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mod_spy problem
by Vitali 19 May '13

19 May '13

Hi, I have a problem with mod_spy - it just plays moh and do not actually spy. Here`s my dialplan: /<extension name="user_spy"> <condition field="destination_number" expression="^\89(\d{4})$"> <action application="answer"/> <action application="userspy" data="$1@${domain_name}"/> </condition> </extension>/ When I dial 891000, for example, it`s just plays moh despite the fact that I`m making and receiving calls from user 1000. When I run a command /userspy_show/ in fs_cli, I get the following output: /: 54995932-2a7c-11de-af08-93e49196b898 1 total spy/ with no user supplied for spying. There`s no errors in logs. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-spy-problem-tp7590846.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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Bounty - FS-5446
by Cal Leeming [Simplicity Media Ltd] 19 May '13

19 May '13

Hello all, I have thrown up a bounty on the following ticket; http://jira.freeswitch.org/browse/FS-5446 Any takers? Thanks Cal

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Complex IVRs using data from MySQL
by Telecube - John 19 May '13

19 May '13

Ok, if someone could point me in the right direction I would be very grateful. I need to integrate database lookups into multi level IVRs to gather call routing info. The lookups are for variable multi digit keypress data as well as multiple sets of routing information. So far I can see my options are xml_curl or hand the call off to a lua script to handle the processing. I can catch the multi digit keypresses with regex and give that detail to a lua script, no problem. I'm hoping to be able to use the mod_ivr process as much as possible but I'm struggling to see how I can get data back into the call flow after dishing the duty off to lua. Can anyone point me down the right path please? Should I be happy to pass it off to lua and finish the lookups and routing in there? Should I try to keep as much as possible in the core ivr? Thanks.. - John

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Detecting Errors when dialing through multiple gateways
by Sean Devoy 18 May '13

18 May '13

Hi All, I have been having pretty good luck with redundancy by dialing through multiple gateways like: <action application="bridge" data="sofia/gateway/cheap/1xxxxxxxxxx, [leg_delay_start=8]sofia/gateway/expensive/1xxxxxxxxxx"/> If gateway cheap fails, the call goes through on the expensive gateway. That is fine, but I would like an email notification of the failure on the cheap gateway. Is there any way to do that? The support group at the gateway vender is very good at correcting the problem when notified. I would love to send them an email of failures automatically. Thanks, Sean

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FreeTDM PRI channel monitoring
by Ashish gautam 18 May '13

18 May '13

Hi, I want to get the status of each PRI channel through event socket. I am sending api command "api ftdm dump <span> <channel>" to get the status of each channel one by one. Its returning lots of other information also which I do not want. Is there any other efficient and recommended way to do this? Regards, -- Ashish

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advise to troubleshoot a call
by Philippe Le Toquin 18 May '13

18 May '13

Hello, Recently I (well my wife) is having voice problem when she is on a call. Typically she can hear people but they can't The problem is that it is not all the time and when it happens it last a few seconds then all back to normal... then later on it happens again. I am still in the process of trying to find if there is a pattern in the problem but I would like to be able to look at the log from FS. What would you recommend to be able to look at the log (what debug level). Any tips to make it easier to identify an issue in the log (like message to look for?) Thanks Philippe

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Ekiga and feature codes
by Michael Mol 17 May '13

17 May '13

One thing that is not at all clear to me is how to get a SIP client such as Ekiga to be able to use feature codes. For example, I have #300 for parking calls, and #301 for retrieving them...but I cannot for the life of me figure out how to get Ekiga access to those features. I've tried: 1. Putting the call on hold, and then using the dialpad. 2. Transferring the call to the feature code address (looks to the remote user like the call was cleared) 3. Dialing #301 from Ekiga results in me hitting an auto-attendant wishing me to dial my party's extension... I'm really not sure how to proceed...

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Freeswitch create bad SIP header
by ik 17 May '13

17 May '13

Hello, I'm new to freeswitch, and I did google this, but can't find an answer that works for me. I have created the following configuration (both at Asterisk and FS sides): https://gist.github.com/ik5/1a0767f69b8065e1c8b8 * The IP addresses are hidden and replaced with "asterisk.example.com" and " freeswitch.example.com" As you can see there with the SIP header, freeswitch create the "From" and "To" field with the same IP/Domain address, and there is no "contact" field to make it contact properly. So what am I missing or doing wrong ? Thanks, Ido

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No audio after resuming from hold
by Carlos Flor 17 May '13

17 May '13

I have 2 soft-clients registered to a FS box. One calls the other and the call is connected. Both parties can hear each other. Party A puts Party B on hold. The soft-client for each party indicates that the call is on hold. Party A takes the call off of hold. Both clients update their displays to indicate that the call is no longer on hold, however neither party can hear audio. This only happens when I have proxy_media set to true (this is a requirement because I am using zrtp and can't use trusted mitm). I have a very simple dialplan setup to test this scenario. <condition field="destination_number" expression="^(1703555000[3|5])$"> <action inline='true' application="export" data="dialed_extension=$1"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="proxy_media=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="user/$1"/> <action application="hangup" data=""/> </condition> I did several packet captures and watched the SDP packets and subsequent RTP packets. During the call, Phone A sends his audio port to FS, FS sends it's B-Leg audio port to Phone B, Phone B responds to FS with it's audio port, and FS finally responds to Phone A with it's A-Leg audio port. RTP packets flow between these ports and everything works. When the call is placed on hold and then taken off of hold, the same process happens, with the clients usually choosing their same ports as before, and FS choosing new ports for each leg. Each client then sends RTP packets to the newly chosen FS audio ports for their corresponding leg of the call. However, FS sends ICMP port unreachable packets back to each phone and no audio passes. I am happy to provide any information (logs, packet captures, etc...) if that will help, just tell me specifically what you would like. I'm running FreeSWITCH Version 1.5.1b+git~20130424T213558Z~eeeb4fa445 (git eeeb4fa 2013-04-24 21:35:58Z). Thanks for any help.

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Skype 4.1 skypopen compatible?
by JoZu 17 May '13

17 May '13

I read this in new Skype 4.1 client for linux: Implemented support for secondary logins Now you can run more than one instance of Skype on Linux simultaneously. Just add the “– secondary” command line option to launch a second client. Its possible to implement this new option in skypopen with Sebastian trick? PD: Read skypopen wiki and the Skype 4.0 ! (unsupported) by Sebastian Fiorentini http://wiki.freeswitch.org/wiki/Skypopen#Skype_4.0_.21_.28unsupported.29 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skype-4-1-skypopen-compatible… Sent from the freeswitch-users mailing list archive at Nabble.com.

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