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Solved SNOM710

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network52

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I just setup 3CX Cloud-based but I am having issues with calling with my snom710 phones, both phones are connected and but can not call each other. this is the error "[CM503003]: Call(C:24): Call to <sip:[email protected]:5060> has failed; Cause: 408 Request Timeout/INVITE from local: but I can call the softphone with the snom710. When I dial *777 it connects.
I have tried several times same error. I need help I can't figure what the problem is.
 
Last edited:
Hi network52,

Thank you for posting your issue on the community site.
First of all, please check if the PBX delivery audio on both extensions are enable. To make sure that this option is enable you can go to extension settings => option tab => troubleshooting. Next, please let us know what version are you on (OS, Version and service pack) and how the phones are connected on the 3CX by Stun, manual provisioning, RPS or SBC. Did you try to open the appropriate ports, disable alg etc? Please try to run the 3cx firewall checker client Application on the following link https://www.3cx.com/blog/voip-howto/3cx-firewall-checker-client/ if you do not use SBC and let us know.
 
Hello GiannosC_3CX,

  1. PBX delivery audio on both extensions are enabled on both phones
  2. I'm using the cloud base which I guess is Linux version and is up to date
  3. The phone are connected with RPS if I try Stun it doesn't connect.
  4. I don't get any error when I run the firewall test on the PBX, everything shows OK
  5. when i used the 3cx firewall checker client Application it failed.
what should i do next?
 
Hi network52,

Please let us know the error of the firewall checker client Application. If it possible please send us a screenshot with out any sensitive information (blur the region of your public IP etc...).
 
Also , please let us know what RTP Audio ports do you use for the two remote extensions?
"Management console => extension => snom710 => Phone Provisioning => Local RTP Audio Ports Start & End "
 
Local SIP Port of Phone: 5060
Local RTP Audio Ports Start: 14000
Local RTP Audio Ports End: 14009

Below is a detailed from the checker

Trying to resolve address of proxy FQDN:'xxxxxxxx.3cx.us'
Server IP has been resolved: 'xxxxxxx:5060'->xxxxxx:5060
Start test, v.1.0.4
STUN resolution result: addr = xxxxxxxx:5091
Extension '101' has been unregistered 408
3CX Phone System Server is unreachable using UDP transport. (408 Request timeout)
Trying to use TCP transport.
Extension 101 is registered
Calling *777
Connection with *777 established
WARN: Contact host is not PBX IP: XXXXXX->10.128.0.38
Possible causes:
1.SIP ALG or a SIP Proxy has been detected between this computer/network and the target 3CX PhoneSystem. This will cause problems.
Check with the firewall or SIP ALG/Proxy documentation on how to resolve this problem. In most cases SIP ALG should be disabled.
2.Your 3CX PhoneSysem Server might have a Dynamic Public IP address. 3CX PhoneSystem MUST have a Static Public IP.
(A Public IP Address that does not change)
3.You might be using a service like Dyn Dns and the DNS Record has been updated.
Dyn Dns is not supported and you need to configure your DNS correctly if you plan to use 3CX Phone System with an FQDN.
4.3CX PhoneSystem's border firewall might not have the correct Port Forwarding rules configured.
Please launch 3CX Management Console, go to Settings > Network> Firweall Checker> Run Firewall checker.
This will give you an exact list of what ports you would need to open. If the test fails, you would need to login to your firewall and open / port forward the required ports.
Audio port is 9026
ERROR: Echo test - The call has been unexpectedly terminated by PBX
Possible causes:
1. Firewall(NAT) is not working properly with calls initiated by remote devices.
2. Network failure
WARN: Contact host is not PBX IP: XXXXXX->10.128.0.38
Possible causes:
1.SIP ALG or a SIP Proxy has been detected between this computer/network and the target 3CX PhoneSystem. This will cause problems.
Check with the firewall or SIP ALG/Proxy documentation on how to resolve this problem. In most cases SIP ALG should be disabled.
2.Your 3CX PhoneSysem Server might have a Dynamic Public IP address. 3CX PhoneSystem MUST have a Static Public IP.
(A Public IP Address that does not change)
3.You might be using a service like Dyn Dns and the DNS Record has been updated.
Dyn Dns is not supported and you need to configure your DNS correctly if you plan to use 3CX Phone System with an FQDN.
4.3CX PhoneSystem's border firewall might not have the correct Port Forwarding rules configured.
Please launch 3CX Management Console, go to Settings > Network> Firweall Checker> Run Firewall checker.
This will give you an exact list of what ports you would need to open. If the test fails, you would need to login to your firewall and open / port forward the required ports.
Audio port is 9026
 

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Hi network52,

First, please check if the SIP SLG is enabled.
Next, please check the RTP ports of both extension are opened and not conflict between them and then that ports are forwarded to each internal IP phone.

For instance:
Ext 101 => Local SIP Port of Phone: 5064
Local RTP Audio Ports Start: 14000
Local RTP Audio Ports End: 14009
Public IP Address YY.YYY.YYY.YYY port forward to Ext-101 phone internal IP Address XXX.XXX.XXX.XXX

Ext 102 => Local SIP Port of Phone: 5065
Local RTP Audio Ports Start: 14010
Local RTP Audio Ports End: 14019

Public IP Address YY.YYY.YYY.YYY port forward to Ext-102 phone internal IP Address XXX.XXX.XXX.XXX

Also please make sure those ports are configured on cloud PBX under extension settings.
Note** The ports must be different on both extensions. Please find below screenshot.
upload_2017-3-22_10-1-30.png

Finally, please launch 3CX Management Console, go to Settings > Network> Firweall Checker> Run Firewall checker.
This will give you an exact list of what ports you would need to open on the 3CX cloud base system. If the test fails, you would need to open the required ports from your firewall.

Also, please visit the following links for more information.
Required ports: https://www.3cx.com/ports-used-3cx-phone-system-v14-v15/
Disable SIP ALG: https://www.3cx.com/docs/firewall-router-configuration-voip/#h.vbntni6x8gau
Configuration for remote extension: https://www.3cx.com/docs/firewall-router-configuration-voip/#h.qibmgjnjn5ns
and https://www.3cx.com/3cxacademy/videos/basic/configuring-ip-phones/
 
GiannosC_3CX,
Is working now i changed

Ext 102 => Local SIP Port of Phone: 5065
Local RTP Audio Ports Start: 14010
Local RTP Audio Ports End: 14019

on one ext. I did just that and changed provisioning method to (STUN-remote) for both ext.

Trying to resolve address of proxy FQDN:'XXXXX.3cx.us'
Server IP has been resolved: 'X.X.X.X:5060'->X.X.X.X:5060
Start test, v.1.0.4
Extension 103 is registered
Calling *777
Connection with *777 established
Audio port is 9084
Echo call - 600 packets received
Audio port is 9084
Calling *888
Waiting for Callback from *888
Answering call from *888
Connection with *888 established
Audio port is 9088
Callback - 606 packets received
Audio port is 9088

thank you
 
Hi network52,

I am glad to hear the issue is resolved and thank you for update the post.:)
 
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