21 Feb
2007
21 Feb
'07
9:29 a.m.
Hello,
I am doing interop testing with Level3, and need to be able to rewrite
the To: field in requests to match their requirements. I can fake the
rewrites inside route[], but although the packets sent out look like
they use the new "To:", the header is never parsed again, so the
internal structures of SER still hold the former (incorrect) value.
When a request hit failure_route[] (for example when a CANCEL is sent
when an INVITE times out) the original, incorrect To: header is then
sent (and there's no way to modify the content of the CANCEL from
within failure_route[]).
Is there a standard way to alter the internally-stored value for To:
(and From:, since I'm asking)?
If not, could someone point me to the right place in the code to start
adding this kind of feature?
Thanks,
Stéphane Alnet
--
http://carrierclass.net/
http://www.linkedin.com/in/stephalnet
21 Feb
21 Feb
10:14 a.m.
Hi Stéphane,
If you search the mailing list you will see that this topic was
discussed already.
Here is the right module to mess with:
http://openser.org/docs/modules/1.2.x/uac.html
For now it can deal only with "From" header.
Hope this helps,
Ovidiu Sas
On 2/21/07, Stéphane Alnet <stephane(a)carrierclass.net> wrote:
Hello,
I am doing interop testing with Level3, and need to be able to rewrite
the To: field in requests to match their requirements. I can fake the
rewrites inside route[], but although the packets sent out look like
they use the new "To:", the header is never parsed again, so the
internal structures of SER still hold the former (incorrect) value.
When a request hit failure_route[] (for example when a CANCEL is sent
when an INVITE times out) the original, incorrect To: header is then
sent (and there's no way to modify the content of the CANCEL from
within failure_route[]).
Is there a standard way to alter the internally-stored value for To:
(and From:, since I'm asking)?
If not, could someone point me to the right place in the code to start
adding this kind of feature?
Thanks,
Stéphane Alnet
--
http://carrierclass.net/
http://www.linkedin.com/in/stephalnet
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
10:33 a.m.
Ovidiu,
Here is the right module to mess with:
http://openser.org/docs/modules/1.2.x/uac.html
For now it can deal only with "From" header.
I had seen UAC but I really need to modify the To: field (I've been
able to work around the From: field so far).
Hope this helps,
Sure! I think I'll try to modify UAC to add a function to modify To:
and see how that goes.
Thank you,
Stéphane
--
http://carrierclass.net/
http://www.linkedin.com/in/stephalnet
4:27 p.m.
Hi Stéphane,
my advice is not to change TO hdr for routing reasons as this is against
RFC3261 - routing is done exclusively based on RURI and Route headers ;
TO hdr has 0-value in routing.
the level3 gw is quite bogus and not SIP compliant.
regards,
bogdan
Stéphane Alnet wrote:
Ovidiu,
Here is the right module to mess with:
http://openser.org/docs/modules/1.2.x/uac.html
For now it can deal only with "From" header.
I had seen UAC but I really need to modify the To: field (I've been
able to work around the From: field so far).
Hope this helps,
Sure! I think I'll try to modify UAC to add a function to modify To:
and see how that goes.
Thank you,
Stéphane
11:29 a.m.
New subject: [Users] Cisco 7960 phones
Hi,
We have inherited some older Cisco 7960 SCCP phones that I would like to
upgrade/change to SIP and use on our OpenSER installation. The phones are
on release 3.1, and the latest release on the Cisco site is 8.2. I have
perused the many forums on the 'Net and tried all the installation tricks
and recipes that have been recommended, all to no avail. The last thing I
can try is to upgrade to intermediate releases to eventually get to the
latest one. Unfortunately, I have not been able to find the older binaries
anywhere.
If anyone has any suggestions, or know where I can find the intermediate
releases, the info would be appreciated.
Thanks,
Scott Yagel
PacketCall, Inc.
syagel(a)packetcall.net
12:38 p.m.
