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NO VOICE BETWEEN TWO REMOTE ENDPOINTS
by M.Emran 07 Jul '08

07 Jul '08

Hi, I have configured FS like this.but voice is not transmitting between two endpoints.Can you help me where i did wrong ? Here is my configuration: - <?ui=2&view=bsp&ver=1qygpcgurkovy#11afef97be4488a0_11afef267e7502b2_> <profile name="internal" domain="$${domain}"> <aliases> <alias name="$${domain}"/> <alias name="default"/> </aliases> <gateways> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/> </gateways> <domains> </domains> <settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="rfc2833-pt" value="101"/>  <param name="sip-port" value="60"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="aggressive-nat-detection" value="true"/>  <param name="rtp-ip" value="$${local_ip_v4}"/>  <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="$${hold_music}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5061"/> <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <param name="tls-version" value="tlsv1"/> <param name="inbound-no-media" value="true"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> -- Regards ---------- M Emran Inspiration Software Ltd Web: www.inspiresoftbd.com

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FS in Windows Vista: Please help with new set up...
by Kin Quek 07 Jul '08

07 Jul '08

Hi All, I finally got FS compiled on Windows Vista (with Visual Studio 2008 Express). I am ready to give it a test run. Since I am a newbie to FS, any suggestion of a test set up will be appreciated. I check the FS web-site but found nothing (examples) applied to Windows. May be I looked at the wrong area, can anyone help? Thanks in advance! Kin

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NO VOICE BETWEEN TWO ENDPOINTS UNDER NAT
by M.Emran 07 Jul '08

07 Jul '08

Hi, I have configured FS like this.but voice is not transmitting between two endpoints.Can you help me where i did wrong ? Here is my configuration: - <#11afef267e7502b2_> <profile name="internal" domain="$${domain}"> <aliases> <alias name="$${domain}"/> <alias name="default"/> </aliases> <gateways> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/> </gateways> <domains> </domains> <settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="rfc2833-pt" value="101"/>  <param name="sip-port" value="60"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="aggressive-nat-detection" value="true"/>  <param name="rtp-ip" value="$${local_ip_v4}"/>  <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="$${hold_music}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5061"/> <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <param name="tls-version" value="tlsv1"/> <param name="inbound-no-media" value="true"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> -- Regards ---------- M Emran Inspiration Software Ltd Web: www.inspiresoftbd.com

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NAT issue with registration responses going to the wrong port
by Alex Kinch 07 Jul '08

07 Jul '08

Hi gang, Got a strange problem with a NAT'd device that's trying to connect to FS. The first REGISTER arrives, without auth, and FS responds with 401 Unauthorized to the correct port on the external IP address of the firewall. However when the second attempt to REGISTER, this time with auth, comes in, FS responds with 200 OK to the local SIP port specified on the device. The device obviously doesn't get the 200 OK so keeps attempting to register. Any ideas? Tried sip-force-contact but that doesn't seem to help. Alex

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Problems with Nokia E51 registration
by jussi.salonen＠huittislainen.com 07 Jul '08

07 Jul '08

Hey, I can't register my Nokia E51 with FS. I have default settings after installing FS (username:1000 passw:1234). I have x-lite on my laptop and it works just fine over my wlan with FS. E51 is in same wlan and I can use that normally for web browsing etc. so I really think there is something wrong in my settings. If some of you have same phone or same kind of problems please let me know? ---------------------------------------------------------------- This mail sent through L-secure: http://www.l-secure.net/

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FS Support for LinuxMCE
by EdPimentl 07 Jul '08

07 Jul '08

LinuxMCE now supports *sterisk ..... however it would be better with FS. LinuxMCE uses an online(avatar) person to walk one through the configuration of asterisk voip in their home. http://video.google.com/videoplay?docid=2176025602905109829&hl=en LinuxMCE is installed on the Kubuntu OS, and utilizes open source applications such as Asterisk, Xine, MythTV, VDR, Firefox, VideoLAN and SlimServer. OS: Kubuntu TV/PVR: MythTV VoIP : Asterisk (Better alternative would be FS) HomeAutomation: Pluto Serveillance Camera Recording: Motion Is anyone currently working on FS support for LinuxMCE..?

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Time out issue and static noise with originate command
by Diego Viola 07 Jul '08

07 Jul '08

Hi all... I have a user account on FS, sometimes when I register with Zoiper or X-Lite it works, but after some time it unregisters itself, I'm not sure why... this didn't happened on Asterisk for me. Does FS handles NAT automatically? This is how I have it setup: softphone => NAT (router) => FS In Asterisk I used to have nat=yes in sip.conf, is there a equivalent for this in FS? Why could my softphone be deregistering itself? This is my user account on FS: Maitei:/usr/local/freeswitch# cat conf/directory/default/diego.xml <include> <user id="diego" mailbox="1000"> <params> <param name="password" value="1234"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> </variables> </user> </include> Another issue... when I use the originate command, with a WAV or GSM file... all I get is static noise, is there a way that this could work?, this also works on Asterisk just fine for me: originate sofia/internal/diego &playback('/root/nin.mp3.wav') But I get static noise on FS. Thanks, Diego

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Rerouting bridge based on error response from far end
by Joseph Bajin 07 Jul '08

07 Jul '08

I will use xml_curl to query a db for the cheapest routes to use. It will come back with a couple of different gateways to use. (LCR idea) Based on the response I get from the first gateway (404,503,500, 486), I want to either reroute to another gateway or send back to the originator the result that I found (486 in this case, other messages should be rerouted unless we hit the last and then send back a final response). How can I do that in a dialplan? Can you point me some documention maybe?

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help with: [ERR] mod_local_stream.c:266 local_stream_file_open() unknown source moh
by Diego Viola 07 Jul '08

07 Jul '08

Hi When I dial a conference I get this: 2008-07-07 00:19:45 [ERR] mod_local_stream.c:266 local_stream_file_open() unknown source moh I think I have all the MOH files: http://pastebin.com/m6a463dfa http://pastebin.com/m7f24dce Maitei:~# grep hold_music /usr/local/freeswitch/conf/vars.xml <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/> Sorry if I'm missing something... I'm still very new in FreeSWITCH, but I'm really enjoying the experience. Thanks, Diego

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Invalid Application bridge
by Grey Man 06 Jul '08

06 Jul '08

Hi, I've just installed FreeSwitch on Windows XP and have been able to get a SIP account registered and placing calls to the FreeSwitch server. The problem I'm having is when I try and place an outgoing call I get an error message of: "switch_core_sessions.c:964 switch_core_session_execute_application() Invalid Application bridge" The dial plan I am using is: <include> <context name="default"> <extension name="test"> <condition field="destination number" expression="^.*$"> <action application="bridge" data="sofia/sip/300(a)sip.mytestserver.com" /> </condition> </extension> </context> </include> It seems as if the "bridge" application cannot be found. Is there anything I can check to see why that may be so? Regards, Greyman.

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