IPECS call drop out and one-way voice issues

[noobatron]

Hi guys,

I'm experiencing a lot of issues with the IPECs system. Users are experiencing intermittent call drop outs anywhere between 30 seconds and 5 mins.

Also, sometimes when the users think they calls has dropped, it really hasn't and the user is experiencing one-way voice (the called party can hear them but the user cannot hear the called party). Funny thing is, when the users puts the person on hold and takes them off, the call is fine again.

The system runs on a cisco switched network with port-based QoS set to trust DSCP markings from the phone and over the trunks.

Also, does anyone know where i can get a configuration manual for the IPECS system? I want to set the diffserve markings properly.

Thanks.

 

[doktor]

Normally this issue is seen due to:

1.
Too old software in the iPECS and the LIP phones.
2.
Network not dimmensioned good enough.

To check #1:
From any display phone connected to iPECS, press [TRAN/PGM] or [PGM] + 72.
The s/w version is displayed - please post it.
Browse to an IP phone's IP address to see the s/w level.
Please post it here.
Also write the number and type of modules and their s/w.
This can be seen in PGM102.

Also check that the default gateway is set up correctly.

///doktor

 

[noobatron]

Hi Doktor,

Thanks for the prompt reply. Here are the details of the system and versions:

SLTM8 v.4.0fc
PRIM v.4.0ec
MFIM100 v.E.0gd

Phones:

LIP8024 v1.1.af
LIP8004D v1.1.0ea. (upgraded later today by voip technician to v1.0fb)

The default gateway is also confirmed and set up correctly.

I'm not sure if this is directly related to calls dropping out but i decided to check the mls qos statistics on a port with an IP phone connected and a call in progress. I wanted the voip technician to mark the voice traffic as EF (dscp 46) and the signalling traffic as CS4 (dscp 40) however, i can only see packets marked with EF and and dscp 0-4 on the port.

The switch is setup to trust dscp values and there is no PC connected behind the phone. The voip technician was having trouble and wasn't able to figure it out.

Any assistance would be greatly appreciated.

Regards,

noobatron

 

[doktor]

Hi again

It seems OK to use EF for voice traffic and AF for signalling.

From this web page, I can see that you need to get upgraded some modules:

http://www.artcom.ru/download/?TYPE=6&MODEL=507

SLTM8 v.4.0fc -> v. 4.0Ja
PRIM v.4.0ec -> v. 4.0Ha
MFIM100 v.E.0gd -> v. 5.0Fb

You should ask your service supplier to do this, or hear about his experience.

///doktor