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Peter J VERNON

2005-Mar-30 20:54 UTC

[Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here

Guys......

I have Asterisk CVS-NHEAD-03/19/05-21:56:28 running on a box here and have a 
couple of Cisco 7960s and a Grandstream phone.

I can make calls from the 7960. When I get a call placed to the 7960 the call 
is setup but there is no audio in either direction. This is for a call placed 
on the local subnet between extensions so I doubt that it is a NAT issue 
though I have tried a number of combinations of this to no avail.

I have tried firmware versions 6 & 7 on the Cisco phones, same result. I
have
tried the phones on two other Asterisk installs and they work fine. I have 
compared sip.conf with these and can see no differences.

If I configure the grandstream up to replace the 7960 it works fine.

I have noticed that the src port (TCP port on the phone) increments during 
the session which seems to be the issue.

Anyone seen this before? Any assistance would be appreciated.

Regards
Peter 

Here are some of the settings.

Sip.conf for the extension:

[9001]
type=friend                     ; either "friend" (peer+user),
"peer"
or "user"
context=extensions
secret=9001
fromuser=Cisco          ; overrides the callerid, e.g. required by FWD
callerid=9001
host=dynamic            ; we have a static but private IP address
nat=never                          ; there is not NAT between phone and 
Asterisk
dtmfmode=rfc2833
canreinvite=no          ; allow RTP voice traffic to bypass Asterisk
incominglimit=5         ; permit only 1 outgoing call at a time
outgoinglimit=5
progressinband=yes
disallow=all
allow=ulaw
mailbox=9001


Phone config:

Loadid:  SW: P0S3-05-3-00  ARM: PAS3ARM1  Boot: PC030300  DSP: PS03AS30
SIP Phone> show config
------ Current *FLASH* Configuration ------

Platform : Cisco IP Phone 7960
Elasped Time: 15:15:24

dhcp_server : 192.168.10.254
my_ip_addr : 192.168.10.17
subnet_mask : 255.255.255.0
defaultgw : 192.168.10.254
dyn_dns_addr_1 : 0.0.0.0
dyn_dns_addr_2 : 0.0.0.0
dns_addr : 203.194.26.236
dns_backup_1: 203.194.26.235
tftp_addr : 192.168.11.2
dyn_tftp_addr : 0.0.0.0
my_mac_addr : 0003:e348:e5cb
domain_name : home.jbo.com.au
my_name : SIP0003E348E5CB
Status Flags : 12310000

image_version : "P0S3-07-3-00"
FirmLoadID : "PC030300"
network_media_type : Auto
network_port2_type : Hub/Switch
tos_media : 5
phone_label : "UNPROVISIONED"
tftp_cfg_dir : ""
phone_password : **********
phone_prompt : "SIP Phone"
language : english
sntp_mode : DirectedBroadcast
sntp_server : 0.0.0.0
time_zone : EAST
dst_offset : 01/00
dst_start_month : April
dst_start_day : 0
dst_start_day_of_week : Sunday
dst_start_week_of_month : 1
dst_start_time : 02/00
dst_stop_month : October
dst_stop_day : 0
dst_stop_day_of_week : Sunday
dst_stop_week_of_month : 0
dst_stop_time : 02/00
dst_auto_adjust : 1
time_format_24hr : 1
date_format : D/M/Y
nat_enable : 0
nat_address : UNPROVISIONED
voip_control_port : 5060
start_media_port : 16384
end_media_port : 32766
sync : "1"
xml_card_dir : ""
xml_card_file : "CARD.XML"
telnet_level : 2
services_url : "http://192.168.11.2/cgi-bin/rss2cisco.pl"
directory_url : "http://192.168.11.2/directory.html"
logo_url : "http://192.168.11.2/asterisk-tux.bmp"
http_proxy_addr : UNPROVISIONED
http_proxy_port : 80
enable_vad : 0
dial_template : ""
callerid_blocking : 0
anonymous_call_block : 0
autocomplete : 1
messages_uri : "199"
dnd_control : 0
preferred_codec : g711ulaw
dtmf_outofband : avt
dtmf_avt_payload : 101
dtmf_db_level : 3
dtmf_inband : 1
line1_name : "9001"
line2_name : "UNPROVISIONED"
line3_name : "UNPROVISIONED"
line4_name : "UNPROVISIONED"
line5_name : "UNPROVISIONED"
line6_name : "UNPROVISIONED"
line1_authname : "9001"
line2_authname : "UNPROVISIONED"
line3_authname : "UNPROVISIONED"
line4_authname : "UNPROVISIONED"
line5_authname : "UNPROVISIONED"
line6_authname : "UNPROVISIONED"
line1_password : **********
line2_password : **********
line3_password : **********
line4_password : **********
line5_password : **********
line6_password : **********
line1_shortname : "9001"
line2_shortname : "UNPROVISIONED"
line3_shortname : "UNPROVISIONED"
line4_shortname : "UNPROVISIONED"
line5_shortname : "UNPROVISIONED"
line6_shortname : "UNPROVISIONED"
line1_displayname : "9001"
line2_displayname : "UNPROVISIONED"
line3_displayname : "UNPROVISIONED"
line4_displayname : "UNPROVISIONED"
line5_displayname : "UNPROVISIONED"
line6_displayname : "UNPROVISIONED"
proxy1_address : "192.168.10.106"
proxy2_address : "UNPROVISIONED"
proxy3_address : "UNPROVISIONED"
proxy4_address : "UNPROVISIONED"
proxy5_address : "UNPROVISIONED"
proxy6_address : "UNPROVISIONED"
proxy1_port : 5060
proxy2_port : 5060
proxy3_port : 5060
proxy4_port : 5060
proxy5_port : 5060
proxy6_port : 5060
sip_retx : 10
sip_invite_retx : 6
timer_t1 : 500
timer_t2 : 4000
timer_invite_expires : 180
timer_register_expires : 3600
proxy_register : 1
proxy_backup : "UNPROVISIONED"
proxy_emergency : "UNPROVISIONED"
proxy_backup_port : 5060
proxy_emergency_port : 5060
outbound_proxy : 192.168.10.106
outbound_proxy_port : 5060
nat_received_processing : 0
mwi_status : 0
call_waiting : 1
user_info : none
cnf_join_enable : 1
remote_party_id : 0
semi_attended_transfer : 1
call_hold_ringback : 0
SIP Phone>

