Cisco7940 after recieving call no voice

[Jako]

Hi,

I Hope there is help on this forum.

I have a 3CX installation with 3 cisco 7940 phones and 1 planet phone. My installation of the cisco 7940 phones is done like the configuration guide from 3cx's discribed. The planet phone has not so much configuration settins en is like a standaard VOIP phone connectet to the 3CX server.

The problem is the recieving calls on thes cisco's phone's. When we answerring a call we mostly don't hear the caller. The line seems to be death. The caller can hear us. After a few time switching on the phone between hold en resume sometimes we get the good connections. This is wierd for the caller because he can hear the switching between the music and the person. But most of the times whe have to call the person back.
All the rest of the functions work perfectly, whe can call out without a problem.

When we recieving calls on the planet Phone all calls are recieved perfectly. After a transfer to a cisco, the problem on the cisco stays. So i am pretty shure that the problem is on the Cisco's. because on a other phone it works fine.

Is there a solution for this ?

 

[worksighted]

Hi. If you call between 2 of the cisco phones can you hear each other?

 

[amygoda]

What is your external line?
do you use stun server ?

 

[Jako]

@amygoda:
For the lines we have 2 ISDN with a patton gateway and 4 VOIP numberst from 3stars provider.
The problem stays the same if we recizeve over the internet (VOIP) or over the ISDN. But the wierd thing is that the problem stays only when recieving on a Cisco. The planet Phone can recieve every phone call perfectly.

@worksighted
Calls between two cisco's there is no problem. There is a problem if 1 cisco recieved a call (with a bit of luck, hold, resume switching) and transfer these to another Cisco. With a transfer the problem is back. But not with a transfer to the planet Phone.

In my little opinion a think a cisco problem? could it be ???

 

[worksighted]

We have about 14 extensions as Cisco 7940s without any issues. Lets check a few things...

First, what audio codec is the 7940 phone set to use?

Second, do you have "PBX Delivers Audio" turned on or off in the PBX for these extensions?

Thanks.

 

[Jako]

Well my codex is "g711alaw" like 3cx configuration guide prefer. I also tested a week with the "g711ulaw" codex. But the problem stayd the same.

And yes a have "PBX Delivers Audio" ON.

Also a had the firmware "P0S3-08-8-00" on and degraded it to "P0S3-08-2-00" to be sure a had exact the same version that the configuration of 3CX described. So now a im running P0S3-08-2-00.

I hope this give you a better look at the problem.

 

[amygoda]

Hello
close all firewalls
try other stun server
try dmz

 

[worksighted]

What happens if you disbale PBX delivers audio?

 

[Jako]

@amygoda:
I did al these things. But after i think about it. There are direct ISDN lines that comes over a patton to the 3cx server. On these lines the problem is the same aldo there are not comming via a internet, firewall. So the problem is not the stun server, DMZ are Firewall.

@worksighted :
I disabled PABX Delivers audio. Tommorow a get feedback of this is working.

And guys, thanks for finding a solution. I keep you posted tommorow.

 

[amygoda]

what is your server firewall?

 

[Jako]

Hi,

The result of the disabeling off the PABX audio was not good. The problem stayed and di give an other problem. When making a call with the cisco's there was a second toon. You could hear 2 differtent waiting sounds (toon).

The firewall of the server is the standard ISS from a SBS 2003 server (where i don't have expiriance with).

Could there be other solutions for my problem, my boss is making me crazy. Whats normal because we loose 70 a 80% of are calls.

 

[worksighted]

Are you using tftp to load configurations to yoru cisco phones? If so....can you post sipdefault.cnf and <mac>.cnf

Mike

 

[Jako]

Like asked the config files

Sip default:

image_version: P0S3-08-2-00

# Proxy Server
proxy1_address: 192.168.38.55

# Proxy Server Port (default - 5060)
proxy1_port:5060

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 60

# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: g711alaw

#Logo URL
logo_url: "http://192.168.38.55:80/SipPhone/logoVDE.bmp"

# Setting voicemail
messages_uri: "99"

-------------------------------------------------------------------------
Phone config:

# Sip Configuration Generic File
image_version: P0S3-08-2-00

# Phone Label
Phone_label: Luc_VDE_

# Line 1 appearance
line1_name: 21

# Line 1 Registration Authentication
line1_authname: 21

# Line 1 Registration Password
line1_password: 21

# Line 2 appearance
line2_name:

# Line 2 Registration Authentication
line2_authname:

# Line 2 Registration Password
line2_password:

 

[Jako]

Like you can see these are the config out the book like 3cx preferd.

