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exec_after*
by Jeroen C. van Gelderen 18 Oct '10

18 Oct '10

Hi, My helpdesk extensions are recorded. When the recording is complete (after bridge) I want to run a script to create a ticket and attach said recording to it. Initially I tried to use [...] Dialplan: FreeTDM/3:1/ Action set(exec_after_bridge_app=system) Dialplan: FreeTDM/3:1/ Action set(exec_after_bridge_arg=/root/wav2ticket.php) [...] but this doesn't result in an invocation of my script and seemingly no helpful diagnostic messages. I changed to using <action application="set" data="api_after_bridge=system /root/wav2ticket.php ${recfile}"/> which DOES work. Can anyone clarify the difference between the two approaches? Cheers, -Slim -- Jeroen C. "Slim" van Gelderen Olympic Sports Data Services Email: slim(a)thegreek.com Phone: +1 876 953 6182 x128

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FreeSWITCH Conference Call This Wednesday Oct 20
by Michael Collins 18 Oct '10

18 Oct '10

Hello all! Here is the agenda page for this Wednesday's call: http://wiki.freeswitch.org/wiki/FS_weekly_2010_10_20 We didn't have a speaker scheduled for Wednesday so I will spend a few minutes talking about the wiki. Please bring your wiki questions. In fact, if you can please send me your wiki questions in advance so that I can research them. If you have any agenda items for the call this week please put them on the list. Thanks! -Michael

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problem when recording conferences
by covici＠ccs.covici.com 18 Oct '10

18 Oct '10

Hi. I am having a problem with conference recordings using auto_record. Now it does record, but with some users if the user is silent for a while, then the first so many milliseconds of his first word cuts off. Is this a function of the energe setting, waste setting or how else can I prevent this? Thanks in advance for any suggestionns. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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problem with freeswitch and Cisco 7960
by Michael W. Lucas 18 Oct '10

18 Oct '10

Hi, I'm using Monday's FreeSWITCH build, with a config originally loaded 19 August 2010. I'm working my way through the FS 1.0.6 book. I have two phones, one Zoiper soft phone and one Cisco 7960. Both register fine, and the Cisco can call Zoiper. Zoiper cannot call the Cisco, however. The Cisco phone sends its register requests from UDP port 50790. When FreeSWITCH sends an INVITE to the Cisco, it tries to contact UDP/50790, and the phone returns an ICMP "destination unreachable." This same phone is also registered to an Asterisk box. When I call that number, asterisk connects to UDP 5060 on this phone. The phone replies from a high-numbered port to 5060. It seems I need some option to tell FreeSWITCH to send the INVITE to 5060? Or am I missing something? Thanks, ==ml -- Michael W. Lucas mwlucas(a)BlackHelicopters.org http://www.MichaelWLucas.com/, http://blather.MichaelWLucas.com/ New book available: Network Flow Analysis http://www.networkflowanalysis.com/

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Binding to virtual interface (ipsec vpn)
by Nicolas Brenner 18 Oct '10

18 Oct '10

Hello everyone, I'm having trouble binding FS to another ip on a virtual interface (eth0:3). The virtual interface's ip address is the gateway to an ipsec vpn, and the only way to access this voip provider's sip server, is through the vpn. I tried setting up an additional sip profile, with rtp-ip and sip-ip as the virtual interface's ip, but when I do that, no packets are sent at all when trying to make a call with originate through the cli (according to tcpdump). I tried also setting ext-rtp-ip and ext-sip-ip to the same virtual interface's ip, but nothing happened, still no packets being sent (according to tcpdump) when trying to make a call. Finally I tried only setting ext-rtp-ip and ext-sip-ip to the virtual interface's ip, and leave rtp-ip and sip-ip to $${local_ip_v4}, then tcpdump started seeing some traffic (when trying to originate a call) which I was able to capture. When I loaded the trace on wireshark though, I noticed the source ip being used to send all the (sip) packets, is the one from eth0 (public ip), and not eth0:3's ip (private ip for vpn). Has anyone had any similiar experiences? how were you able to solve it? Thanks for your help! Nico

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mod_cepstral 5.1 known issues?
by Marc Kellem 18 Oct '10

18 Oct '10

The mod_cepstral wiki page says "Warning: the suggested version to use is 4.x since there are known issues with 5.1 (which is closed source)". What exactly are the known issues? http://wiki.freeswitch.org/wiki/Mod_cepstral Thanks, Marc Kellem

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Mod_h323 problem
by John Carpenter 18 Oct '10

18 Oct '10

Hi, I am trying to bridge an incoming h323 call to an external SIP provider, I am using latest git release. If I use G729 codec I get message "Unsupported ptime of 2 on write Audio codec G.729{sw} for connection [0xb4c0ee10]" and call fails to connect. If I use ulaw codec call fails to connect with "Write PDU fail: no control channel" If I make a straight SIP to SIP call though same provider all work ok. I have posted log of call in http://pastebin.freeswitch.org/14216 because it is rather large. my h323.conf.xml file looks like this <configuration name="h323.conf" description="H323 Endpoints"> <settings> <param name="trace-level" value="10"/> <param name="context" value="public"/> <param name="dialplan" value="XML"/> <param name="codec-prefs" value="PCMA,PCMU,GSM,G729"/> <param name="use-rtp-timer" value="true"/>  <param name="rtp-timer-name" value="soft"/>    <param name="gk-address" value=""/>  <param name="gk-identifer" value=""/>  <param name="gk-interface" value=""/>  <param name="gk-retry" value="30"/>  <param name="faststart" value="true"/>  <param name="h245tunneling" value="true"/>  <param name="h245insetup" value="true"/>  <param name="jitter-size" value="60"/>  <param name="progress-indication" value="8"/>  <param name="alerting-indication" value="8"/>  <param name="endpoint-name" value="fs"/> <param name="fax-old-asn" value="true"/> </settings> <listeners> <listener name="default"> <param name="h323-ip" value="$${local_ip_v4}"/> <param name="h323-port" value="1720"/> </listener> </listeners> </configuration> And this is the log of the xml_curl dialplan that is executed <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="php dialplan"> <context name="public"> <extension name="normal"> <condition field="destination_number" expression="^(\d+)$"> <action application="set" data="bypass_media=false"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="sofia/gateway/arbinet-o/$1"/> </condition> </extension> </context> </section> </document> Any help will be greatly appreciated regards, John Carpenter

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Freeswitch Video Capture and Playback
by Esben Stien 18 Oct '10

18 Oct '10

I'm using ekiga with mod_fsv, trying to record and play back video. When I dial the record extension, it seems to record something, as the video file gets bigger. Trying then to dial the extension for play back, just hangs up, with freeswitch saying: od_fsv.c:247 File version does not match! There seems to be no information on the FSV format or the mod_fsv module on the wiki. Is this at all supposed to work?. What clients and codecs were successful?. Any pointers as to what I can try?. -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n

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mod_erlang_event core dump on load with shortname=false on Mac
by Seven Du 18 Oct '10

18 Oct '10

Hi, I experience this a few days ago, everything is ok with shortname=true, but shortname=false causes immediately core dump on load. I also pulled the latest git. problem remains. It has nothing to do with the latest change. The last change does help. I set <param name="nodename" value="freeswitch(a)192.168.1.128"/> then no more crash. So, the problem might be because it has trouble on get the fully qualified domain name on my box. Any other experience on this? I can make a jira if confirmed a bug. Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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some questions of freeswitch & X-Lite
by 江 Chiang 至中 Chih-Chung Wybie 18 Oct '10

18 Oct '10

Hello I set two SIP servers 60.248.175.37 & 60.248.175.38 and use X-Lite to call my question is how can I call between these servers? if I can , how shoould I set in *.xml? or X-Lite? thank you for your attension Best Regards Gary

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