May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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...

3 2

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

3 5

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.

1 0

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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originate_timeout
by Andrew Cassidy 17 May '13

17 May '13

Hi guys, is it originate_timeout or origination_timeout? Both have almost identical wiki pages: http://wiki.freeswitch.org/wiki/Variable_originate_timeout http://wiki.freeswitch.org/wiki/Variable_origination_timeout -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

2 1

server side features with Polycom soundpoint phones
by Jason Holden 17 May '13

17 May '13

Does anyone have any documentation on this specificly for dnd and cf?

2 1

Re-Invite t38 is incomplete when t38_passtrough=true
by Antonio Silva 17 May '13

17 May '13

Hi, i have the following scenario when sending a fax: ATA => FS => Provider => destination The ATA tries to send a fax to destination and in the middle of the call there is the re-invite in t38 from the provider. In dialplan i have: <action application="export" data="t38_passthru=true"/> The invite from the provider is: INVITE sip:gw+TRUNKIP@10.10.1.254:5060;transport=udp;gw=TRUNKIP SIP/2.0 Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bKno6j3r305os1nm4ph2a1sb0000010.1 Call-ID: badda7ee-3980-1231-8db4-b4b52f5e67f8 From: <sip:4774@10.10.1.1:5060>;tag=dg2aai02-CC-52 To: "5313" <sip:nousername@10.10.1.254:5060>;tag=t6DB42KFXarmN CSeq: 2 INVITE Contact: <sip:4774@10.10.1.1:5060;transport=udp> Max-Forwards: 69 Content-Length: 437 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 38317113 38317116 IN IP4 10.10.1.2 s=Sip Call c=IN IP4 10.10.1.2 t=0 0 m=image 29090 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy m=audio 34264 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=silenceSupp:off - - - - a=ecan:fb on - a=X-fax a=fmtp:101 0-15 And the invite sent to the ATA is: INVITE sip:5313@192.168.1.97:5060;ref=5313 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.254;rport;branch=z9hG4bK87p0gpcv4p31g Max-Forwards: 70 From: <sip:4774@sipdomain.net>;tag=mKN5ZFeQ19XDr To: "5313" <sip:5313@sipdomain.net>;tag=5de97158a7c051b0o0 Call-ID: 6aa68878-1a412510(a)192.168.1.97 CSeq: 44056889 INVITE Contact: <sip:4774@10.10.1.254:5060;transport=udp> User-Agent: suvoz Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 291 v=0 o=FreeSWITCH 1368772703 1368772705 IN IP4 10.10.1.254 s=FreeSWITCH c=IN IP4 10.10.1.254 t=0 0 m=image 13196 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:500 a=T38FaxMaxDatagram:500 a=T38FaxUdpEC:t38UDPRedundancy As you can see the audio part is not send to the ata. I understand that is another "call" between the ATA and FS, but should FS send an invite to the ATA containing a audio part as well? I'm not sure that is a bug or Is the normal behavior? should i open a jira for it? Searching for previous issues i found this, but there was no reply ... http://freeswitch-users.2379917.n2.nabble.com/Doubts-about-T38-passthru-td7… Thanks for the help, António

1 0

New codec integration and concurrent calls limit questions
by Fernando Hernandez 17 May '13

17 May '13

Hello everyone, we are working on a videoconferece platform with freeSWITCH integrated. Now we have a the possibility to try a new codec, proprietary, and we have never gone too deep with freeSWITCH. I have tried to take a look at the wiki but I didn´t find the answer to the next question, Is it possible to integrate that new codec into freeSWITCH? according to the codec provider it is a G.279 implementation, with several improvements. Any licence issue? difficulty to achieve the integration? On the other hand, we also have another issue. I have checked freeSWITCH wiki trying to find any possible limit for concurrent calls, and according to it, it doesn't seem to be a limit of SIP but purely the RTP. But with the open source video conference platform we are working now this is limited to 25, and according to the developers it is due to the audio. Do you have any idea about this issue? We have run some test and we go over 25 people in the meeting freeSWITCH uses 125% of CPU (ref. 8 cores machine and each core represent 100% capacity => 800%), is this a normal behavior? Thank you very much, Fernando Hernandez

5 7

Freeswitch & Opsview Monitoring
by Alex Ynema 17 May '13

17 May '13

I thought I'd post this for people reference. I've configured opsview (Nagios) to monitor my current sessions & graph it so I can match performance & usage. On the Freeswitch server install the opsview-agent & set service to auto start Create check_freeswitch_sessions.sh in /usr/local/nagios/libexec & chkmod +x the script -------------------------------------------------------- #!/bin/bash set fsStatus=" " /usr/local/freeswitch/bin/fs_cli -x "show calls count" > /tmp/fsStatus if [ $? -ne 0 ]; then echo "Critical: Freeswitch not responding!" rm /tmp/fsStatus exit -1 fi fsCalls=`grep "total."</tmp/fsStatus` #Strip unwanted chars from the channels responses to get pure ints. fsCalls=${fsCalls/ total./} # Build a message up message="Active Sessions: $fsCalls" echo $message rm /tmp/fsStatus #exit $fsCalls exit 0 -------------------------------------------------------- The add the following to /usr/local/nagios/nrpe.cfg -------------------------------------------------------- command[check_freeswitch_sessions]=/usr/local/nagios/libexec/check_freeswitch_Sessions.sh -------------------------------------------------------- On Opsview server use check_nrpe service check to remotely run the check_freeswitch_sessions service check. Then on your opsview server edit /usr/local/nagios/etc/map and add the following -------------------------------------------------------- # Service type: Freeswitch Sessions # ouput:Active Sessions: 0 /output:Active.*?(\d+)/ and push @s, [ "session", [ "sessions", GAUGE, $1 ] ]; -------------------------------------------------------- *Alex Ynema** *| IT Consultant alex(a)opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 <http://www.ynema.net/> <http://www.ynema.net/>IT Consultant for Open Systems Support www.opensystems.net.au

2 2

remove
by Andre 16 May '13

16 May '13

Hi, can you remove me from the distro? It's just too many emails. I'll add myself back later when I'm ready to use freeswitch. Thank you, Andre

2 1