July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

This is "G"'s extensible way of serializing structured data
by EdPimentl 08 Jul '08

08 Jul '08

FYI http://code.google.com/apis/protocolbuffers/docs/overview.html What is PB? *Protocol Buffers* – A language-neutral, platform-neutral, extensible way of serializing structured data for use in communications protocols, data storage, and more. -E

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Wednesday meeting moved to 9AM Central, 3PM GMT
by Brian West 08 Jul '08

08 Jul '08

We have decided to move the meeting earlier since 5PM resulted in very few showing up. This also gives us a chance of more people in Europe to join in and take part. We could also Schedule some so Asia can get involved also. So Wednesday at 9AM CST, 3PM GMT on the July 8th. Any thoughts? Brian West sip:brian@freeswitch.org

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3-way call
by Juan Backson 08 Jul '08

08 Jul '08

Hello I am new to Freeswitch. Can someone show me an example on implementing a 3-way calling feature within the dialplan? Regards, Juan

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mod_xml_cdr: reporting back errors from curl posts
by Faraz R. Khan 08 Jul '08

08 Jul '08

I have made a PHP script which processes the XML and feeds it into my DB. Is there anyway to return error codes from the PHP script? die() or exit(1) or echo 'Error' do not work. Is mod_xml_cdr expecting some defined format or a return? I am testing it by observing the console. I have also set log-err path in xml_cdr.conf to / but XMLs are not being generated even if I deliberately cause an error on the PHP side. btw- LOVE the 'callflow' part of the xml tree! simply amazing. My script can 'follow through' calls thanks -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz

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endpoint status in JS
by Ghulam Mustafa 08 Jul '08

08 Jul '08

i want to get status(presence) of an endpoint from within JS/Python. is it possible? i didn't find anything related to this in docs. Regards, -- Ghulam Mustafa <gmustafa(a)emergen.biz>

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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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