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3869
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35
Helpful
15
Replies

My remote SIP phone don't receive calls

wrobynson
Level 1
Level 1

Hi everyone.

 

I have a problem: my SIP phone (3rd party) coudn't receive calls. It can make calls, bat I can't hear the other side.

voice register dn  6
 number 7275
 allow watch
 name 7275
 label TF4 7275
 mwi
!
voice register pool  69
 busy-trigger-per-button 2
 id mac 000B.000B.000B
 number 1 dn 6
 dtmf-relay rtp-nte sip-notify
 username 7275 password 0069x0000009999
 description GXP 2000 SJC
 codec g711ulaw
CME_3945#sh voice register pool  69
 Pool Tag 69
Config:
  Mac address is 000B.000B.000B
  Number list 1 : DN 6
  Proxy Ip address is 0.0.0.0
  Current Phone load version is Grandstream GXP2000 1.2.5.3
  Class of Restriction List Tag: default
    Incoming corlist name is PERFIL-5
  DTMF Relay is enabled, rtp-nte, sip-notify
  Call Waiting is enabled
  DnD is disabled
  Video is disabled
  Camera is disabled
  Busy trigger per button value is 2
  Description is GXP 2000 SJC
  keep-conference is enabled
  registration expires timer max is 3600 and min is 60
  username 7275 password 0069x0000009999
  Lpcor Type is local
  Lpcor Incoming is PSTNTrunk
  Lpcor Outgoing is

  Transport type is udp
  service-control mechanism is not supported
  registration Call ID is 1425b1bdf56d7bcf@10.132.8.45
  Registration method: per line
  Privacy feature is not configured.
  Privacy button is disabled
  active primary line is: 7275

  contact IP address: 10.132.8.100 port 5060

  conference admin: no
  conference add mode: all
  conference drop mode: never
  paging-dn: config 0 [multicast]  effective 0 [multicast]

Dialpeers created:

Dial-peers for Pool 69:

dial-peer voice 40080 voip
 corlist incoming PERFIL-5
 destination-pattern 7275$
 session target ipv4:10.132.8.100:5060
 session protocol sipv2
 dtmf-relay rtp-nte sip-notify
 codec  g711ulaw bytes 160
  after-hours-exempt   FALSE

The phone IP address is 10.132.8.45. But the created dial-peer to this extension (7275) point to 10.132.8.100

CME_3945#sh dial-peer voice sum | i 7275
40080  voip  up   up             7275$              0  syst ipv4:10.132.8.100:50

So, the calls to extension 7275 are forward to wrong IP 10.132.8.100. I can confirm this by debugging ip packet.

 

Can somebody help me, please?

 

 
1 Accepted Solution

Accepted Solutions

Do a debug ccsip message to see what information that is passed between the router and the phone to if possible see if there is something with that IP that is included in the registration.

My guess is that the phone has your outside router as the gateway and it is doing address translation for the traffic. This is the reason for why you see this IP in the dial peer. I would call this a network “problem” or at least at a minimum a network misconfguration on your part.



Response Signature


View solution in original post

15 Replies 15

I would remove Voice register dn as well as pool and check if dial-peer 40080 is getting deleted as well, if yes then re-add dn and pool. Alternatively, try to add new dn and pool and register this phone with that new dn to confirm if it's problem with only dn 7275 and dial-peer 40080. It's possible that CME might have not cleaned up old dial-peer when another device with IP 10.132.8.100 was registered to it in past. 

Also, make sure there is no real phone (another device you may not be aware of) with IP 10.132.8.100 using same dn 7275. You can do basic checks like ping to 10.132.8.100 or check ARP table on switch etc. 

Ok.
Deleting POOL and DN...

CME_3945(config)#do sh dial-peer voice sum | i 7275
40080  voip  up   up             7275$              0  syst ipv4:10.132.8.100:50
CME_3945(config)#
CME_3945(config)#no voice register pool  69
CME_3945(config)#
CME_3945(config)#no voice register dn  6
CME_3945(config)#
CME_3945(config)#do sh dial-peer voice sum | i 7275
! <-- nothink -->
CME_3945(config)#do sh run | i 10.132.8.100
! <-- nothink -->
CME_3945(config)#

Reconfig with new TAGs ...

voice register dn  58
 number 7275
 allow watch
 name 7275
 label TF4 7275
 mwi
!
voice register pool  78
 busy-trigger-per-button 2
 id mac 000B.000B.000B
 number 1 dn 58
 dtmf-relay rtp-nte sip-notify
 username 7275 password 0069x0000009999
 description GXP 2000 SJC
 codec g711ulaw

But...

CME_3945#sh dial-peer voice sum | i 7275
40118  voip  up   up             7275$              0  syst ipv4:10.132.8.100:50

This 'bastart' IP insists on appearing...

Topology:

topologia.png

 

The first time, the phone was on the internal network. But he didn't even register ...

Is there a way to delete this dial-peer automaticaly created?

 

Thank you.

Please provide bit more informations regarding what you are trying to achieve ?

 

 



Response Signature


As already mentioned, extension 7275 is registered with CME but does not receive calls.
This is because the dial-peer created for it has the wrong target IP.
The question is: how to fix this? 

From the Diagram, 7275 extension  is in a different  network and registered with the CME_3945 where 7271 and 7272 is registered . 

 

your problem is SIP phone (3rd party) couldn't receive calls. It can make calls, but I can't hear the other side.

 

if 7271 can hear you, and if its only 7275 who cannot hear, check your network reachability. its should be two way.

And you mentioned 7275 unable to receive call, you definitely need to check your network.

 

 

There is no network problem.

 

The problem is wrong IP (10.132.8.100) point to in created dial-peer (CME) for this phone (the correct would be 10.132.8.45).

Asper the network diagram 10.132.8.100 is  your  Nat IP. CME might be receiving the registration request from 10.132.8.100. You need to check you network.

 

 

 

 



Response Signature


 Both problem SIP phone (3rd party) couldn't receive calls and  It  can't hear the other side is due to the NAT IP 10.132.8.100.

 

Those dial-peer which you mentioned are virtual dial-peer which get created by CME  based on the extension configured. in your case CME might be getting registration request from your Nated IP. so CME create dial-peer with destination target 10.132.8.100

 

You need to Check you network configuration.

 

 



Response Signature


CME would automatically create dial-peer for each phone/extension, you can not delete it. Moreover, calls won't work without this dial-peer.

 

Looks like your phone IP 10.132.8.45 is getting NATted to 10.132.8.100. 

Check the default gateway on this phone and also trace the network path between phone and CME. If this phone is on internal network, then there should not be NAT router in the path, or even if it's there then exclude phone's IP (.45) from the NAT.

 

 

Any other ideas? 

what's ur CME ip address ?

 

my SIP phone (3rd party) coudn't receive calls. Receive call from Where ?

 

It can make calls, bat I can't hear the other side. calls to outside or internal extensions

 

 

 



Response Signature


I'm talk about only internal calls. Extension to extension...

 

7275 is a remote extension. But the calls to this extension are forwarded to another IP.

Do a debug ccsip message to see what information that is passed between the router and the phone to if possible see if there is something with that IP that is included in the registration.

My guess is that the phone has your outside router as the gateway and it is doing address translation for the traffic. This is the reason for why you see this IP in the dial peer. I would call this a network “problem” or at least at a minimum a network misconfguration on your part.



Response Signature


You're right!

pane.png

After a long time, I found the problem ...

Thank you, very much !