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SRTP
by Mitch Johnson 07 Mar '11

07 Mar '11

When I dial 9664 to test the tls/srtp configuration it says that the call is secure, however, when I dial another phone configured for tls/srtp the call doesn't go through, the automated attendant comes online to say that the extension is not available and then puts the call to voicemail. I read the Secure RTP wiki, I do see a similar entries in the dialplan under the extension name global, for both inbound and outbound: <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/>  <action application="export" data="sip_secure_media=true"/> </condition> So if I can make the test call to 9664 on both phones, which I assume is using the inbound part by connecting the call. Any help in figuring this out would be greatly appreciated. BTW, I did buy the book, but there's no mention of SRTP/TLS in there. Thanks, Mitch

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Recent changes to fs_cli
by peely 07 Mar '11

07 Mar '11

Hi, It looks like there's been a recent change to fs_cli which catches CTRL + C's and forces them to be CTRL + D's or specific commands. I quite often do an fs_cli > output.txt for example, then use nano to investigate test calls. It seems that now a CTRL + C doesn't end the command and I have to kill my connection. THis used to be quite useful so I wonder if it would be possible to have this functionality back? Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recent-changes-to-fs-cli-tp60… Sent from the freeswitch-users mailing list archive at Nabble.com.

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continual registers
by lloyd thomas 07 Mar '11

07 Mar '11

Just notice that my polycom 550 seems to register every 15 secs. Thought reg.1.server.1.expires="600" on polycom config would fix it, seems not. Also tried my xlite client, which also displays the same tendency. Both are behind NAT'ed networks but FS box is on public IP address. Can I extend the register timeout and how? Is it wise to?

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tell FS use native files
by Madovsky 06 Mar '11

06 Mar '11

I'd like to use spx files for voicemail and all hold music I tried to change voicemail.conf.xml "wav" to "" or "spx" but no effect. which settings I need to change to force FS use native files Thanks

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trunk ping
by Madovsky 06 Mar '11

06 Mar '11

I have a trunk that auth by IP so in FS I set auth like <param name="register" value="false" /> butt ping in log show [WARNING] sofia.c:4028 Ping failed trunk_3 with code 408 - count -1/-1/1, state DOWN however FS is using this trunk without any problem. is it a normal message behavior ? Thanks

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srtp clarification
by Mitch Johnson 06 Mar '11

06 Mar '11

My previous post may have suggested that the TLS/SRTP was not working. Where in fact, the TLS works like a charm. The problem comes in when I require SRTP only on the phones. When SRTP s turned off it works great, and so does TLS. I've been trying to understand how the voice part of the call is setup using SRTP. When I go through the logs, I don't see anything that says that SRTP failed anywhere. I'm pretty sure it's somewhere in my configuration. In Asterisk I had to define the transport mechanism of tls and encryption=yes to make it supposed to work. But then I never got it working there either, the difference with Asterisk is that it was showing SRTP as failing, but there's a bug causing that so it was pretty much a brick wall for me. Am I supposed to do something under the user profile or somewhere else where that call is encrypted using SRTP? I followed the TLS and SRTP guides to do the setup. Any help on this would be greatly appreciated. As with any problem, it's consuming my life until I can sort it out. Thanks so much ahead of time, Mitch

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writing to file in javascript
by gigerlin 06 Mar '11

06 Mar '11

Dear Freeswitch users, I am developing some javascript applications that run within Freeswitch. I can read without difficulty from files located on the same machine as Freeswitch, but I can't write or create new files. The following code issues an error when trying to open a file in "write mode". var fd = new File("test.txt"); fd.open("write"); fd.write("Hello\n"); fd.close(); The error I got is: 2011-02-18 11:07:25.893733 [ERR] test.js:7 Error: File operation open failed I thought it was a sort of sandbox model issue. However I could generate a file using the following system command: session.execute("system", "echo Hello > test.txt"); The same error occurs when I tried to use the javascript function: var tmp = fetchURLFile("http://myserver/test.html", "test.txt"); This creates an empty test.txt file. I tried then to run the following system command: session.execute("system", "wget -O test.txt http://myserver/test.html"); This as well creates an empty test.txt file. If I run "wget -O test.txt http://myserver/test.html" directly from the system prompt this correctly downloads and creates the test.txt file. So my questions are: what did I miss? Is there some parameter to configure that I haven't found? Thank you in advance for your help, Gilles Gerlinger PS: Freeswitch is running over Ubuntu server 10.10 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/writing-to-file-in-javascript… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Pastebin
by David Villasmil 06 Mar '11

06 Mar '11

Hello people, Why is it that even though I'm logged into freeswitch.org, when I click on the pastebin link, it asks for user/pass and doesn't recognize my credentials? it's very annoying. Thanks David

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Voice quality of bridged call comparing to direct connect
by Yihui Li 06 Mar '11

06 Mar '11

Dear all, I have a accounts with sipgate.co.uk. When I connect directly form X-lite to sipgate with only PCMA codec. The voice quality is good. Recently I set a FS server on Linode VPS and use FS to bridge call from sipgate to X-lite on my computer, The voice quality become much worse with lots of noise. If the call is bridged through skypopen to my skype account, the quality is much better. If the noise is cause by my network speed or PCMA codec, how can sipgate's server avoid it? Is there any way to reduce noise of the bridged call? Thanks.

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Old hung calls
by DJB International 05 Mar '11

05 Mar '11

I am running FS git-deec244 2011-02-24 10-22-47 -0600. I ran into a problem the other day when our ISP was down badly for couple hours (fortunately after midnight). The problem was that after the internet went back up; there were so many old hung calls before the internet got cut off in the core.db (calls & channels table) database. Here are my questions: - What is the proper way to clean those calls up, or will FS clean those calls out from database automatically? - If I decided to restart FS which I know it will clean up those calls, will those calls get written to mod_cdr_csv? Here is my sip profile: <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="routing"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="$${internal_auth_calls}"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="auth-all-packets" value="false"/> <param name="disable-transfer" value="true"/> <param name="disable-register" value="true"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-bypass-media" value="true"/> <param name="apply-inbound-acl" value="domains"/> </settings> Thank you, -djbinter

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