So they did acknowledge it's a mistake? I was especially curious if it was intentional on their part or not cos it seems so odd. I'll be sure to aim for a worldwide deployment of FS so I can use that same excuse some day. =D j.k. On Mon, Jun 2, 2008 at 11:46 AM, UV <uv@talknet.com.au> wrote:
We already contacted the company and they may or may not fix their phones (my Chinese is not what it used to be J) however, regardless, this phone's customer-base in Australia is huge as well as worldwide and a mass firmware update is not a viable option.
For us, this is one of those instances where a custom SIP "flavor" implementation to support these devices is immanent.
These phones are working on in-band DTMF for years because of their "special" RFC2833 implementation.
We'll test the patch and let you know.
Thanks for the support!
------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Tuesday, June 03, 2008 2:01 AM
*To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Shutting down RFC
Can you contact company who makes this phone and ask them to explain what reasoning they would have to deny a call based on an innocuous sip header that should not influence the codec negotiation in any way? Especially one recommended by the RFC.
This may be a Denial Of Service on the phone and I'm sure they would want to be notified that their commercial product that they sell and make money on is not behaving properly.
On Mon, Jun 2, 2008 at 10:35 AM, UV <uv@talknet.com.au> wrote:
Working call scenario capture:
http://pastebin.freeswitch.org/4546
Non-working call scenario capture:
http://pastebin.freeswitch.org/4547
Both calls initiated via an event socket command: api originate sofia/default/1001 &park
Let me know if there's more info needed.
UV ------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Monday, June 02, 2008 11:39 PM
*To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Shutting down RFC
Can you capture the FS console in debug level with TPORT_LOG=1 and max sofia debug env set please?
start up FS in the foreground with these env vars.
#TPORT_LOG=1 NUA_DEBUG=9 NTA_DEBUG=9 SU_DEBUG=9 /usr/local/freeswitch/bin/freeswitch
enter: console loglevel debug (or press f8)
reproduce the problem
thx
On Sun, Jun 1, 2008 at 11:09 PM, UV <uv@talknet.com.au> wrote:
Hi Brian,
I might have a blonde moment but I don't understand how you want the raw sip packets... Isn't that just binary data?
I've cleaned my post in pastebin and removed all the Ethernet and tcp/ip data off to make it a bit more readable: http://pastebin.freeswitch.org/4545
I also attach the two capture files here.
Hopefully, one of these is what you mean by raw...
Sorry
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, June 02, 2008 7:13 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Shutting down RFC
Please put up the raw sip packets for us to see. This broken down stuff in wireshark is so hard to read.
/b
On Jun 1, 2008, at 2:22 PM, UV wrote:
It's freeswitch on one side (server) and a Net-820PH/Netcomm V85 IPPhones on the other.
Let me know how can I help troubleshoot it.
If there's a way I can disable the Allow-Events headers, I'll be able to confirm my hypothesis.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Monday, June 02, 2008 3:15 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Shutting down RFC
Is this freeswitch on both sides of the call here? Can i get into the bod live to trouble shoot this?
Mike
On Jun 1, 2008, at 6:35 AM, UV wrote:
I've checked, double checked and triple checked. Both configurations are identical.
Take a look at the pastebin capture: http://pastebin.freeswitch.org/4544 You'll see both calls are exactly the same other than the Allow- Events addition.
Ideas, anyone?
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, June 01, 2008 6:08 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Shutting down RFC
The Allow-Events header isn't part of the SDP. Please double check to make sure you actually have mod_voipcodecs loaded. Check to make sure the codecs are registered via "show codec" at the console.
/b
On May 31, 2008, at 11:46 PM, UV wrote:
I agree with that assumption, Brian, but I have identical IP phone, identical Conf directory, identical codec settings on both sides, identical originate call scenario with two different FS versions and the only thing different on the network packet level is one additional line in the FS SIP SDP.
If you wish, I pasted both calls WireShark captures for you to see at: http://pastebin.freeswitch.org/4544
Let me know what you think.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, June 01, 2008 5:42 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Shutting down RFC
488 means you have no codecs in common. Please review the allowed codecs on both sides to ensure you have at the very least one in common.
/b
On May 31, 2008, at 1:53 PM, UV wrote:
I recently upgraded to build 8697 (v1) but had to downgrade it back after some of my IPPhones returned "488 Not Acceptable Here" for Freeswitch Originate requests.
After a short investigation I found the new build (new Sofia?) is now supporting RFC3265 for specific event notification and every call originated from the switch has now this added to the SDP header: "Allow- Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary" - which, I assume is the cause for the 488 message.
Is there any way to disable this feature?
Many thanks!
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400