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FreeSWITCH process dies
by Sirish Masur Mohan 06 Feb '13

06 Feb '13

Hi, In last 2 days, I have observed that FreeSWITCH process, after being up for about a day or so, dies unexpectedly. This has happened twice, on two different servers. There is no indication of failure in freeswitch log. I am using a 32-bit Ubuntu, with FreeSWITCH version being FreeSWITCH Version 1.3.13b. Any clues on how to debug this? Thanks! With regards, Sirish

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407 proxy authorization, 302 redirect
by Miha 06 Feb '13

06 Feb '13

Hi, I set manul-redirect in sip_profiles and when compes 302 FS handles it like it should (I have configured my dialplan for handling manul-redirects). After looking with whireshark I noticed that 407 is not happening when it goes for 302. How to deal whit this issue so that every 302 will be also authorized? thanks! miha

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DTMF both inband and RFC2833?
by yyu0378 06 Feb '13

06 Feb '13

hi: I use peer reg at fs.If peer only send inband DTMF signal,fs passthrough it.But if peer send both inband and RFC2833,then fs can`t passthrough signal.I try add start_dtmf at dialplan,no change.This is bug?Or how I can pass both them?Thank you!

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GIT Is Back.... Upgrades complete
by Ken Rice 05 Feb '13

05 Feb '13

Hey Guys, The GIT Server is now upgraded and back online, Special Thanks To Ray ³[intra]lanman² Chandler for taking care of this stuff... K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Get recorded file from callcenter via API.
by Mimiko 05 Feb '13

05 Feb '13

Hello. How to retrieve a recorded audio file by callcenter using API call to FS? Or by other meaning except using direct file access. -- Mimiko desu.

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SPA-2102 Gives 486 instead of Call Waiting
by Avi Marcus 05 Feb '13

05 Feb '13

Here's an intersting one that seems to have no results on google or in debug info: I have an SPA-2102 (firmware 5.2.10) that returns 486/Busy Here when a call comes in to the active line. "call waiting" is turned on, so it should be beeping in instead... I got these logs off the SPA but it doesn't seem very helpful: (and there should be more, I think) ==> /var/log/syslog <== Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]<<$FreeSWITCH: 5060(1148) Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]<<$FreeSWITCH: 5060(1148) Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]->$FreeSWITCH: 5060(312) Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]->$FreeSWITCH: 5060(312) Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il SIP/2.0 100 Trying Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il #012 Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il #012 Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]->$FreeSWITCH: 5060(338) Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]->$FreeSWITCH: 5060(338) Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il SIP/2.0 486 Busy Here Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il #012 Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il #012 Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il [0]<<$FreeSWITCH: 5060(334) Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il [0]<<$FreeSWITCH: 5060(334) Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 I saw one suggestion to do a factory reset that fixed the problem, but that's a bit troublesome at the moment since the client can't load provisioning files (dumb network.) -Avi Marcus

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NAT traversal - the final say..!
by Cal Leeming [Simplicity Media Ltd] 05 Feb '13

05 Feb '13

*Any and all feedback on this thread would be much welcomed.* Hello, * * There seems to be a large number of discussions surrounding NAT traversal, as well as lots of documentation, but with no concrete answers. The NAT related wiki documentation is tedious, and depending on the outcome of this thread, I'd like to spend some time cleaning it up. The most common problem (the same as ours) was having a router with broken ALG and a softphone that does not seem to work with STUN. The following REGISTER is sent from a phone. REGISTER sip:1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:57787 ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:2000@192.168.1.102:57787;rinstance=0c7190b115a36513> To: "foxx"<sip:2000@1.2.3.4:5060> From: "foxx"<sip:2000@1.2.3.4:5060>;tag=83311448 Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI. CSeq: 7 REGISTER Expires: 120 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Supported: replaces User-Agent: 3CXPhone 6.0.25732.0 Content-Length: 0 As you can see, the client's public IP is not specified anywhere. FreeSWITCH offers several ways around this, the main ones being; * NDLB-connectile-dysfunction * NDLB-force-rport * apply-nat-acl * sip-force-contact The one that has worked in our case was "NDLB-connectile-dysfunction" (otherwise known as NAT HACK), however there seems to be a lot of negative comments about using this. >From what I can tell, the general argument is that NAT HACK is considered a non RFC compliant hack, and the SIP phones should be doing a better job of keeping to the RFCs. In principle, this is a fair argument - but in practise, it's not a reasonable assumption that all phones are RFC compliant, and (imho) not a reasonable argument to have this functionality disabled by default. So, I'd like to present the following arguments; * Are there any other negative aspects about using NDLB-connectile-dysfunction, other than it is a non compliant RFC hack? * Why is NDLB-connectile-dysfunction not enabled by default when certain conditions are met? In the event that FreeSWITCH receives a REGISTER from a phone specifying a Contact/Via as 192.168.0.0/16, but received on a public IP, then it should be obvious that NAT is broken and automatically try to circumvent it. * People seem to get confused between server side and client side NAT problems, and that they both need to be resolved in a different way. The documentation doesn't seem to reflect this clearly.

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Fwd:
by Thangappan M 05 Feb '13

05 Feb '13

http://www.cluae.com/joomla/components/com_weblinks/5pocb1.php?s=lf

1 0

Dialpan: Hunt Group which never stops - continue when no answer from all extensions
by K K 05 Feb '13

05 Feb '13

Hi all, I have my hunt group which is working ok, but when there will be no answer from the last extension it stops and my client gets the "temporary unavailable" response. I would like to achieve that my hunt group will never stop, it means that when the last extension will not accept the call, then again the first one will be called. I have tried with the options originate_continue_on_timeout=true and continue_on_fail=true but without any effect, the behaviour was still the same. Thank you for help.

2 2

DTMF duration unit
by François Delawarde 05 Feb '13

05 Feb '13

Hi all, I don't clearly get what unit we should use in dtmf-duration parameters (core min-dtmf-duration / max-dtmf-duration and sofia dtmf-duration). The variable name and some parts of the code/wiki imply it's in milliseconds (for example in sofia.c, we can see a line saying profile->dtmf_duration = 100;). However, the default values in switch_types.h suggest the values should be based on timestamp like in RTP. Something like: minimum = 400 @ 8kHz => 50ms default = 2000 @ 8kHz => 250ms maximum = 192000 @ 8kHz => 24s If the later is the right one, should I change values when using WB codecs that are NOT sampling at 8kHz? Also, is the "unit" consistent across all endpoint modules? Thanks, François.

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