Thank you Melissa. I love the phone but the dial keypad is a little bouncy.
I was hoping for a more solid feel like on the analog PT390's or my quality
standard, the Nortel 9417CW.
Other than the MWI problem, I'd like more documentation on the configuration
paramters. I have found little online configuration documentation other
than very basic stuff on the Sayson website. I'd like to know about all the
things that can be modified in the Aastra.cfg and <mac addr>.cfg files. I
stumbled on the dialplan command which solved my first big issue with the
phone. I'm sure there are other things I would like to have access to. I
want to know about shared call appearances, busy lamps, volume control
parameters, etc., which are controlled by the config files.
Is there a document or additional config info you can send me? These phones
are going to knock the Polycoms and Grandstreams out of the water once the
Asterisk and other open source communities understand their quality and
flexibility. I'd like to be on the cheering squad!
Joe McConnaughey
----- Original Message -----
From: "Melissa Lee" <melissa.lee@sayson.com>
To: <kenn10@comcast.net>
Cc: "David Sayson" <frisketdog@hotmail.com>;
<asterisk-users@lists.digium.com>
Sent: Monday, August 22, 2005 4:30 PM
Subject: RE: MWI problems on 9133i
Hi Joe:
Thank you for your feedback.
MWI problem on 9133i is a known bug and will be fixed in the August
release. The August release will be available to the public shortly.
Please check the www.sayson.com web site periodically.
Sincerely,
Melissa Lee
-----Original Message-----
Today's Topics:
23. Aastra 9133i Phone and MWI (Joe McConnaughey)
Message: 23
Date: Mon, 22 Aug 2005 08:29:38 -0400
From: "Joe McConnaughey" <kenn10@comcast.net>
Subject: [Asterisk-Users] Aastra 9133i Phone and MWI
To: <asterisk-users@lists.digium.com>
Message-ID: <000b01c5a715$29f12af0$6501a8c0@JoesAthlon64>
Content-Type: text/plain; charset="iso-8859-1"
Hello -
I have just purchased an Aastra 9133i SIP phone for testing with
Asterisk. Its a little flakey but overall is a far superior phone to the
others in the $179 range.
I have an issue regarding the message waiting indicator. The phone does
not seem to respond to the "NOTIFY" command from Asterisk. Searching
archives seems to indicate that this was previously an issue on the 480i
telephone but that it got corrected. I have downloaded the lastest
firmware (1.2.1) onto the phone but no go. Below is an exerpt of the
SIP DEBUG from CLI.
I'd like feedback on whether this is an Aastra issue or an Asterisk
issue. Anyone have experience with these phones? It appears that
Asterisk sends the message five times. Perhaps it is awaiting a
response from the phone which never comes. In any case, the MWI never
activates.
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:2008@192.168.1.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK3b581d6c
From: "" <sip:@192.168.1.251>;tag=as77c9889f
To: <sip:2008@192.168.1.201>
Contact: <sip:192.168.1.251>
Call-ID: 4a4e5c1d173010e3027c5d226cac810b@192.168.1.251
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 43
Messages-Waiting: yes
Voice-Message: 1/0
(no NAT) to 192.168.1.201:5060
Scheduling destruction of call
'4a4e5c1d173010e3027c5d226cac810b@192.168.1.251' in 15000 ms
Retransmitting #1 (no NAT):
NOTIFY sip:2008@192.168.1.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK3b581d6c
From: "" <sip:@192.168.1.251>;tag=as77c9889f
To: <sip:2008@192.168.1.201>
Contact: <sip:192.168.1.251>
Call-ID: 4a4e5c1d173010e3027c5d226cac810b@192.168.1.251
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 43
Messages-Waiting: yes
Voice-Message: 1/0
<SNIP>
to 192.168.1.201:5060
Retransmitting #5 (no NAT):
NOTIFY sip:2008@192.168.1.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK3b581d6c
From: "" <sip:@192.168.1.251>;tag=as77c9889f
To: <sip:2008@192.168.1.201>
Contact: <sip:192.168.1.251>
Call-ID: 4a4e5c1d173010e3027c5d226cac810b@192.168.1.251
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 43
Messages-Waiting: yes
Voice-Message: 1/0
to 192.168.1.201:5060
Sayson Technologies Ltd.
210 - 1910 Quebec St
Vancouver, BC V5T4K1
Canada
Phone: 604.730.1842
Fax: 604.732.8726
*****************************************************************************
This email and any files transmitted with it are confidential material. They
are intended solely for the use of the designated individual or entity to
whom they are addressed. If the reader of this message is not the intended
recipient, you are hereby notified that any dissemination, use, distribution
or copying of this communication is strictly prohibited and may be unlawful.
If you have received this email in error please notify info@sayson.com and
permanently delete the e-mail and files.
