X Lite

Hi,

I have configured x-lite on two systems using asterisk server(as proxy).
Now when i try to call from one extension to another, “Call failed : not found” is displayed.

and on running sip show peers command,following messgae is displayed.In this it doesn’t show port 5060,is it ok??

actel55*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
1003 (Unspecified) D 5060 Unmonitored
1002/1002 192.168.3.31 D 29228 Unmonitored
1001/1001 (Unspecified) D 0 Unmonitored
1000/1000 192.168.3.12 D 54446 Unmonitored

Plz help!!

rgds
Vibhor

please, mention the context where your sips live
show us your extensions.conf (.ael?) with this context

SIP extensions lie in sip.conf file as follows :

[general]

[1000]
type=friend
context=phones
host=dynamic

[1001]
type=friend
context=phones
host=dynamic

[1002]
type=friend
context=phones
host=dynamic

[1003]
type=friend
context=phones
host=dynamic

What about extensions.conf? :smiley:

there should be smth lke

[phones]
...
...
...
[other_context]
...
...

Hello all!

I have a similar problem with x-lite phones and 3CX VoIP Phone.
Here are my sip.conf and extensions.conf

sip.conf

[general]

[1000]
type=friend
context=phones
host=dynamic
nat=yes

[1001]
type=friend
context=phones
host=dynamic
nat=yes


extensions.conf

[globals]

[general]
autofallthrough=yes

[default]

[incoming]
exten => 500,1,Dial(IAX2/guest@misery.digium.com/s)

[unknow]
exten => _X.,n,Answer()
exten => _X.,n,SayNumber(${EXTEN})
exten => _X.,n,Hangup()

[phones]
include => incoming
include => unknow

exten => 1000,1,Dial(SIP/1000,30,mg)
exten => 1000,n,Playback(vm-goodbye)
exten => 1000,n,Hangup()

exten => 1001,1,Dial(SIP/1001,30,mg)
exten => 1001,n,Playback(vm-goodbye)
exten => 1001,n,Hangup()

exten => 700,1,VoiceMailMain()

Even though an extension is defined in sip.conf, it needs to be in extensions.conf. The sip file tells asterisk physically about the sip connection, but the extensions file tells it what to do when people want to call that extension.

If you want to be able to call extension 1000, you need in your extensions something like:

exten => 1000, 1, dial(SIP/${EXTEN},15}
exten => 1000, 2, VoiceMail(${EXTEN})
exten => 1000, 3, Hangup()

That tells asterisk if somebody calls ext 1000, try and first ring the extension (which is defined in sip.conf). After 15 seconds go to voicemail.

You can get much fancier, but the point is you need explicit instructions in your extensions.conf for what should happen.

Dear tuvia!
I think you went to the theme = (

problem is that the command sip show peers can see all of my phones with the status Unmonitored.

asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 1000/1000 192.168.77.67 D N 60481 Unmonitored 1001/1001 192.168.77.67 D N 36474 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]

how do I fix?

Add this into the [default] section of extensions.conf.

exten => _100X, 1, dial(SIP/${EXTEN},15}
exten => _100X, 2, VoiceMail(${EXTEN})
exten => _100X, 3, Hangup()

Dear tuvia!

The fact that you offer me no help.
Actually in my opinion is not related to my problem. Calls within the system running smoothly. Only one problem. Sip show peers command returns:

asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 1000/1000 192.168.77.67 D N 60481 Unmonitored 1001/1001 192.168.77.67 D N 36474 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]

The status of the phones Unmonitored, but I want to be Monitored. I think we have any option described in the file sip.conf

Monitored has nothing to do with your problem.

I did not see your extensions.conf before. try moving the phones into default.

Ok!

I changed the context in sip.conf for the phone to default. And update the settings of my extensions.conf. We will see that this happens. But I think everything will remain unchanged.

I meant to move the [phones] code in extensions.conf to the [default] section.

I understand you =) and do as you said

Dear tuvia!

I did everything as you said! And I was right that nothing has changed. See what I did:

sip.conf

[code][general]

[1000]
type=friend
context=default
host=dynamic
nat=yes

[1001]
type=friend
context=default
host=dynamic
nat=yes
[/code]

extensions.conf

[code][globals]

[general]
autofallthrough=yes

[default]
exten => _100X,1,Dial(SIP/${EXTEN},mg)
exten => _100X,2,Playback(vm-goodbye)
exten => _100X,3,Hangup()
[/code]

sip show peers

asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 1000/1000 192.168.77.67 D N 65480 Unmonitored 1001/1001 192.168.77.67 D N 61456 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]

I think that there must be something changed in the config sip.conf. Probably not enough of a choice.

To complete the picture, giving sip show peer 1000 and 1001

sip show peer 1000

* Name : 1000 Secret : <Not set> MD5Secret : <Not set> Context : default Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : 2865 Insecure : no Nat : Always ACL : No T38 pt UDPTL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 192.168.77.67 Port 65480 Defaddr->IP : 0.0.0.0 Port 5060 Transport : UDP Def. Username: 1000 SIP Options : replaces replace Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none) Auto-Framing : No 100 on REG : No Status : Unmonitored Useragent : 3CXVoipPhone 3.1.6288.0 Reg. Contact : sip:1000@192.168.77.67:65480;rinstance=d61755e2b86841b0 Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs

sip show peer 1001

* Name : 1001 Secret : <Not set> MD5Secret : <Not set> Context : default Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : 2810 Insecure : no Nat : Always ACL : No T38 pt UDPTL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 192.168.77.67 Port 61456 Defaddr->IP : 0.0.0.0 Port 5060 Transport : UDP Def. Username: 1001 SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none) Auto-Framing : No 100 on REG : No Status : Unmonitored Useragent : X-Lite release 1103d stamp 53117 Reg. Contact : sip:1001@192.168.77.67:61456;rinstance=d1da666677d40ecc Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs

I draw attention to the “Status”

I think the problem with sip.conf?

You do not understand what monitor means, it has nothing to do with making calls. SIP peers are always unmonitored, monitor means watching the call log details. Normal use a peer would never show monitored. Look elsewhere for the problem.

Dear tuvia.

Thanks for your reply, I did not know that they are always in this state and I thought it strange. =)

thanks

SIP peers are always unmonitored, monitor means watching the call log details. Normal use a peer would never show monitored

Unmonitored means that in sip.conf, for the the friend/peer no “qualify” option is specified, try add “qualify=yes”, do a “sip reload” and you should see a change in the status field, if the friend is registered you’ll see the time it needs for Asterisk to reach the friend.

This is how I defined a sip friend for one of our Grandstream phones:

[Marco.Bruni]
type=friend
context=internals
host=dynamic
secret=******
qualify=yes
busy-limit=1
call-limit=2

This is the result of the command “sip show peer Marco.Bruni”, when the sip friend is registered to Asterisk:

* Name       : Marco.Bruni
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : internals
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 2
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : 1169
  Insecure     : no
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 192.168.200.182 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: Marco.Bruni
  SIP Options  : (none)
  Codecs       : 0x8000e (gsm|ulaw|alaw|h263)
  Codec Order  : (none)
  Auto-Framing:  No
  Status       : OK (25 ms)
  Useragent    : Grandstream GXP1200 1.1.6.16
  Reg. Contact : sip:Marco.Bruni@192.168.200.182:5060;transport=udp

Cheers.

Marco Bruni
www.marcobruni.net

hi mbruni

You really helped me solve my problem.
Many thanks you!

+1 to you… :wink: