 6202 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:262
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv

<------------->

Hi ,
Is it possible to have two meetme room in asterisk 1.6 which each one have a different language? I mean, one room the annoucement is in Portuguese an another in english?
Today I can change over the sip.conf and it is valid for all room.
regards!

Att,

Flavio Roberto Miranda

MSN:flaviormiranda@hotmail.com
Skype: flaviormiranda

We have a queue that agents log into through the dial plan. Extension
Sip/101 logs in as Agent/101

We have ‘ringinuse = no’ in the queues.conf file.

The issue is that when Ext 101 is on a ‘non queue’ call (they placed a
call, someone called their DID, etc) they still receive queue calls.

Is there a way to stop this from happening?

-Matt

Hi!

I’ve configured LDAP to read both users and extensions from LDAP server.
However, I’m experiencing problems with state tracking. Previously when
using static files, I was able to map extension number with channel
state using:

[sip_phones]
….
exten => 100,hint,SIP/user
exten => user,hint,SIP/user
..
rest of the dialplan
…

Thus when someone called the user, hint SIP/user showed channel state as
BUSY and I was able to use call limits etc. Now I’ve added this line to
[sip_phones]:

switch => Realtime/@

My hints, and call limits as well, stopped working. I’ve tried to move
hints to LDAP (which would be ideal situation for me), setting
AstPriority to “hint” but I don’t think they are event fetched. So the
question is… I’m I doing something wrong or it’s just impossible to
use those two solutions (hints + LDAP) together?

PS. I’m using Asterisk 1.6.2 if it helps with anything.

smime.p7s

Hi,

I have some problem with my provider. While the sip registration is
successful, i intermittently encounter problem in dialing out. I receive 603
Declined error in my Sjphone client. The asterisk log shows line is
busy/congestion.

Appreciate if help or direction can be provided.

Thanks.

CK