Extensions Unreachable???

reynolwi

Joined
May 5, 2008
Messages
100
Likes
0
Points
0
#1
Im using elastix 1.0-17 and i have 3 grandstream gxp2000 phones. One phone works perfectly with no problems. I can call in and call out with no problems. The other 2 phones act like they arent even there. If i try and call the extension assigned to them i get the unavailable message, but yet i can call from them to the working phone and it rings. The phones show they are registered but i cant call them.

Any ideas?
 

rejil.rajan

Joined
Apr 8, 2007
Messages
154
Likes
0
Points
0
#2
Give the command sip show peer < extension of the phone that is not working >

* Name : 200
Secret : <Set>
MD5Secret : <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : 200@default
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 0
Dynamic : Yes
Callerid : "Name of the user" <200>
Expire : 3432
Insecure : no
Nat : No
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : IP Address of MachinePort 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 200
SIP Options : (none)
Codecs : 0x1c000c (ulaw|alaw|h261|h263|h263p)
Codec Order : (ulaw,alaw,h263,h263p,h261)
Status : Unmonitored
Useragent : Grand Stream
Reg. Contact : sip:200@<IP Address of the Phone>:5060;rinstance=98c2f50b705fde55

If the IP address field is correct then this shows that the phone is registered. If you could sent the exact log when making a call to the device, I could tell you the exact issue
 

reynolwi

Joined
May 5, 2008
Messages
100
Likes
0
Points
0
#3
Ok here is the information:

This extension, 20301 (shown below) works perfect. You can call in, and call out to this extension.
elastix*CLI> sip show peer 20301
elastix*CLI>

* Name : 20301
Secret : <Set>
MD5Secret : <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : 20301@default
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 50
Dynamic : Yes
Callerid : "device" <20301>
MaxCallBR : 384 kbps
Expire : 2534
Insecure : no
Nat : Always
ACL : No
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 10.25.18.80 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 20301
SIP Options : (none)
Codecs : 0xc (ulaw|alaw)
Codec Order : (ulaw:20,alaw:20)
Auto-Framing: No
Status : OK (45 ms)
Useragent : Grandstream GXP2000 1.1.6.16
Reg. Contact : sip:20301@10.25.18.80:5060;transport=udp


These extensions, 30301 and 30302 (shown below) only let you call out from them. If you try and call the extension from a phone it rolls over to voicemail instantly like the phone isnt on.

elastix*CLI> sip show peer 30301
elastix*CLI>

* Name : 30301
Secret : <Not set>
MD5Secret : <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : 30301@default
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 50
Dynamic : Yes
Callerid : "device" <30301>
MaxCallBR : 384 kbps
Expire : 1767
Insecure : no
Nat : Always
ACL : No
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 10.25.18.82 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 30301
SIP Options : (none)
Codecs : 0xc (ulaw|alaw)
Codec Order : (ulaw:20,alaw:20)
Auto-Framing: No
Status : OK (25 ms)
Useragent : Grandstream GXP2000 1.1.6.16
Reg. Contact : sip:30301@10.25.18.82:5060;transport=udp




elastix*CLI> sip show peer 30302
elastix*CLI>

* Name : 30302
Secret : <Set>
MD5Secret : <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : 30302@default
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 50
Dynamic : Yes
Callerid : "device" <30302>
MaxCallBR : 384 kbps
Expire : 1600
Insecure : no
Nat : Always
ACL : No
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 10.25.18.81 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 30302
SIP Options : (none)
Codecs : 0xc (ulaw|alaw)
Codec Order : (ulaw:20,alaw:20)
Auto-Framing: No
Status : OK (26 ms)
Useragent : Grandstream GXP2000 1.1.6.16
Reg. Contact : sip:30302@10.25.18.81:5060;transport=udp
 

rejil.rajan

Joined
Apr 8, 2007
Messages
154
Likes
0
Points
0
#4
Hi

Can you send me a log of the call that is being made from the working phone to the not working phone

Please set the verbose as 3 and paste the log
 

reynolwi

Joined
May 5, 2008
Messages
100
Likes
0
Points
0
#5
Ok i have no idea what happened, but i changed a different extension and now it works. I was messing with extension 50202 and changed the fax extension from freepbx to disabled and now for some reason it works. It still shows that the secret is not set for 30301 but it now says 30302 has the secret set.
 

rejil.rajan

Joined
Apr 8, 2007
Messages
154
Likes
0
Points
0
#6
If you see the logs properly you might be able to see if the device is forwarding the calls to another extension, or if it is kept on DND. Please also ensure the same extensions are not part of any other application in Asterisk. Deleteing a recreating the extension should resolve the Not Set issue
 

Members online

No members online now.

Latest posts

Forum statistics

Threads
30,915
Messages
130,920
Members
17,595
Latest member
feparra121
Top