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
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@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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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
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
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@packetcall.net
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@packetcall.net
Users mailing list Users@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
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@packetcall.net
-----Original Message----- From: Stephen Varga [mailto:varga@zbzoom.net] Sent: Wednesday, February 21, 2007 11:38 AM To: syagel@packetcall.net Cc: users@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
Hi,
The password is cisco if it's default without any spesific config.
-Atle * Scott Yagel syagel@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@packetcall.net
-----Original Message----- From: Stephen Varga [mailto:varga@zbzoom.net] Sent: Wednesday, February 21, 2007 11:38 AM To: syagel@packetcall.net Cc: users@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.sb2I 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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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@packetcall.net
-----Original Message----- From: Stephen Varga [mailto:varga@zbzoom.net] Sent: Wednesday, February 21, 2007 11:38 AM To: syagel@packetcall.net Cc: users@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.sb2I 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
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/ipp79...
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
-
Atle Samuelsen -
Bogdan-Andrei Iancu -
Ovidiu Sas -
Scott Yagel -
Stephen Varga -
Stéphane Alnet