Voice Problem in SIP

Asterisk Asterisk Support
1

Hi All
I have generate the call connectivity between 2 linux system one server & one client on sip using Xlite . But iam not able to hear voice betwen client & server.

MY extension.conf file consist of

[sip]
exten => 1000,1,Dial(SIP/1000)
exten => 2000,1,Dial(SIP/2000)

My SIP.conf consist of

[1000]
type=friend
host=dynamic
defaultip=192.168.1.48
username=1000
secret = password
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip
callerid=“user1”

[2000]
type=friend
host=dynamic
defaultip=192.168.1.40
username=2000
secret = password
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=2000 ; Mailbox for message waiting indicator
context=sip
callerid=“user2”

,What all changes i have to make Kindly help

2

２　linux. 1 is server, 1 is client. where is the other client? on you client , you can hear the welcome message. but you can’t make calls. because defaultly one pc only has one sip port 5060 for sip.

telecomchinasourcing.com

3

Client is here only its for testing .Do i have to change the port on client side or server side .How can i change,& to which port number

4

When i gave the command
sip show peers
Name/username Host Dyn Nat ACL Port Status
2000/2000 192.168.1.33 D 5060 Unmonitored
1000/1000 192.168.1.48 D N 5061 Unmonitored
1000 is the server & client is 2000

5

Are you using NAT ?

6

Ya ,i am using NAT
i gave only this command

iptables -I RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT

7

Ya ,i am using NAT
i gave only this command

iptables -I RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT
Kindly tell me setting that i have to change in sip.conf & extension.conf
My sip .conf is

[code][1000]
type=friend
host=dynamic
defaultip=192.168.1.48
username=1000
secret = password
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip
callerid="Server"
nat=yes ; X-Lite is behind a NAT router
canreinvite=no

[2000]
type=friend
host=dynamic
defaultip=192.168.1.33
username=2000
secret = password
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=2000 ; Mailbox for message waiting indicator
context=sip
;allow=ulaw
callerid=“Client” [/code]

EXTENSION.conf

[sip] exten => 1000,1,Dial(SIP/1000) exten => 2000,1,Dial(SIP/2000)

8

Seems to me that you have to open some other ports, for RTP communication and messaging. I’m using these:

-A INPUT -p tcp -m tcp --dport 5060 -j ACCEPT -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT -A INPUT -p udp -m udp --dport 5036 -j ACCEPT -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT

If your server is behind a NAT and has an external IP which is used by the client in question you have to fill out sip_nat.conf.

nat=yes externip="the public IP" localnet=192.168.0.0/255.255.255.0 qualify=yes

9

I could find the file sip_nat.conf ,how can we do without using nat

10

hi,
Are your server and client machine is in same LAN. If your answer is yes then please change in sip.conf the line

nat=never

for example your sip file looked like this

[1000]
callerid="Server"
username=1000
secret = password
type=friend
host=dynamic
port=5060
nat=never ; X-Lite is behind a NAT router
qualify=yes
record_in=Adhoc
record_out=Adhoc
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip
canreinvite=no

and If your answer is no then,
[1000]
callerid="Server"
username=1000
secret = password
type=friend
host=dynamic
port=5060
nat=yes ; X-Lite is behind a NAT router
qualify=yes
record_in=Adhoc
record_out=Adhoc
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip
canreinvite=no

If it doesn’t work the please let me know

Regards,
Deepen

11

Ya its in the same LAN .but still not working ,when iam calling from client(2000)to server(1000) ,On the asterisk console it is showing the message like

Executing [1000@sip:1] Dial("SIP/2000-085f65c8", "SIP/1000") in new stack [May 12 09:25:14] WARNING[4508]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/2000-085f65c8' status is 'CHANUNAVAIL'

my sip .conf file is

[code][general]

externip=192.168.1.48
localnet=192.168.1.1/255.255.255.0
canreinvite=no ; force relaying

[1000]
callerid="Server"
username=1000
secret = password
type=friend
host=dynamic
port=5060
nat=never ; X-Lite is behind a NAT router
qualify=yes
record_in=Adhoc
record_out=Adhoc
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip
canreinvite=no

[2000]
type=friend
host=dynamic

defaultip=192.168.1.33
username=2000
secret = password
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=2000 ; Mailbox for message waiting indicator
context=sip
nat=never
qualify=yes
allow=ulaw
callerid=“Client” [/code]

12

HI again
Now if iam giving a call from server(1000)to client(2000) ,after getting connected i can hear my voice in server only but not on client side.I meant that only on server i can hear voice but not on the other end(client) .

localhost*CLI> sip show users
Username Secret Accountcode Def.Context ACL NAT
2000 password sip No No
1000 password sip No No

13

Hye again When iam dialling from client to sever iam getting this msg

May 12 10:55:55] NOTICE[8740]: rtp.c:788 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.1.48 == Spawn extension (sip, 1000, 1) exited non-zero on 'SIP/2000-085f65c8' -- Unregistered SIP '1000' -- Got SIP response 489 "Bad event" back from 192.168.1.48 -- Unregistered SIP '2000'
What i think is to transmit silence or “silence suppressionâ€

14

As the error says "Comfort noise support incomplete in Asterisk ". Asterisk does not support silence suppression. If you don’t want asterisk to complain then just turn it off in your phone.

15

Yes i did ,iam using XLite on both side ,but still not working

16

hi,

It’s seem that there may be firewall problem so, do one thing please stop your firewall using lokkit command for your RedHat Linux firewall and try out once.

Regards,
deepen

17

Ya i disabled Security level & SELinux on both client &server but issue is still exist.
Iam calling from (client(2000)192,168,1,33) to (server(1000)192,168,1,48) it showing this message again

[May 12 19:41:36] NOTICE[7373]: rtp.c:788 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.1.48 == Spawn extension (sip, 1000, 1) exited non-zero on 'SIP/2000-095b31f0'
Iam sending again the setting of sip .conf & extnsion.conf .Thanks for so much for helping me
SIP.conf

[code][general]

externip=192.168.1.48
localnet=192.168.1.1/255.255.255.0
canreinvite=no ; force relaying
port=5060
srvlookup=yes

[1000]
allow=all
callerid="Server"
username=1000
secret = password
type=friend
host=dynamic
nat=never ; X-Lite is behind a NAT router
qualify=yes
record_in=Adhoc
record_out=Adhoc
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip
canreinvite=no

[2000]
allow=all
username=2000
secret = password
type=friend
host=dynamic
nat=never
qualify=yes
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=2000 ; Mailbox for message waiting indicator
context=sip
callerid=“Client” [/code]

EXT.conf

[sip] exten => 1000,1,Dial(SIP/1000) exten => 2000,1,Dial(SIP/2000)

18

on server side IP is forty eight not 4

19

externip=192.168.1.48
localnet=192.168.1.1/255.255.255.0

why you give this, it’s totally useless. just remove it and that remaining 3 lines add separately in each user. But it not solve your problem.

Somebody may give you an solution.

Regards,
deepen

20

hi
Problem still occurring when i gave the command set rtp debug & called from server to client
I get this type of message

Sent RTP P2P packet to 192.168.1.48:10000 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.33:10000 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.33:10000 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.33:10000 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.48:10000 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.33:10000 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.48:10000 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.48:10000 (type 00, len 000160)

CLIENT to server

Sent RTP P2P packet to 192.168.1.48:10000 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.48:10000 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.48:10000 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.48:10000 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.48:10000 (type 00, len 000160)

Any further solution