[Asterisk-Users] 183 Session in Progress

[Radius]

Hi all,
 
From Cisco 7960 I made outgoing calls through 
Cisco AS5300 to PSTN by   exten => 
_,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,r). 150.11.131.2 is the Cisco 
AS5300 PSTN gateway.
 
7960 rings for the first 2 seconds, then display 
"Session Progress (183)" with no more rings while the phone at the other 
side of PSTN is ringing. However, calls can be answered and there is 
no problem for phone converation. The same problem happens on CIsco ATA186. 
However, it does NOT happen on Grandstream phones. It looks like the call setup 
problem is only for Cisco products.
 
Anybody knows what caused it and how can it be 
fixed??
 
Many Thanks.
 
Ben

[Asterisk-Users] CLI command

[Radius]

Hi all,
 
Here is a simple question. How can I know if a call 
is in pass-thru mode, i.e. * is not in the media path???
 
Thanks.
 
Ben

[Asterisk-Users] 481 Call Leg/Transaction Does Not Exist

[Radius]

Hi all,
 
Windows Messenger 4.6 behind NAT works fine 
with * for me, except the NOTIFY for MWI and voicemail. The NOTIFY 
message triggers a 481 error. How can I make it right? I am using *  
current stable release.
 
Thanks.
 
Ben

[Asterisk-Users] Asterisk in pass-thru mode

[Radius]

Hi all,
 
Below is what I did to run Asterisk in pass-thru 
mode:
 
sip.conf:
[general]
disallow=all
allow=ulaw
canreinvite=yes
 
For each channel, canreinvite=yes is enabled. No 
dial command has 't' option.
 
However, it seems that Asterisk still stay in the 
media path and bridge the 2 end points. Am I missing 
something??? 
 
 
sip*CLI> show 
channels    Channel  
(Context    Extension    Pri )   State 
Appl. 
DataSIP/5001-c60b  
(company1    
1   )  Up Bridged Call  
SIP/1234-faf1  SIP/1234-faf1  (company1   
5001 1   
)  Up 
Dial  
SIP/5001|20|r2 active channel(s)
sip*CLI> sip show 
channelsPeer 
User/ANR    Call ID  Seq 
(Tx/Rx)  Lag  Jitter  
Format192.168.1.101    5001    
257684717aa  00104/0  0ms  ms  
ULAW210.17.211.5 1234    
003094c2-fd  00104/00102  0ms  ms  ULAW2 active 
SIP channel(s)
 
Thanks.
Ben

[Asterisk-Users] Calls to Cisco PSTN gateway

[Radius]

Hi all,
 
A Cisco ATA186 configured with g711ulaw, NAT=yes 
and canreinvite=yes,  made calls through Asterisk to a Cisco 5300 gateway 
out to a PSTN line with errors as follows:
 
    -- Executing 
Dial("SIP/ata186-c1cf", "SIP/[EMAIL PROTECTED]:5060|30|r") 
in new stack    -- Called 29086988@110.100.231.2:5060Apr 15 
16:11:22 WARNING[1116941120]: chan_sip.c:2049 process_sdp: Error in codec string 
'ideo 0 '
Asterisk was configured with allow=ulaw. Any idea 
for this problem??
 
Thanks.
 
Ben

[Asterisk-Users] PSTN calls do NOT hang up

[Radius]

Hi all,
 
In my Asterisk setup, incoming calls through 
Cisco PSTN gateway to Asterisk extensions sounds work fine. All calls can be 
terminated properly after hangup. However, when calls were forwarded to 
voicemail, after recording & hangup the PSTN calls and cisco FXO 
port remained connected unless cisco port was manually shut/no shut. # key 
used to hang up the call did NOT help. Did anyone experience the same 
problem??
 
--
 
sip*CLI>
    -- Executing 
Answer("SIP/-0811b4b8", "") in new stack    -- Executing 
Wait("SIP/-0811b4b8", "1") in new stack    -- Executing 
VoiceMail("SIP/-0811b4b8", "u6917") in new stack    -- 
Playing 'voicemail/default/6917/unavail' (language 'en')    
-- Playing 'vm-intro' (language 'en')    -- Playing 'beep' 
(language 'en')    -- x=0, open writing:  
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: gsm, 
0x81254f8    -- x=1, open writing:  
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav49, 
0x80fb178    -- x=2, open writing:  
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav, 
0x811af70    -- Playing 'vm-msgsaved' (language 
'en')    -- Executing Hangup("SIP/-0811b4b8", "") in new 
stack  == Spawn extension (sip, 6917, 4) exited non-zero on 
'SIP/-0811b4b8'sip*CLI>
 
---

cisco#sh voice call
1/0/1  vtsp level 
0 state = S_CONNECTvpm level 1 state = FXOLS_CONNECT vpm level 0 state = 
S_UP
 
--
 
dial-peer voice 999 
voip destination-pattern 8... session protocol 
sipv2 session target ipv4:10.1.1.1:5065 session transport 
udp codec g711ulaw no vad!
exten => 
6917,1,Answerexten => 6917,2,Wait(1)exten => 
6917,3,VoiceMail(u${EXTEN})exten => 6917,4,Hangup
Thanks.
Ben

[Asterisk-Users] Voicemail retrieval from Cisco 7960

[Radius]

Hi all,
 
I installed a new Cisco 7960 running SIP. 
I can make/receive calls to/from other extensions and leave voicemails. The LED 
on the handset turned RED, indicating voicemail for 7960. When I pressed 
the message button, 7960 gave me a dial tone only. What should I do to 
configure 7960 for voicemail retrieval??
 
Thanks.
 
Ben