Actually it is NOT a bug in FreeSWITCH, Your provider is broken, If they wish to start the auth over they should put stale=true, now we have to hack sofia to break the rules per the RFC and reset it as if we received a stale. Again NOT OUR BUG. I’m growing ever tired of SIP and all the work arounds we must do to accept an obvious broken behavior. At some point it has to stop and we have to make vendors fix it (i.e. the people you $PAID$ money for a commercial product thats behaving badly). Yet everyone comes to us the FREE software and makes us work around it which is not fair. sigh… -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Apr 9, 2014, at 9:25 AM, Snabel Kabiya <snabel@lexifone.com> wrote:
Hi,
This seems to be a freeswitch bug:
http://jira.freeswitch.org/browse/FS-6287
Thanks, Snabel Kaabiya
On Tue, Apr 8, 2014 at 9:25 PM, <freeswitch-users-request@lists.freeswitch.org> wrote: Send FreeSWITCH-users mailing list submissions to freeswitch-users@lists.freeswitch.org
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Today's Topics:
1. Re: mod_skinny --> Calling numbers with various lenght (MS-R-IT) 2. Re: Delayed sound after answer (Anthony Minessale) 3. Re: Delayed sound after answer (Oleg Stolyar) 4. Re: [ERR] sofia_reg.c:2110 "some gateway" Registration Failed with status Unauthorized [401]. failure #37 (Snabel Kaabiya) (Brian West)
---------- Forwarded message ---------- From: "MS-R-IT" <martin.schoepfer@ms-r-it.de> To: "'FreeSWITCH Users Help'" <freeswitch-users@lists.freeswitch.org> Cc: Date: Tue, 8 Apr 2014 19:58:15 +0200 Subject: Re: [Freeswitch-users] mod_skinny --> Calling numbers with various lenght Hello Nathan,
thanks for your reply can i have checked this hack but i don´t think this is a solution to communicate. Are there any plans to get this function into freeswitch in the future? I check now how I could handle the cisco phones temporary with a small asterisk chan_sccp connected to freeswitch. Is there any way to get chan_sccp transformed to Freeswitch! Because its Opensource too! I only able to program some smalls things in C or C++ so I have to ask if it is any reason and then I can check it by myself!
Thank you for your informations. I think you have luck with NANP!! :-)
Mit freundlichen Grüßen
Martin Schöpfer
____________________________________________________________________________ __________
MS-R-IT Martin Schöpfer Regelungs- und Informationstechnik Bergham 3 83052 Bruckmühl Tel.: 08062 7238870 Fax.: 08062 7238871 Mobil: 0176 80088253
-----Ursprüngliche Nachricht----- Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Nathan Neulinger Gesendet: Dienstag, 8. April 2014 19:41 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] mod_skinny --> Calling numbers with various lenght
On 04/08/2014 12:10 PM, MS-R-IT wrote:
Hello,
yes they can overlap you can have different length with overlap like my company siemens the central has the number 3-digit prefix (yes this can also be) then 4-digit four our site (like 9221) and 0 for our central and 5910 for me directly.
There are many else expressions I could give you emergency call in Germany is 110 and 112 but there can be number with 110XXXXX too!
This is not something you'll be able to do with the current built-in dial handling for mod_skinny currently because of how it processes the dialing.
The key part of the current processing for this is in skinny_handle_keypad_button_message() where it calls skinny_session_process_dest() with the non-null 2nd to last argument (append_dest).
The processing sets call to routing state after each digit in that routine.
What would need to happen is some sort of time delayed callback to have it not send the call to routing state until a certain time has elapsed.
You could do a workaround of requiring a terminating digit (# or something like that) to "dial" - that would work with current code, but is definitely a hack.
-- Nathan
------------------------------------------------------------ Nathan Neulinger nneul@mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect
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---------- Forwarded message ---------- From: Anthony Minessale <anthony.minessale@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Tue, 8 Apr 2014 13:04:48 -0500 Subject: Re: [Freeswitch-users] Delayed sound after answer Its probably based on your network. Are you behind NAT? Sometimes it takes a few seconds for the automatic translation to kick in. You can add a sleep 1000 after answer to ensure you have established media before you play the tone.
On Mon, Apr 7, 2014 at 5:10 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: Hi guys,
I have the following in my dialplan:
<condition field="${<some field>}" expression="<some expression>"> <action application="answer"/> <action application="set" data="api_hangup_hook=curl <some URL>"/> <action application="playback" data="tone_stream://v=0;%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>
Notice the line in bold. In most cases the tone plays fine but in some cases it is shorter than 1 second and in a few cases it does not play at all. Has anyone run into this? Is there a way to make the occasional media delay after answer go away?
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-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+
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---------- Forwarded message ---------- From: Oleg Stolyar <olegstolyar@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Tue, 8 Apr 2014 12:12:42 -0600 Subject: Re: [Freeswitch-users] Delayed sound after answer Thanks Anthony, I'll try that.
On Tue, Apr 8, 2014 at 12:04 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: Its probably based on your network. Are you behind NAT? Sometimes it takes a few seconds for the automatic translation to kick in. You can add a sleep 1000 after answer to ensure you have established media before you play the tone.
On Mon, Apr 7, 2014 at 5:10 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: Hi guys,
I have the following in my dialplan:
<condition field="${<some field>}" expression="<some expression>"> <action application="answer"/> <action application="set" data="api_hangup_hook=curl <some URL>"/> <action application="playback" data="tone_stream://v=0;%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>
Notice the line in bold. In most cases the tone plays fine but in some cases it is shorter than 1 second and in a few cases it does not play at all. Has anyone run into this? Is there a way to make the occasional media delay after answer go away?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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---------- Forwarded message ---------- From: Brian West <brian@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Tue, 8 Apr 2014 13:25:05 -0500 Subject: Re: [Freeswitch-users] [ERR] sofia_reg.c:2110 "some gateway" Registration Failed with status Unauthorized [401]. failure #37 (Snabel Kaabiya) This is a bug in the provider, If they want to start auth over they should say stale=‘true’, this way the NC will start back at 1. not 18e -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Apr 8, 2014, at 7:07 AM, Steven Ayre <steveayre@gmail.com> wrote:
It's not the first request because it's sending a challenge response using a nonce, which it must have received in a previous challenge (401)... It can't just make that up.
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