SPA 502G one way audio strange problem

[tcieslinski]

Hi,

I have few 502G registered in CUCM (sip thir-party-basic). Everything is working fine but,

- when i call from 502G to PSTN Local Number (via PRI) it's ok

- when i call from 502G to GSM (via the same PRI) i can't hear far end, GSM phone hears me ok

- when GSM calls 502G everything is ok?

What's wrong ??

Can you help me

Regards,.

Tomasz

[David Carr]

Typically with one way audio and voip, the issue mainly lies in the firewall.  Your firewall is allowing sip traffic to come through and that builds the call after that the call uses the RTP protocol and those ports vary based on the provider.  I typically have the port range 10000 to 20000 on a trigger status so the firewall will not block the traffic coming in through it.  That explains why they can hear you and you can't hear them.  The PSTN line don't use the firewall to connect.

Hope this Helps.

[tcieslinski]

Hi,

Unofortunatelly ther is no firewall in this deployment - all devices have full IP access (no access list or fw)

Regards,

Tomasz

[David Carr]

Have you tried changing codecs on the phone to see if that alleviates the problem?  I would try that and see if that and let us know.

[tcieslinski]

Hi,

Yes I tried,  but it didn't help.

We have simple CUCM configuration with no transcoding, regions, etc. Phone is configured to use g711u or g711a.  We observed that one-way audio is not only for GSM phones. There are some numbers that you call form linksys that you can' t hear far end.

Mayby there is a bug in phone firmaware?

Regards,

Tomasz

[David Carr]

What Firmware do you have on your spa502g?  The latest is the 7.4.6 for the phones.

Here is the link for that  http://www.cisco.com/cisco/software/release.html?mdfid=282724649&softwareid=282463651 

Upgrade your firmware and let me know how it goes. 

[viki]

Tomasz, did you resolve this problem ? We have exactly the same. We are running SPA502G with 7.4.8a firmware na CME 7.1 with SIP.

[tcieslinski]

Hi,

We resolved that issue.

The problem was with MGCP Gateway and MTP configuration for DTMF

The following configuration has fixed the problem:

mgcp dtmf-relay voip codec all mode nte-ca

mgcp package-capability fm-package

Hope it will work for you.

Tomasz

[bodgeup]

Could you give a description of your setup? ie what devices are involved?

Thanks

JK

[viki]

I can call from my mobie to this number but I can’t call to PSTN. When SPA502G user dial my mobile, my mobile is ringing, when I press answer button connection automatically ends.

Phone is SPA502G with
Software Version:        7.4.8a   
Hardware Version:        1.0.2(0001)

Internal calls are working fine.

We have more than 30 SPA922 phones with exacly the same config and all is working, so this is not related to dail-peer od eny other config.

SPA502G is running on defaults settings. Only proxy and username configured.

Codec are as follow

Preferred Codec: 729a		Use Pref Codec Only: NO
Second Preferred Codec: 711a		Third Preferred Codec: 711u

My setup is:

VOICE REGISTER GLOBAL
=====================
CONFIG [Version=7.1]
========================
  Version 7.1
  Mode is cme
  Max-pool is 42
  Max-dn is 144

voice service voip
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
fax protocol cisco
h323    
sip     
  registrar server expires max 3600 min 120
!        
!        
!        
voice class codec 1
codec preference 1 g729r8
codec preference 2 g729br8
!        
voice class codec 42
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 5 g729r8
!

voice register dn  30
number 22xxxxx89
no-reg

voice register pool  30
id mac C89C.1D6D.88A0
number 1 dn 30
voice-class codec 1  (I tryed also with voice-class codec 42 - the same situation)

All other SPA922 use the same settings:

voice register dn  YY
  number 22xxxxxZZ
  no-reg

voice register pool  YY

id mac C89C.1D6D.

number 1 dn YY

voice-class codec 1