Kristian Kielhofner

2005-Mar-30 22:05 UTC

head link

[Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here

Peter J VERNON wrote:> Guys......
> 
> I have Asterisk CVS-NHEAD-03/19/05-21:56:28 running on a box here and have
a
> couple of Cisco 7960s and a Grandstream phone.
> 
> I can make calls from the 7960. When I get a call placed to the 7960 the
call
> is setup but there is no audio in either direction. This is for a call
placed
> on the local subnet between extensions so I doubt that it is a NAT issue 
> though I have tried a number of combinations of this to no avail.
> 
> I have tried firmware versions 6 & 7 on the Cisco phones, same result.
I have
> tried the phones on two other Asterisk installs and they work fine. I have 
> compared sip.conf with these and can see no differences.
> 
> If I configure the grandstream up to replace the 7960 it works fine.
> 
> I have noticed that the src port (TCP port on the phone) increments during 
> the session which seems to be the issue.
> 
> Anyone seen this before? Any assistance would be appreciated.
> 
> Regards
> Peter 
Peter,

	This one bit several of us.  Upgrade to CVS-Head as of 3/22/2005 or later.

--
Kristian Kielhofner

irakli.natsvlishvili@thinkingvoice.com

2005-Mar-30 22:10 UTC

head link

[Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here

G'day mate,

I've got 15 7960/7940 in my office with firmware 7.4 and have no problems.
>I can make calls from the 7960. When I get a call placed to the 7960 the
call>is setup but there is no audio in either direction.
Is call from 7960 to 7960?
> I have tried firmware versions 6 & 7 on the Cisco phones, same result.
Means - something wrong is with your config...


[9001]>type=friend                     ; either "friend" (peer+user),
"peer"
>context=extensions
>secret=9001
>fromuser=Cisco          ; overrides the callerid, e.g. required by FWD
>callerid=9001
>host=dynamic            ; we have a static but private IP address
>nat=never                          ; there is not NAT between phone and
>dtmfmode=rfc2833
^^^^^^^^^^^^^^^ see below, in phone section.
> canreinvite=no          ; allow RTP voice traffic to bypass Asterisk
Hmm... something tells me that RTP stream goes to asterisk, instead of
phones' ip addresses. Could you check, what parameter do you have in
global and second 7960's section?
> progressinband=yes
Why do you need this?
>disallow=all
>allow=ulaw
Seconf 7960 has the same config in SIP.CONF?
>dhcp_server : 192.168.10.254
>my_ip_addr : 192.168.10.17
>subnet_mask : 255.255.255.0
>defaultgw : 192.168.10.254
>tftp_addr : 192.168.11.2
Does phone receive sipdefault.cnf and SIPxxxx.cnf file from TFTP?
>dtmf_outofband : avt
>dtmf_avt_payload : 101
>dtmf_db_level : 3
>dtmf_inband : 1
You've set in Asterk's config DTMF as out_of_band, while in phone's
config
you'set as in_band. Corret it first.
>proxy1_address : "192.168.10.106"
do sip debug ip ip_address_of_7960 ad take a look.

