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Fault Occurred in U1910 SPC600 when testing call transfer feature

Publication Date:  2013-07-23 Views:  40 Downloads:  0
Issue Description
User A is a SIP user registered to the U1910 SPC600. User B is an external user. The Cisco Voice Gateway AV5400 connects the IP PBX U1910 to external phone lines. When user A calls user B and then wants to transfer the call to another user C the call is suddenly dropped. The same scenario happens when user A wants to put the call on hold, or when testing the call forwarding feature. 

On the wireshark capture the following disconnect code is present:

SIP/2.0 422 Session Timer too small
Alarm Information
SIP/2.0 422 Session Timer too small
Handling Process
1. Verify the call rights of user A, B, C  - Done, rights for external calls existed

2. Capture packets using wireshark  -  Done, verify SIP signaling and SIP headers

3. Verify user session timer settings on the U1910   -   Done

4. Verify user session timer feature on the Cisco gateway   -   Done

5. Verify the wireshark capture and see who is disconnecting the call and with what cause   -   Cisco Gateway disconnects the call with SIP/2.0 422 Session Timer too small

6. Increase the session timer setting for user A on the U1910 and make test call again – Increased to 3600 s and test succeeded
Root Cause
1. Make a simple call from user A to the external user B

2. Make a simple call from user A to the external user C

3. Make a simple call from user B to user C

4. Verify IP Phone session timer for user A

5. Verify user A session timer parameter on the U1910 PBX

6. Verify the session timer settings on the Cisco AV5400 Gateway

7. Increase user A session timer setting on the U1910

8. Make test call scenario
Suggestions
If a call fails try to follow these steps:

1. Verify the calling rights of the users
2. Verify connectivity of the users
3. Verify firewall for VoIP calls
4. Make a wireshark capture to see who disconnects the call and identify the disconnect cause reason
5. Verify SIP settings on the Huawei IP PBX,  and also compatibility of call features with third party gateways

END