cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
1741
Views
0
Helpful
10
Replies

Not able to make outgoing calls using DX650

Ajay Wadhwani
Level 1
Level 1

Hi, 

I have a DX650 phone which was working perfectly fine till last week. This phone is installed at a SRST site. All of a sudden, outgoing external calls from the DX650 stopped. I am able to make internal calls and am able to receive calls (internal and external). However, outgoing external calls are not working. 

Below are the steps carried out from my end in order to understand the issue: 

1. Changed the handset from a DX650 to a CP-7821 phone. Phone registered and incoming and outgoing calls (internal and external) calls are working perfectly fine. 

2. Changed the CSS profile of the DX650 phone to other profiles (like receptionist, normal users). Phone registers but only incoming and outgoing (internal) are working. Outgoing (external) calls are not working.

3. Changed the CSS profile of the DX650 from the site to HO configuration and the DX650 phone registers and incoming and outgoing calls (internal and external) calls are working perfectly fine. 

4. Changed the phone to another DX650 phone and with the same CSS profile. Same result as Point 2. Outgoing (external) calls are not working.

Could you kindly explain what seems to be the issue and what needs to be done in order to correct this issue? 

10 Replies 10

Shri2
Level 1
Level 1

hi,

Can you provide the call Flow for this issue.

Did you see the call hitting on GW/Cube?  if yes please provide debugs from remote site gw/cube and HO gw/cube to compare the working/nonworking calls. If you dont see call on gw/cube then i would recommend to do DNA on CUCM to see whats happening.

Hi Shrikant,

 We were running the "debug voip vtsp session" command in the voice gateway to get the calls the call hitting from DX-650 and you can see in the attached log (LOG_01).Attaching here the output from CP-7821 configured with same extension and CSS as it was in DX650.( LOG_02). Calls are going from CP-7821 in this case.

 Advice me , is there any firmware compatibility play in between.

 Regards,

 Rinchuraj  

I can also see calls are hitting gateway. I can see setup request sent by gw then i believe we got disconnect from telco. Unfortunately above log doesn't shows much info to understand the cause of failure.

Is it a CAS controller or CCS controller. if its a CCS , please run "debug ISDN q931 and debug voip ccapi inout". Also which protocol is used on gw ? SIP/MGCP/H323? we may need respective protocol debug as well. please share above logs for working and non working calls. 

Hi, 

 As mentioned, I had taken the logs from the voice gateway.Please find the attached file.

Protocol using in Gateway : H323

Regards,

 Rinchuraj 

Raj,

As Mikolaj mentioned try to add bearer cap speech in PRI port.

Below link also explains the same :

https://supportforums.cisco.com/document/13051/outbound-calls-out-pstn-gateways-fail-cause-code-i-0x84ef-protocol-error-unspecified

Thanks Shrikant,

 Issue resolved.......

 Regards,

 Rinchuraj 

Glad to hear that its working.

Shrikant N

Rate useful comments

I believe you were trying to make a call from DN ending with 5951 and then DN ending with 5955 (or these are other calls being set up the same time).

Can you see the difference? If 5955 is your DX phone, then for this one you can see as below (call reference 0x0CC2/0x8CC2)

Transfer Capability = Unrestricted Digital

For other calls you can see (for example call with reference 0x0CC1/0x8CC1):

Transfer Capability = Speech

Try to change it as I suggested in my first post by setting what is below on the voice port that connects to PSTN.

voice-port 0/1/0:15

 bearer-cap speech 

Regards,

Mikolaj

Thanks Mikolaj Moryto,

 Issue resolved......

 Regards,

 Rinchuraj 

Mikolaj Moryto
Level 1
Level 1
Hi,
My suspect is HO's CSS gives you different way of sending calls out. I mean you reach different patterns which send calls through different gateways/trunks.
How is your site connected to PSTN? Do you have ISDN line or SIP trunk? It may be helpful to see your config and some debugs.
If you do use ISDN, try to use this command on your voice-port which you connect to the PSTN:
bearer-cap speech
Regards,

Mikolaj

Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: