Hi. I'm completely new to freeswitch and would like to migrate from asterisk. I have a couple of questions though: 1) I understand I won't be able to use my B410P Digium ISDN (mISDN) BRI card yet, right? 2) my TE120P Digium PRI card should work. 3) can freeswitch work in a master-master cluster? Example: FreeSWITCH-1 --- MySQL nbcluster tables --- FreeSWITCH-2 I suppose I can do this if I enable ODBC in FreeSWITCH. But will both FreeSWITCH boxes get "confused" if they use the "same" clustered database (active/active)? (by the way, does the built-in sqlite support clustered tables?) SIP softphones and ATA phones would register via DNS SRV (or custom method) to any one of the servers but only one (the idea is to load balance registrations). Say, extension 101 registers to FS-1 and 102 registers to FS-2. Could 101 contact 102 simply by looking it up the clustered database (eg. querying the table may yield something like sip:102@FS-2)? This would be similar to Asterisk 1.4+ Realtime SIP registrations. In Asterisk 1.2 I use IAX2 friend trunks between the 2 servers and DUNDi for the lookup. Would queues and ring groups also work in a "clustered" freeswitch system? In case of "queues", where is the caller position stored? Who's counting? FS-1 or FS-2? Could the counter be stored in the clustered SQL table so that both FreeSWITCH servers get it right in case calls come in on both servers? example: PSTN---PRI---FreeSWITCH-1 --- clustered SQL --- FreeSWITCH-2---PRI---PSTN Calls coming in from the PSTN reach both FS-1 and FS-2. I would like these calls to enter a "clustered" queue which bridges calls to operator extensions which can be registered on either server. The caller positions should be coherent across both servers. Has anyone done this already? Thanks for your feedback. Vieri
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!
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!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org
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!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org
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Brian West sip:brian@freeswitch.org
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!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org
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Brian West sip:brian@freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org
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Brian West sip:brian@freeswitch.org
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_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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Brian West sip:brian@freeswitch.org
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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!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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Brian West sip:brian@freeswitch.org
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Brian West sip:brian@freeswitch.org
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!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Brian West sip:brian@freeswitch.org
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Brian West sip:brian@freeswitch.org
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Brian West sip:brian@freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Brian West sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org>
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Brian West sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org>
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Brian West sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org>
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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
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!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Brian West sip:brian@freeswitch.org <mailto:sip%3Abrian@freeswitch.org>
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Brian West sip:brian@freeswitch.org <mailto:sip%3Abrian@freeswitch.org>
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Brian West sip:brian@freeswitch.org <mailto:sip%3Abrian@freeswitch.org> _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Those all look fine. you can manually remove the code that adds the Allow-Events. The patch is : http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=8467 If that fixes it, you should contact your phone manufacturer, as the rfc states that we SHOULD add that header according to rfc 3265 section 3.3.7. Mike On Jun 2, 2008, at 11:35 AM, UV 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!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Brian West sip:brian@freeswitch.org
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Brian West sip:brian@freeswitch.org
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Brian West sip:brian@freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
We already contacted the company and they may or may not fix their phones (my Chinese is not what it used to be :-)) 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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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Brian West sip:brian@freeswitch.org <mailto:sip%3Abrian@freeswitch.org>
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Brian West sip:brian@freeswitch.org <mailto:sip%3Abrian@freeswitch.org> _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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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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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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
On Sat, May 31, 2008 at 11:16 AM, Vieri <rentorbuy@yahoo.com> wrote:
Hi. I'm completely new to freeswitch and would like to migrate from asterisk. I have a couple of questions though:
1) I understand I won't be able to use my B410P Digium ISDN (mISDN) BRI card yet, right?
Correct, we have not gotten to the task of supporting BRI just yet.
2) my TE120P Digium PRI card should work.
Any zaptel compatible card should work with OpenZAP on our beta PRI stack which is able to at least run on euro and DMS dialects.
3) can freeswitch work in a master-master cluster? Example:
FreeSWITCH-1 --- MySQL nbcluster tables --- FreeSWITCH-2
I suppose I can do this if I enable ODBC in FreeSWITCH. But will both FreeSWITCH boxes get "confused" if they use the "same" clustered database (active/active)? (by the way, does the built-in sqlite support clustered tables?)