New subject: [Users] Cisco 7960 phones
Scott,
Put the attached files into your tftp server that the phones are pulling
their configs and image from. You should also have the following image
files in the same tftp server:
P003-08-2-00.bin
P0S3-08-2-00.loads
P0S3-08-2-00.zip
P003-08-2-00.sbn
P0S3-08-2-00.sb2
I have used these files to successfully upgrade phones.
If you have SIPxxxxxxxxxxxx.cnf file created for the phones, make sure
the image_version line is commented out.
The biggest problem I have with this version is using that it reads the
dialplan.xml file but it does not work; I am guessing there is a syntax
change but can not been able to find SIP documentation related to the
8.2 code on Cisco's website. I have been staying with the 7.5 version
of the code because of this problem.
I found watching which files are being requested by phone a big help in
getting the loads to work properly
Let me know what happens.
Steve
<file:///L:/tmp/SIPDefault.cnf.cnf>
Scott Yagel wrote:
Hi,
We have inherited some older Cisco 7960 SCCP phones that I would like to
upgrade/change to SIP and use on our OpenSER installation. The phones are
on release 3.1, and the latest release on the Cisco site is 8.2. I have
perused the many forums on the 'Net and tried all the installation tricks
and recipes that have been recommended, all to no avail. The last thing I
can try is to upgrade to intermediate releases to eventually get to the
latest one. Unfortunately, I have not been able to find the older binaries
anywhere.
If anyone has any suggestions, or know where I can find the intermediate
releases, the info would be appreciated.
Thanks,
Scott Yagel
PacketCall, Inc.
syagel(a)packetcall.net
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
P003-08-2-00
# SIP Default Generic Configuration File
# Image Version
image_version: "P0S3-08-2-00"
# Proxy Server
proxy1_address: "" ; Can be dotted IP or FQDN
proxy2_address: "" ; Can be dotted IP or FQDN
proxy3_address: "" ; Can be dotted IP or FQDN
proxy4_address: "" ; Can be dotted IP or FQDN
proxy5_address: "" ; Can be dotted IP or FQDN
proxy6_address: "" ; Can be dotted IP or FQDN
# Proxy Server Port (default - 5060)
proxy1_port: 5060
proxy2_port: 5060
proxy3_port: 5060
proxy4_port: 5060
proxy5_port: 5060
proxy6_port: 5060
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 360
# Codec for media stream (g711ulaw (default), g711alaw, g729a)
#preferred_codec: g711ulaw
preferred_codec: g711ulaw
# TOS bits in media stream [0-5] (Default - 5)
tos_media: 5
# 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: avt
# 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 10
sip_invite_retx: 6 ; Default 6
timer_invite_expires: 180 ; Default 180 sec
####### New Parameters added in Release 2.0 #######
# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: dialplan
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "" ; Example: ./sip_phone/
# Time Server (There are multiple values and configurations refer to Admin Guide for
Specifics)
sntp_server: "" ; SNTP Server IP Address
sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast
(default)
time_zone: EST ; Time Zone Phone is in
dst_offset: 1 ; Offset from Phone's time when DST is in effect
dst_start_month: April ; Month in which DST starts
dst_start_day: "" ; Day of month in which DST starts
dst_start_day_of_week: Sun ; Day of week in which DST starts
dst_start_week_of_month: 1 ; Week of month in which DST starts
dst_start_time: 02 ; Time of day in which DST starts
dst_stop_month: Oct ; Month in which DST stops
dst_stop_day: "" ; Day of month in which DST stops
dst_stop_day_of_week: Sunday ; Day of week in which DST stops
dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month
dst_stop_time: 2 ; Time of day in which DST stops
dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 0 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
# 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)
# Caller ID Blocking (0-disbaled, 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-disabled, 1-enabled, 2-disabled no user control, 3-enabled no
user control)
anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls)
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101 ; Default 101
# Sync value of the phone used for remote reset
sync: 1 ; Default 1
####### New Parameters added in Release 2.1 #######
