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Jeff Donovan

2004-Feb-11 09:15 UTC

[Asterisk-Users] unable to open ../voicemail/context/exten/msg0000

greetings
i have sortof got this working.

When checking mail i get an error.

Unable to play message 
/var/spool/asterisk/voicemail/home/2204/INBOX/msg0000 (format GSM); No 
such file or directory

okay,...
here are the contents of the directory
/var/spool/asterisk/voicemail/home/2204/INBOX

-rwx------  1 root  wheel  906 Feb 10 23:23 msg0000.WAV
-rwx------  1 root  wheel  858 Feb 10 23:23 msg0000.gsm
-rw-r--r--  1 root  wheel  228 Feb 10 23:24 msg0000.txt
-rwx------  1 root  wheel   44 Feb 10 23:24 msg0001.WAV
-rwx------  1 root  wheel    0 Feb 10 23:24 msg0001.gsm
-rw-r--r--  1 root  wheel  228 Feb 10 23:24 msg0001.txt

* note the extension is not listed in the error not sure if that makes 
a difference.

configs available if needed.

-----------------------------------
jeff donovan
basd network operations
(610) 807 5571 x4
AIM  xtdonovan
fwd# 248217

Maciek Kaminski

2004-Feb-11 09:38 UTC

head link

[Asterisk-Users] Can't connect KPhone to asterisk

Anyone managed to make KPhone work with Asterisk?

For me it looks as if KPhone does not ACK transactions, i.e.:

KPhone --INVITE--> Asterisk
Asterisk --Trying --> KPhone
Asterisk --OK --> KPhone
KPhone doest not acknowlege. Asterisk keeps resending OKs, KPhone 
INVITES. Both timeouts.

By the way: KPhone offers PCMU, GSM, iLBC in INVITE, Asterisk answers 
with PCMU and PCMA with doest not seem to be correct as it should answer 
with subset of codecs offered(as far as I understood SIP RFC). Another 
issue that bothers me is that Asterisk seems to start media transmission 
as soon as it send OK not after it received ACK. Begining of 
conversation may lost this way, isn't it?

Asterisk and KPhone logs below:

---------------------------------------------------------------------------------------------
Asterisk log:
---------------------------------------------------------------------------------------------
Sip read:
INVITE sip:700@polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700@polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
Call-ID: 1239913767@192.168.0.3
Subject: sip:maciejka@192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski"
<sip:maciejka@192.168.0.3;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

11 headers, 9 lines
Using latest request as basis request
Sending to 192.168.0.3 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format GSM
Found description format iLBC
Capabilities: us - 12, them - 1030/0, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:4186 check_user: Setting NAT on 
RTP to 0
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:5277 handle_request: Check for 
res for maciejka
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:1128 find_user: Call from user 
'maciejka' is 1 out of 0
Looking for 700 in default
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:3572 build_route: build_route: 
Contact hop: "Maciek Kaminski"
<sip:maciejka@192.168.0.3;transport=udp>
list_route: hop: <sip:maciejka@192.168.0.3;transport=udp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Length: 0


 to 192.168.0.3:5060
    -- Executing Answer("SIP/maciejka-b4b6", "") in new
stack
We're at 192.168.0.2 port 15200
Answering with preferred capability 4
Answering with preferred capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Type: application/sdp
Content-Length: 153

v=0
o=root 3363 3363 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 192.168.0.3:5060
    -- Executing Festival("SIP/maciejka-b4b6", "Press 1 to heaven
press
2 to go to hell press 3 to disconnect.") in new stack
  == Parsing '/etc/asterisk/festival.conf': Found
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:318 festival_exec: Text 
passed to festival server : Press 1 to heaven press 2 to go to hell 
press 3 to disconnect.
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:395 festival_exec: Passing 
text to festival...
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:414 festival_exec: Passing 
data to channel...
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:424 festival_exec: 
Festival WV command


Sip read:
INVITE sip:700@polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700@polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
Call-ID: 1239913767@192.168.0.3
Subject: sip:maciejka@192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski"
<sip:maciejka@192.168.0.3;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

11 headers, 9 lines
Ignoring this request
We're at 192.168.0.2 port 15200
Answering with preferred capability 4
Answering with preferred capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Type: application/sdp
Content-Length: 153

v=0
o=root 3363 3364 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 192.168.0.3:5060
Retransmitting #1 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Type: application/sdp
Content-Length: 153

v=0
o=root 3363 3363 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 192.168.0.3:5060
  == Spawn extension (default, 700, 2) exited non-zero on 
'SIP/maciejka-b4b6'
Feb 11 17:12:37 DEBUG[180236]: chan_sip.c:1207 sip_hangup: 
find_user(maciejka) - decrement inUse counter
set_destination: Parsing <sip:maciejka@192.168.0.3;transport=udp> for 
address/port to send to
set_destination: set destination to 192.168.0.3, port 5060
Reliably Transmitting:
BYE sip:maciejka@192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK77cfdb1c
From: <sip:700@polimorfia>;tag=as3b0a9ff0
To: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
Contact: <sip:700@192.168.0.2>
Call-ID: 1239913767@192.168.0.3
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.0.3:5060

---------------------------------------------------------------------------------------------
KPhone log:
---------------------------------------------------------------------------------------------

SipClient: Sending: 17:18:40.048
--------------------------------
INVITE sip:700@polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700@polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
Call-ID: 1239913767@192.168.0.3
Subject: sip:maciejka@192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski"
<sip:maciejka@192.168.0.3;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

SipClient: Sending to '192.168.0.2:5060'
SipClient: Receiving message...

SipClient: Received: 17:18:40.168
---------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Length: 0


SipClient: Receiving message...

SipClient: Received: 17:18:40.182
---------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Type: application/sdp
Content-Length: 153

v=0
o=root 3363 3363 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

SipTransaction: Retransmit 1 (500)

SipClient: Sending: 17:18:40.551
--------------------------------
INVITE sip:700@polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700@polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
Call-ID: 1239913767@192.168.0.3
Subject: sip:maciejka@192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski"
<sip:maciejka@192.168.0.3;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

SipClient: Receiving message...

SipClient: Received: 17:18:40.568
---------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka@192.168.0.3>;tag=B62B188
To: <sip:700@polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767@192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700@192.168.0.2>
Content-Type: application/sdp
Content-Length: 153

v=0
o=root 3363 3364 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

SipClient: Receiving message...

asterisk users - Feb 2004 - unable to open ../voicemail/context/exten/msg0000

[Asterisk-Users] unable to open ../voicemail/context/exten/msg0000

[Asterisk-Users] Can't connect KPhone to asterisk