– Executing [5556@xlitephone:1] Dial(“SIP/xlite1-00000000”, “SIP/xlite2”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/xlite2
– Got SIP response 482 “Loop Detected” back from 10.0.0.50:5060
– SIP/xlite2-00000001 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/xlite1-00000000’ status is ‘CONGESTION’
i need help pls
Configuration error. The host and port addresses for the X-Lite have been configured as those for Asterisk. At least one should be different for the X-Lite.
I’m assuming 10.0.0.50 is the Asterisk box. Otherwise you will have to provide detailed network topologies.
Thanks David for the reply,yes the IP for the asterisk server is 10.0.0.50,how do you mean by i have to configure another host and port for the xlite,kindly reply ASAP,am under pressure,thanks
here is the portion in sip.conf and extension.conf for the xlite phones:
ext.cnf
[xlitephone]
exten => 5555,1,Dial(SIP/xlite1)
exten => 5556,1,Dial(SIP/xlite2)
sip.conf
xlite
; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
;qualify=3000
context=xlitephone
host=10.0.0.50
username=mine
secret=mine
nat=yes
;allow=gsm
allow=ulaw
allow=alaw
disallow=all
You can only have one application using port 5060 on one IP address.
I’m a little confused as I thought X-Lite is Windows only and Asterisk is *nix only, so I can’t work out how you have both on the same address, but if you do have both on the same address, one will fail to bind to port 5060, so you will need to change the local port number in its configuration and the remote port number in the other’s.
The other possibility is that you have misunderstood the meaning of the hostname parameter. This is the host name of the device not the local host name. It can also be “dynamic”, if the device registers.
thanks david,am using the xlite on windows while the asterisk is on centos,isnt this arrangement possible?
ve changed host=dynamic now i get this eroor:
Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/xlite1-00000002’ status is ‘CHANUNAVAIL’
The phone hasn’t registered.
how i do sip show peers i see the phones there,how do i register the phones and new to asterisk
how do i register the phones,am new to asterisk
That’s an X-Lite question, not an Asterisk one.
If you are that new to Asterisk, you should read asteriskdocs.org/, the X-Lite user documentation, and RFC 3261, or a good summary of it.
You should have a sip.conf entry for each peer you are registering, please see my revision to your sip.conf below. Also, you shouldn’t need nat=yes on the peers if they are on the same lan as your asterisk server.
What version of xlite are you using?
[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
;qualify=3000
context=xlitephone
host=dynamic
username=xlite1
secret=mine
nat=yes
;allow=gsm
allow=ulaw
allow=alaw
disallow=all
[xlite2]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
;qualify=3000
context=xlitephone
host=dynamic
username=xlite2
secret=mine
nat=yes
;allow=gsm
allow=ulaw
allow=alaw
disallow=allhi Jbarth,thanks,am using xlite version 5
Setting up xlite with asterisk is pretty well covered here:
Thanks,that article was very helpful but i still have error:
Call from ‘203’ (10.0.0.2:5060) to extension ‘201’ rejected because extension not found in context ‘xlitephone’.
despite having the ext. in the extension.conf as shown below:
[general]
static=yes
writeprotect=no
[xlitephone]
exten => 55,1,Playback(demo-echotest) ; Let them know what’s going on
exten => 55,2,Echo ; Do the echo test
exten => 55,3,Playback(demo-echodone) ; Let them know it’s over
exten => 200,1,Dial(SIP/200)
exten => 201,1,Dial(SIP/201)
exten => 202,1,Dial(SIP/202)
exten => 203,1,Dial(SIP/203)
Its ok now,thanks
hi Jbarthle, how are u?do u ve any documentation how to add a yealink T-22Pand snom 320 phones to Asteriskpbx
The snom 320 should have all kinds of documentation on the internet about how to set it up. It is one of the more popular phones. If found this for the yealink: http://mbit.com.au/material/Yealink-T28%20User%20Manual-V51.0.pdf. That should get you going for that model.
Pls am having a little challenge,i added a polycom soundpoint P321 phone, the phone can receive calls from other phones but it cannot make calls, see below the sip.conf:
[400]
type=friend
context=esofties
dtmfmode=rfc2833
secret=mypassword
host=10.0.0.80(ip of the phone)
mailbox=10@phone
disallow=all
allow=ulaw
callerid=400
progressinband=no,
Any help pls