# Backup Proxy Support
proxy_backup: "" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)
# Emergency Proxy Support
proxy_emergency: "" ; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060)
# Configurable VAD option
enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable
####### New Parameters added in Release 2.2 ######
# NAT/Firewall Traversal
nat_enable: 0 ; 0-Disabled (default), 1-Enabled
nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record
only)
voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060)
start_media_port: 16384 ; Start RTP range for media (default - 16384)
end_media_port: 32766 ; End RTP range for media (default - 32766)
nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled
# Outbound Proxy Support
outbound_proxy: "" ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060 ; default is 5060
####### New Parameter added in Release 3.0 #######
# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default)
####### New Parameters added in Release 3.1 #######
# 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 the phone)
telnet_level: 2 ; 0-Disabled (default), 1-Enabled, 2-Privileged
####### New Parameters added in Release 4.0 #######
# XML URLs
services_url: "" ; URL for external Phone Services
directory_url: "" ; URL for external Directory location
logo_url: "" ; URL for branding logo to be used on phone display
# HTTP Proxy Support
http_proxy_addr: "" ; Address of HTTP Proxy server
http_proxy_port: 80 ; Port of HTTP Proxy Server (80-default)
# Dynamic DNS/TFTP Support
dyn_dns_addr_1: "" ; restricted to dotted IP
dyn_dns_addr_2: "" ; restricted to dotted IP
dyn_tftp_addr: "" ; restricted to dotted IP
# Remote Party ID
remote_party_id: 0 ; 0-Disabled (default), 1-Enabled
####### New Parameters added in Release 4.4 #######
# Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control)
call_hold_ringback: 0 ; Default 0 (Call Hold Ringback feature is off)
####### New Parameters added in Release 6.0 #######
# Dialtone Stutter for MWI
stutter_msg_waiting: 0 ; 0-Disabled (default), 1-Enabled
# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 0 ; 0-Disabled (default), 1-Enabled
1:41 p.m.
New subject: [Users] Cisco 7960 phones
Steve,
Thanks, this worked really well. I managed to update one of the phones, at
least to a certain point. It downloaded the new image and read at least
part of the configuration file. It is now sitting at the "Phone
Unprovisioned" state. The only problem I have now is that, to complete the
provisioning, I can't seem to get the Network Configuration unlocked. The
old method of hitting **# doesn't seem to work any more.
Thanks again,
Scott Yagel
PacketCall, Inc.
syagel(a)packetcall.net
-----Original Message-----
From: Stephen Varga [mailto:varga@zbzoom.net]
Sent: Wednesday, February 21, 2007 11:38 AM
To: syagel(a)packetcall.net
Cc: users(a)openser.org
Subject: Re: [Users] Cisco 7960 phones
Scott,
Put the attached files into your tftp server that the phones are pulling
their configs and image from. You should also have the following image
files in the same tftp server:
P003-08-2-00.bin
P0S3-08-2-00.loads
P0S3-08-2-00.zip
P003-08-2-00.sbn
P0S3-08-2-00.sb2
I have used these files to successfully upgrade phones.
If you have SIPxxxxxxxxxxxx.cnf file created for the phones, make sure
the image_version line is commented out.
The biggest problem I have with this version is using that it reads the
dialplan.xml file but it does not work; I am guessing there is a syntax
change but can not been able to find SIP documentation related to the
8.2 code on Cisco's website. I have been staying with the 7.5 version
of the code because of this problem.
I found watching which files are being requested by phone a big help in
getting the loads to work properly
Let me know what happens.
Steve
1:47 p.m.
New subject: [Users] Cisco 7960 phones
Hi,
The password is cisco if it's default without any spesific config.
-Atle
* Scott Yagel <syagel(a)packetcall.net> [070221 19:41]:
Steve,
Thanks, this worked really well. I managed to update one of the phones, at
least to a certain point. It downloaded the new image and read at least
part of the configuration file. It is now sitting at the "Phone
Unprovisioned" state. The only problem I have now is that, to complete the
provisioning, I can't seem to get the Network Configuration unlocked. The
old method of hitting **# doesn't seem to work any more.
