Hi all,
I can’t establish calls betewwn two SIP communicator ,i’m using asterisk 1.4 .that is at the one terminal can hear but other side can’t hear, and also some time can’t able to make call to another sip communicator…
Even i can’t recieve any error or warnings in my console.
plz do reply
read the sticky at the top of the users forum. then, if the NAT help etc there doesn’t do it for you, come back and post the information anyone will need to help you, i.e. configs for the devices concerned.
also enabling sip debugging will help.
type
CLI> sip debug
at the asterisk CLI for enabling sip debugging
Hi all,
Thanks for your reply,
I’m using sip communicator(in java that is intergrated with erp product ) and asterisk is interfaced with this.
i’m able to make calls between pingtel and Voip user,
and also i can able to make call from Sip communicator to pingtel or Voip phone.
but now i’m can’t make calls between 2 sip communicator… it mean i can able to make a call and receive… but i can’t hear the other person voice. but my voice he can able to hear…
some times i can’t able to make (Between 2 sip comm.)call also…
I’m using asterisk 1.4 versoin…
could u tell me any suggestions…
Regards,
nsthi
Have you tried the suggestions posted?
yes i tried…
i got following
<— Transmitting (no NAT) to 192.168.2.33:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.33:5060;branch=z9hG4bKc776b81de4276b4875c23da227ad5647;received=192.168.2.33
From: “Name” sip:409@192.168.2.205:5060;transport=udp;tag=10884088
To: “Name” sip:409@192.168.2.205:5060;transport=udp;tag=as12587e5e
Call-ID: 86acc7b6992df93d746697c005bcd3de@192.168.2.33
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 0
Date: Wed, 06 Dec 2006 09:56:08 GMT
Content-Length: 0
please tell me how can i debug it …
actually there is no command in sip debug…
so i used sip set debug…
but i can’t hear the other person voice. but my voice he can able to hear
Should be NAT traversal problem.
search this forum for the solution.
hi ,
i 've set nat=no in sip.conf…
is i need to set it as yes…
any other settings …plz let me know
hi if u r in free …could u come to chat yahoo or google…
i need to discuss it with more…
i don’t understand why you can’t post the relevant configs and setup descriptions. we’ve asked for your example configs. we’ve told you what you need to read. if you really want help, you need to help us help you.
hi …
this is the my sip cobfig file …
i 've listed just two samples here…
[502502]
type=friend
context=from-sip
secret=adaptime
username=502502
callerid=“502502” <502502>
host=dynamic
nat=no
canreinvite=yes
dtmfmode=rfc2833
restrictcid=yes
incominglimit=5
disallow=all
allow=ulaw
[501501]
type=friend
context=from-sip
secret=adaptime
username=501501
callerid=“501501” <501501>
host=dynamic
nat=no
canreinvite=yes
dtmfmode=rfc2833
restrictcid=yes
incominglimit=5
disallow=all
allow=ulaw
could u need more files??
so they are on the same network as each other ? and the same network as the Asterisk server ? are they soft phones ? if so, do you have a firewall on the machine that’s running the software ?
you have canreinvite set to yes on them, do you want Asterisk to step out of the media path ?
hi,
i’m using same network as asterisk server and my PBx agent and also my users… …like within office with 20 PCs.
In asterik server + pbx agent machine the firewall is not running…
I’m really don’t know the real purpose of canreinvite and Nat…
please let me know if u need more info.
-nsthi
Plz do reply…
I’m using same network for asterisk server and sip communicator…
I’m able to make a call &receive calls between IP phones and Pingtel and also mixed of this things…
but if i used sip communicators ,i can’t hear the other person voice but he can hear my voice…
my firwall is off and my sip.conf file nat=no
i’m using asterisk 1.4…
plz do reply
the fact is, you’re missing something. either the rtp ports are being used by another process, or are being blocked by another process, or you’ve got it configured incorrectly.
thanks for ur reply.
plz let me know , where i’ve to debug it.
and how can i identify… my rtp ports are not functioning…
-nsthi
thanks for ur reply.
plz let me know , where i’ve to debug it.
and how can i identify… my rtp ports are not functioning…
-nsthi
have you asked the developers of SIP Communicator about this ?
no, but asterisk-1.2 sever works well without any interference…the problem occur when trying to use asterisk 1.4 server…
pl;ease let me know …how can i set my sip or other conf file to work preperly.
i.e, nat,canreinvite…etc…
-nsthi
I entered the command "sip set debug ip "…
got the following console message…
what it mean…
SIP/2.0 501 Not implemented
plz do answer…
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘555391623fb573132740b37514343cc8@192.168.2.205’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.2.20:5060:
OPTIONS sip:192.168.2.20:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.205:5060;branch=z9hG4bK51e769c6;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.2.205;tag=as34327c46
To: sip:192.168.2.20:5060;transport=udp
Contact: sip:asterisk@192.168.2.205
Call-ID: 24a058fc77ce99140c8de1fb4c77398b@192.168.2.205
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Mon, 18 Dec 2006 11:03:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<— SIP read from 192.168.2.20:5060 —>
SIP/2.0 501 Not implemented
Via: SIP/2.0/UDP 192.168.2.205:5060;branch=z9hG4bK51e769c6;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.2.205;tag=as34327c46
To: sip:192.168.2.20:5060;transport=udp;tag=-1791357446
Call-ID: 24a058fc77ce99140c8de1fb4c77398b@192.168.2.205
CSeq: 102 OPTIONS
Content-Length: 0