Any sugestions ?

 

[worksighted]

Here are copies of my working sipdefault.cnf and mac.cnf files for you......obviously you need to fill in xxxx's

# Image Version
image_version: "P0S3-08-9-00"

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "60"

# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: "g711alaw"

# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default)

# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: "1" ; 0-Disabled, 1-Enabled (default)

# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged

# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: "1"

# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt )
dtmf_outofband: "none"

# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: "3"

# SIP Timers
timer_t1: "500" ; Default 500 msec
timer_t2: "4000" ; Default 4 sec
sip_retx: "10" ; Default 11
sip_invite_retx: "6" ; Default 7
timer_invite_expires: "180" ; Default 180 sec


# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: "1" ; Default 0 (Do Not Disturb feature is off)

# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: "0" ; Default 0 (Disable sending all calls as anonymous)

# Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls)

# Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control)
call_waiting: "1" ; Default 1 (Call Waiting enabled)

# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: "101" ; Default 100

# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"

#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: "0"

#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "0"

# URL for branding logo - blank loads cisco image
logo_url: ""

# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: "0" ; Default 0 (Do Not Disturb feature is off)

call_waiting: "1"

# Voicemail button
messages_uri: 9999

#time settings
sntp_mode : DirectedBroadcast
sntp_server : xx.xx.xx.xx
time_zone : EST
dst_offset : 1
dst_start_month : Mar
dst_start_day : 2
dst_start_day_of_week : Sun
dst_start_week_of_month : 1
dst_start_time : 02
dst_stop_month : Nov
dst_stop_day : 1
dst_stop_day_of_week : Sun
dst_stop_week_of_month : 1
dst_stop_time : 2
dst_auto_adjust : 1


---------------------------------------------



#SIP Config Generic File for 7940G
image_version: "P0S3-08-9-00"

# Proxy Server
proxy1_address: "xxxxxxxx"
proxy2_address: "xxxxxxxx"

# Proxy Server Port (default - 5060)
proxy1_port:"5060"
proxy2_port:"5060"

#Line 1 appearance
line1_name: "xxxx"

#Line 1 Registration Authentication
line1_authname: "xxxx"

#Line 1 Registration Password
line1_password: "xxxxxxxx"

#Line 2 appearance
line2_name: "xxxx"

#Line 2 Registration Authentication
line2_authname: "xxxx"

#Line 2 Registration Password
line2_password: "xxxxxxxxx"

# NAT/Firewall Traversal
#nat_enable: "1"
#nat_address: "xx.xx.xx.xx" ;
#voip_control_port: "5060"
#start_media_port: "16384"
#end_media_port: "16397"
#nat_received_processing: "1"

# Phone Label
phone_label: "xxxxxxxx" ;

# Voicemail button
messages_uri: "9999"

 

[Jako]

Here are copies of my working sipdefault.cnf and mac.cnf files for you......obviously you need to fill in xxxx's


#SIP Config Generic File for 7940G
image_version: "P0S3-08-9-00"

# NAT/Firewall Traversal
#nat_enable: "1"
#nat_address: "xx.xx.xx.xx" ;
#voip_control_port: "5060"
#start_media_port: "16384"
#end_media_port: "16397"
#nat_received_processing: "1"


Could you tell me wath exactly i have to fill in in the nat adress (XX.xx.xx.xx).
Is this mi wan IP adres or mij lan 3cx server IP ?

 

[Jako]

At hte moment a im testing with the new config and firmware.

Ill keep you posted. (fingers crosd)

 

[worksighted]

you dont need the nat address part. That is used for remote phoens only. Just leave it commented out with the #

 

[worksighted]

and you will want to change the firmware reference to whatever you are using.

 

[Jako]

Ower first expirience with the new config is good.

The calls are comming perfectly, whe have to waite till tomorrow to be sure of this evaluation.

What did i do:
Change the firmware of the cisco's to P0S3-08-9-00
Change the config to the standaard that is posted in this thread. (I did change the IP's and extensions )

One thing a have to test is without the "NAT" setting.
@worksighted : Are you sure that this would give influence to this at the moment good working system ??

There is only one problem that stays:
The outside calls are recieving well. When we want to transfer it the ather person, he recieve the call but have to push 1 time hold and resume. After that there is a perfect communication.
This is a office with a few persons (in the same deskspace) so it isn't neccesary to first contact the person and after that pass the line to that person. Could it be possible that the problem would be gone if we could configure after piking up a external line, directly pass the line true another extension without waiting till thy pick up.
This would be better and maybe the problem dissepiar.

Thanks in advance