Every place you can choose ODBC in FS was designed to be able to be shared.
SIP softphones and ATA phones would register via DNS SRV (or custom method) to any one of the servers but only one (the idea is to load balance registrations).
Say, extension 101 registers to FS-1 and 102 registers to FS-2. Could 101 contact 102 simply by looking it up the clustered database (eg. querying the table may yield something like sip:102@FS-2)? This would be similar to Asterisk 1.4+ Realtime SIP registrations. In Asterisk 1.2 I use IAX2 friend trunks between the 2 servers and DUNDi for the lookup.
I know asterisk realtime all to well. I am credited with helping to design it though i believe it was improperly executed from my vision. I personally coded the original res_config* hooks which were later developed into realtime. I also was responsible for making it possible to share SIP registrations over realtime in asterisk http://bugs.digium.com/view.php?id=4832 wow what an ordeal that was ..... You will find by studying the xml_hooks and mod_xml_curl http://wiki.freeswitch.org/wiki/Mod_xml_curl and this overview http://www.freeswitch.org/node/123 that the distributed logic to obtain external information in FreeSWITCH will out perform the realtime in both raw capability as well as in the simplicity of implementation to features gained ratio. If you use a sofia profile with ODBC on multiple machines, if you register to any machine all of them will resolve the correct contact address when trying to make an outbound call. Presence, however such as subscriptions must still go to the original box because there is an open SIP dialog that must be maintained. So if you have box1 .. box10 and you register and it hits box2, you can call this user from all 10 boxes and it will find the correct contact addr.
Would queues and ring groups also work in a "clustered" freeswitch system? In case of "queues", where is the caller position stored? Who's counting? FS-1 or FS-2? Could the counter be stored in the clustered SQL table so that both FreeSWITCH servers get it right in case calls come in on both servers? example:
PSTN---PRI---FreeSWITCH-1 --- clustered SQL --- FreeSWITCH-2---PRI---PSTN
Calls coming in from the PSTN reach both FS-1 and FS-2. I would like these calls to enter a "clustered" queue which bridges calls to operator extensions which can be registered on either server. The caller positions should be coherent across both servers.
Has anyone done this already?
All the things stored in mod_limit, the limit app, the db app etc. Can be shared via ODBC the same way as SIP. For advanced clustering, you can use an external control entity such as mod_event_socket and combinations of mod_xml_curl so you can dynamically generate a dial plan to send the call to PARK state which is a passive state where you can control the channel on a staging box then distribute them to satellite machines running applications or mod_fifo for call queuing .
Thanks for your feedback.
Vieri
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Hi
If you use a sofia profile with ODBC on multiple machines, if you register to any machine all of them will resolve the correct contact address when trying to make an outbound call. Presence, however such as subscriptions must still go to the original box because there is an open SIP dialog that must be maintained. So if you have box1 .. box10 and you register and it hits box2, you can call this user from all 10 boxes and it will find the correct contact addr.
To configure the sofia profile with ODBC, I just need to configure the following flags in the sip_profiles right? <!--If you have ODBC support and a working dsn you can use it instead of SQ$ <!--<param name="odbc-dsn" value="dsn:user:pass"/>--> Is there any script I need to run to create the schema necessary for storing the location info?
All the things stored in mod_limit, the limit app, the db app etc. Can be shared via ODBC the same way as SIP.
For advanced clustering, you can use an external control entity such as mod_event_socket and combinations of mod_xml_curl so you can dynamically generate a dial plan to send the call to PARK state which is a passive state where you can control the channel on a staging box then distribute them to satellite machines running applications or mod_fifo for call queuing .
Could someone elaborate a bit more how to do it? Does it mean I need a dedicated staging server to park all the agents' channels? When the call is PARK on the staging box, how to bridge the PARKed call to an incoming call that reaches the other boxes? I don't totally understand how to "distribute the PARKed calls to satellite machines"... Thanks for your advice. Regards, Pete
participants (6)
-
Anthony Minessale
-
Brian West
-
Michael Jerris
-
Pete Kay
-
UV
-
Vieri