- FreeSWITCH-users - FreeSWITCH Mailing Lists

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

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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?

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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

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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

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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

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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

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Sdp manipulation help
by Daniel Ivanov 16 May '13

16 May '13

I am having trouble putting srtp to optional mode. UAs send sdp with rtp/avp and a:crypto lines which im permitting. But i want to decide whether the call uses srtp or zrtp. The UAs send also zrtp-hash. I export sip_secure_media to false but am still seeing the a:crypto lines being sent to the b-leg. I cut them off in my dialplan with set switch_r_sdp, but that breaks the whole negotiation, because the sdp isn't properly patched afterwards for nat( i also set a bunch of other vars afterwards). Is there a polite way of telling fs to remove the a:crypto lines and break the srtp and thus activating zrtp. Running 1.2.8git with inbound_zrtp_passthru to true.

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mod_cdr_csv is not add cdr data in csv file
by Shahzad Bhatti 16 May '13

16 May '13

Hi everyone, i am using *FreeSWITCH Version 1.5.1b+git~20130515T182820Z~37d109b107 (git 37d109b 2013-05-15 18:28:20Z)* on CentOS and for the mod_cdr_csv my conf. is as: <configuration name="cdr_csv.conf" description="CDR CSV Format"> <settings>   <param name="default-template" value="example"/>  <param name="debug" value="false"/> <param name="rotate-on-hup" value="false"/>  <param name="legs" value="a"/>  <param name="master-file-only" value="true"/> </settings> <templates> <template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}");</template> <template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template> <template name="snom">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${call_clientcode}","${sip_rtp_rxstat}","${sip_rtp_txstat}","${sofia_record_file}"</template> <template name="linksys">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${sip_p_rtp_stat}"</template> <template name="asterisk">"${accountcode}","${caller_id_number}","${destination_number}","${context}","${caller_id}","${channel_name}","${bridge_channel}","${last_app}","${last_arg}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${amaflags}","${uuid}","${userfield}"</template> <template name="opencdrrate">"${uuid}","${signal_bond}","${direction}","${ani}","${destination_number}","${answer_stamp}","${end_stamp}","${billsec}","${accountcode}","${userfield}","${network_addr}","${regex('${original_caller_id_name}'|^.)}","${sip_gateway_name}"</template> </templates> </configuration> but the issue is that when i use *cdr_csv rotate* on fs_cli then sometime it not works and also the cdr data is not added in the csv file sometimes any suggestions is highly appreciated Regards Shahzad Bhatti

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Re: [Freeswitch-users] IP change detected -> no internal profile
by info 16 May '13

16 May '13

On Tue, May 14, 2013 at 10:31 AM, Siri MM < sirimmfs(a)gmail.com > wrote: Hello All, I am facing an issue with freeswitch installed on my device. For some reason, IP address on my device sometimes seems to be 'fluctuating' between the actual Static IP, and 127.0.0.1. Freeswitch is able to detect this change everytime, and reconfigure accordingly. However, during one such occassion, it hasn't brought up the internal profile, and isn't binding to port 5060. No other program uses this port on my device. Working log: http://pastebin.freeswitch.org/20912 Not-working log: http://pastebin.freeswitch.org/20915 Would appreciate any inputs Thanks! Hi Siri, I am a bit mystified as to how things are supposed to work if (how?) your IP get's switched between static and 127.0.0.1 (again, how?) This feels just wrong on so many levels :-) 127.0.0.1 normally is bound to the loopback interface, which, on linux, is called lo your other IP should be bound to an eth0 interface, or an internal ip and and the static IP on a router with some sort of NAT or port redirection. AFAIK linux should not allow you bind 127.0.0.1 to eth0 if it is allready bound to lo, so my best guess would be that you loose ethX and you are binding FS to all available interfaces... Alas, I don't know how to answer, because even if there should be a problem bringing up the internal profile on such a "switch", my guess is that that's probably *not* the problem you should be looking to fix for right now. Especially since, if somehow the IP really get's switched to 127.0.0.1, you will be talking only to yourself and I do not see how this should create a working setup (unless you use only an internal sip/(pri|bri) card or some USB channelbanks/gprs sticks, etc. and you only will be doing robocalls or automated IVR's ... Well, sorry mate that I can't help you more, but this really is a strange szenario, best regards, Ray

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IP change detected -> no internal profile
by Siri MM 16 May '13

16 May '13

Hello All, I am facing an issue with freeswitch installed on my device. For some reason, IP address on my device sometimes seems to be 'fluctuating' between the actual Static IP, and 127.0.0.1. Freeswitch is able to detect this change everytime, and reconfigure accordingly. However, during one such occassion, it hasn't brought up the internal profile, and isn't binding to port 5060. No other program uses this port on my device. Working log: http://pastebin.freeswitch.org/20912 Not-working log: http://pastebin.freeswitch.org/20915 Would appreciate any inputs Thanks!

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