Thanks again,
Scott Yagel
PacketCall, Inc.
syagel(a)packetcall.net
-----Original Message-----
From: Stephen Varga [mailto:varga@zbzoom.net]
Sent: Wednesday, February 21, 2007 11:38 AM
To: syagel(a)packetcall.net
Cc: users(a)openser.org
Subject: Re: [Users] Cisco 7960 phones
Scott,
Put the attached files into your tftp server that the phones are pulling
their configs and image from. You should also have the following image
files in the same tftp server:
P003-08-2-00.bin
P0S3-08-2-00.loads
P0S3-08-2-00.zip
P003-08-2-00.sbn
P0S3-08-2-00.sb2
I have used these files to successfully upgrade phones.
If you have SIPxxxxxxxxxxxx.cnf file created for the phones, make sure
the image_version line is commented out.
The biggest problem I have with this version is using that it reads the
dialplan.xml file but it does not work; I am guessing there is a syntax
change but can not been able to find SIP documentation related to the
8.2 code on Cisco's website. I have been staying with the 7.5 version
of the code because of this problem.
I found watching which files are being requested by phone a big help in
getting the loads to work properly
Let me know what happens.
Steve
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
2:30 p.m.
New subject: [Users] Cisco 7960 phones
Scott
You can set the password in the SIPDefault.cnf
I add the following line to the SIPDefault.cnf file
phone_password: "adgjm"
Makes it real easy to get into config mode until you get your settings
the way you want them, then I use a more unique specific one in the
SIP<mac-address>.cnf file for each phone.
You can get to the SIP code documentation on the Cisco website without a
login. It does a pretty good job of describing the parameters and the
various config files.
As Atle mentioned the default is 'cisco' until you set it in a config file.
Steve
Scott Yagel wrote:
Steve,
Thanks, this worked really well. I managed to update one of the phones, at
least to a certain point. It downloaded the new image and read at least
part of the configuration file. It is now sitting at the "Phone
Unprovisioned" state. The only problem I have now is that, to complete the
provisioning, I can't seem to get the Network Configuration unlocked. The
old method of hitting **# doesn't seem to work any more.
Thanks again,
Scott Yagel
PacketCall, Inc.
syagel(a)packetcall.net
-----Original Message-----
From: Stephen Varga [mailto:varga@zbzoom.net]
Sent: Wednesday, February 21, 2007 11:38 AM
To: syagel(a)packetcall.net
Cc: users(a)openser.org
Subject: Re: [Users] Cisco 7960 phones
Scott,
Put the attached files into your tftp server that the phones are pulling
their configs and image from. You should also have the following image
files in the same tftp server:
P003-08-2-00.bin
P0S3-08-2-00.loads
P0S3-08-2-00.zip
P003-08-2-00.sbn
P0S3-08-2-00.sb2
I have used these files to successfully upgrade phones.
If you have SIPxxxxxxxxxxxx.cnf file created for the phones, make sure
the image_version line is commented out.
The biggest problem I have with this version is using that it reads the
dialplan.xml file but it does not work; I am guessing there is a syntax
change but can not been able to find SIP documentation related to the
8.2 code on Cisco's website. I have been staying with the 7.5 version
of the code because of this problem.
I found watching which files are being requested by phone a big help in
getting the loads to work properly
Let me know what happens.
Steve
1:41 p.m.
New subject: [Users] Cisco 7960 phones
Scott,
We have inherited some older Cisco 7960 SCCP phones
that I would like to
upgrade/change to SIP and use on our OpenSER installation. The phones are
on release 3.1, and the latest release on the Cisco site is 8.2.
Have a look at
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7…
for the upgrade options. It look like you would have to upgrade to 6.3
first (to get the new boot loader) then to 8.2
Unfortunately, I have not been able to find the older
binaries
anywhere.
Have you tried to ask Cisco? You should be able to download them from
CCO if you have a valid software agreement. You should even be able to
download 8.4.
Stéphane
--
http://carrierclass.net/
http://www.linkedin.com/in/stephalnet
6491
days inactive
6491
days old
9 comments
6 participants
participants (6)
-
Atle Samuelsen -
Bogdan-Andrei Iancu -
Ovidiu Sas -
Scott Yagel -
Stephen Varga -
Stéphane Alnet