Good luck!

John Breeden

2005-Mar-31 01:30 UTC

head link

[Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here

This bit me too. Had to turn nat off on the 7960Gs

Kristian Kielhofner wrote:
> Peter J VERNON wrote:
>
>> Guys......
>>
>> I have Asterisk CVS-NHEAD-03/19/05-21:56:28 running on a box here and 
>> have a couple of Cisco 7960s and a Grandstream phone.
>>
>> I can make calls from the 7960. When I get a call placed to the 7960 
>> the call is setup but there is no audio in either direction. This is 
>> for a call placed on the local subnet between extensions so I doubt 
>> that it is a NAT issue though I have tried a number of combinations 
>> of this to no avail.
>>
>> I have tried firmware versions 6 & 7 on the Cisco phones, same 
>> result. I have tried the phones on two other Asterisk installs and 
>> they work fine. I have compared sip.conf with these and can see no 
>> differences.
>>
>> If I configure the grandstream up to replace the 7960 it works fine.
>>
>> I have noticed that the src port (TCP port on the phone) increments 
>> during the session which seems to be the issue.
>>
>> Anyone seen this before? Any assistance would be appreciated.
>>
>> Regards
>> Peter 
>
>
> Peter,
>
>     This one bit several of us.  Upgrade to CVS-Head as of 3/22/2005 
> or later.
>
> -- 
> Kristian Kielhofner
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>

HILLMANN, DARREN

2005-Mar-31 09:46 UTC

head link

[Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here

I had the same problem recently, and it didn't have anything to do with
NAT.
Try looking at the port range in the rtp.conf file.  Make sure it
matches the port range configured on your 7960s.

- Darren


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Peter J
VERNON
Sent: Wednesday, March 30, 2005 10:54 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 and Asterisk,I think I have a curly
one here


Guys......

I have Asterisk CVS-NHEAD-03/19/05-21:56:28 running on a box here and
have a 
couple of Cisco 7960s and a Grandstream phone.

I can make calls from the 7960. When I get a call placed to the 7960 the
call 
is setup but there is no audio in either direction. This is for a call
placed 
on the local subnet between extensions so I doubt that it is a NAT issue

though I have tried a number of combinations of this to no avail.

I have tried firmware versions 6 & 7 on the Cisco phones, same result. I
have 
tried the phones on two other Asterisk installs and they work fine. I
have 
compared sip.conf with these and can see no differences.

If I configure the grandstream up to replace the 7960 it works fine.

I have noticed that the src port (TCP port on the phone) increments
during 
the session which seems to be the issue.

Anyone seen this before? Any assistance would be appreciated.

Regards
Peter 

Here are some of the settings.

Sip.conf for the extension:

[9001]
type=friend                     ; either "friend" (peer+user),
"peer"
or "user"
context=extensions
secret=9001
fromuser=Cisco          ; overrides the callerid, e.g. required by FWD
callerid=9001
host=dynamic            ; we have a static but private IP address
nat=never                          ; there is not NAT between phone and 
Asterisk
dtmfmode=rfc2833
canreinvite=no          ; allow RTP voice traffic to bypass Asterisk
incominglimit=5         ; permit only 1 outgoing call at a time
outgoinglimit=5
progressinband=yes
disallow=all
allow=ulaw
mailbox=9001


Phone config:

Loadid:  SW: P0S3-05-3-00  ARM: PAS3ARM1  Boot: PC030300  DSP: PS03AS30
SIP Phone> show config
------ Current *FLASH* Configuration ------