*****************************************************************************
Walter Willis
2005-Aug-22 16:27 UTC
[Asterisk-Users] problem client sip (ser) to client sip (asterisk)
i am configure ser:
if (method=="INVITE") {
if (uri=~"sip:1[1-9][0-9]+@.*") {
rewritehostport("192.168.0.183:5080");
};
};
an asterisk:
sip.conf
; config Xlite
[1234]
;context=sip
context=from-ser
type=friend
auth=md5
username=1234
secret=chooseapassword
;fromdomain=sorcier.com.pe ; para prueba de ser -asterisk
callerid="First Extension" <1234>
host=dynamic
canreinvite=no
;disallow=all
;allow=gsm
;allow=ulaw
;allow=alaw
;and conexion the ser to asterisk
;
[ser-sip]
type=friend ; permitimos llamadas entrantes y
salientes. Usar peer si solo es MWI
context=ser-asterisk ; este es el contexto que usan las
llamadas entrantes
;host=sorcier.com.pe ; Este es tu hostname o IP del servidor SER
host=192.168.0.183
fromdomain=sorcier.com.pe ; este es tu SER_DOMAIN (nombre de dominio del SER)
;insecure=very ; Permite que las llamadas que viene del
SER pasen a Asterisk
insecure=yes
;mailbox=user@context ; esto es para listar las cuentas de voicemail
;i am copy the voip-info
and the file the extensions.conf
; Configuracion al servidor ser, para llamada de ida
[from-ser]
exten => _X.,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,tr)
[ser-asterisk]
; Ignora el d?gito 0
;ignorepat => 0
; conexion a un telefono sip
;exten => _0X.,1,Dial(SIP/${EXTEN:1},90,Ttr)
;exten => _0X.,1,Dial(SIP/${EXTEN},20,Ttr)
;exten => _0X.,1,Dial(SIP/1234,20,Ttr)
;exten => _0X.,1,Dial(SIP/1234@ser-sip,20,Ttr)
exten => _0X.,1,Dial(SIP/${EXTEN})
i am probe diferents combinations, but no work
debug with asterisk and view itis:
Sip read:
INVITE sip:1234@192.168.0.183:5080 SIP/2.0
Record-Route: <sip:192.168.0.183;ftag=78607191;lr=on>
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0
Via: SIP/2.0/UDP
192.168.0.185:5060;rport=5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar <sip:rbolivar@sorcier.com.pe>;tag=78607191
To: <sip:1234@sorcier.com.pe>
Contact: <sip:rbolivar@192.168.0.185:5060>
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91@192.168.0.185
CSeq: 3143 INVITE
Max-Forwards: 16
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 299
v=0
o=rbolivar 23151399 23151750 IN IP4 192.168.0.185
s=X-Lite
c=IN IP4 192.168.0.185
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 13 lines
Using latest request as basis request
Sending to 192.168.0.183 : 5060 (non-NAT)
Found peer 'ser-sip'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.185:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for 1234 in ser-asterisk
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar <sip:rbolivar@sorcier.com.pe>;tag=78607191
To: <sip:1234@sorcier.com.pe>;tag=as1ca211c4
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91@192.168.0.185
CSeq: 3143 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234@192.168.0.183:5080>
Content-Length: 0
to 192.168.0.183:5060
Sip read:
INVITE sip:1234@192.168.0.183:5080 SIP/2.0
Record-Route: <sip:192.168.0.183;ftag=78607191;lr=on>
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1
Via: SIP/2.0/UDP
192.168.0.185:5060;rport=5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar <sip:rbolivar@sorcier.com.pe>;tag=78607191
To: <sip:1234@sorcier.com.pe>
Contact: <sip:rbolivar@192.168.0.185:5060>
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91@192.168.0.185
CSeq: 3143 INVITE
Max-Forwards: 16
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 299
v=0
o=rbolivar 23151399 23151750 IN IP4 192.168.0.185
s=X-Lite
c=IN IP4 192.168.0.185
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 13 lines
Ignoring this request
Sip read:
ACK sip:1234@192.168.0.183:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0
From: rbolivar <sip:rbolivar@sorcier.com.pe>;tag=78607191
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91@192.168.0.185
To: <sip:1234@sorcier.com.pe>;tag=as1ca211c4
CSeq: 3143 ACK
User-Agent: Sip EXpress router(0.10.99-dev14-tcp (i386/linux))
Content-Length: 0
8 headers, 0 lines
Destroying call 'E059F9B8-5EE6-4B5C-8D34-D5E0034BED91@192.168.0.185'
Sip read:
INVITE sip:1234@192.168.0.183:5080 SIP/2.0
Record-Route: <sip:192.168.0.183;ftag=78607191;lr=on>
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1
Via: SIP/2.0/UDP
192.168.0.185:5060;rport=5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar <sip:rbolivar@sorcier.com.pe>;tag=78607191
To: <sip:1234@sorcier.com.pe>
Contact: <sip:rbolivar@192.168.0.185:5060>
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91@192.168.0.185
CSeq: 3143 INVITE
Max-Forwards: 16
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 299
v=0
o=rbolivar 23151399 23151750 IN IP4 192.168.0.185
s=X-Lite
c=IN IP4 192.168.0.185
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 13 lines
Using latest request as basis request
Sending to 192.168.0.183 : 5060 (non-NAT)
Found peer 'ser-sip'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.185:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for 1234 in ser-asterisk
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar <sip:rbolivar@sorcier.com.pe>;tag=78607191
To: <sip:1234@sorcier.com.pe>;tag=as5a7c3a50
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91@192.168.0.185
CSeq: 3143 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234@192.168.0.183:5080>
Content-Length: 0
to 192.168.0.183:5060
Sip read:
ACK sip:1234@192.168.0.183:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1
From: rbolivar <sip:rbolivar@sorcier.com.pe>;tag=78607191
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91@192.168.0.185
To: <sip:1234@sorcier.com.pe>;tag=as5a7c3a50
CSeq: 3143 ACK
User-Agent: Sip EXpress router(0.10.99-dev14-tcp (i386/linux))
Content-Length: 0
8 headers, 0 lines
Destroying call 'E059F9B8-5EE6-4B5C-8D34-D5E0034BED91@192.168.0.185'
Sip read:
0 headers, 0 lines
the client sip (SER) call to client sip (asterisk) and return error 404
WHAT IS THE PROBLEM??? OR HOW TO FIX THE ERROR??