Platform : Cisco IP Phone 7960
Elasped Time: 15:15:24

dhcp_server : 192.168.10.254
my_ip_addr : 192.168.10.17
subnet_mask : 255.255.255.0
defaultgw : 192.168.10.254
dyn_dns_addr_1 : 0.0.0.0
dyn_dns_addr_2 : 0.0.0.0
dns_addr : 203.194.26.236
dns_backup_1: 203.194.26.235
tftp_addr : 192.168.11.2
dyn_tftp_addr : 0.0.0.0
my_mac_addr : 0003:e348:e5cb
domain_name : home.jbo.com.au
my_name : SIP0003E348E5CB
Status Flags : 12310000

image_version : "P0S3-07-3-00"
FirmLoadID : "PC030300"
network_media_type : Auto
network_port2_type : Hub/Switch
tos_media : 5
phone_label : "UNPROVISIONED"
tftp_cfg_dir : ""
phone_password : **********
phone_prompt : "SIP Phone"
language : english
sntp_mode : DirectedBroadcast
sntp_server : 0.0.0.0
time_zone : EAST
dst_offset : 01/00
dst_start_month : April
dst_start_day : 0
dst_start_day_of_week : Sunday
dst_start_week_of_month : 1
dst_start_time : 02/00
dst_stop_month : October
dst_stop_day : 0
dst_stop_day_of_week : Sunday
dst_stop_week_of_month : 0
dst_stop_time : 02/00
dst_auto_adjust : 1
time_format_24hr : 1
date_format : D/M/Y
nat_enable : 0
nat_address : UNPROVISIONED
voip_control_port : 5060
start_media_port : 16384
end_media_port : 32766
sync : "1"
xml_card_dir : ""
xml_card_file : "CARD.XML"
telnet_level : 2
services_url : "http://192.168.11.2/cgi-bin/rss2cisco.pl"
directory_url : "http://192.168.11.2/directory.html"
logo_url : "http://192.168.11.2/asterisk-tux.bmp"
http_proxy_addr : UNPROVISIONED
http_proxy_port : 80
enable_vad : 0
dial_template : ""
callerid_blocking : 0
anonymous_call_block : 0
autocomplete : 1
messages_uri : "199"
dnd_control : 0
preferred_codec : g711ulaw
dtmf_outofband : avt
dtmf_avt_payload : 101
dtmf_db_level : 3
dtmf_inband : 1
line1_name : "9001"
line2_name : "UNPROVISIONED"
line3_name : "UNPROVISIONED"
line4_name : "UNPROVISIONED"
line5_name : "UNPROVISIONED"
line6_name : "UNPROVISIONED"
line1_authname : "9001"
line2_authname : "UNPROVISIONED"
line3_authname : "UNPROVISIONED"
line4_authname : "UNPROVISIONED"
line5_authname : "UNPROVISIONED"
line6_authname : "UNPROVISIONED"
line1_password : **********
line2_password : **********
line3_password : **********
line4_password : **********
line5_password : **********
line6_password : **********
line1_shortname : "9001"
line2_shortname : "UNPROVISIONED"
line3_shortname : "UNPROVISIONED"
line4_shortname : "UNPROVISIONED"
line5_shortname : "UNPROVISIONED"
line6_shortname : "UNPROVISIONED"
line1_displayname : "9001"
line2_displayname : "UNPROVISIONED"
line3_displayname : "UNPROVISIONED"
line4_displayname : "UNPROVISIONED"
line5_displayname : "UNPROVISIONED"
line6_displayname : "UNPROVISIONED"
proxy1_address : "192.168.10.106"
proxy2_address : "UNPROVISIONED"
proxy3_address : "UNPROVISIONED"
proxy4_address : "UNPROVISIONED"
proxy5_address : "UNPROVISIONED"
proxy6_address : "UNPROVISIONED"
proxy1_port : 5060
proxy2_port : 5060
proxy3_port : 5060
proxy4_port : 5060
proxy5_port : 5060
proxy6_port : 5060
sip_retx : 10
sip_invite_retx : 6
timer_t1 : 500
timer_t2 : 4000
timer_invite_expires : 180
timer_register_expires : 3600
proxy_register : 1
proxy_backup : "UNPROVISIONED"
proxy_emergency : "UNPROVISIONED"
proxy_backup_port : 5060
proxy_emergency_port : 5060
outbound_proxy : 192.168.10.106
outbound_proxy_port : 5060
nat_received_processing : 0
mwi_status : 0
call_waiting : 1
user_info : none
cnf_join_enable : 1
remote_party_id : 0
semi_attended_transfer : 1
call_hold_ringback : 0
SIP Phone> 



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asterisk users - Mar 2005 - Cisco 7960 and Asterisk, I think I have a curly one here

[Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here

[Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here

[Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here

[Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here

[Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here