December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

from FS to * CALL_REJECTED
by Zhuravlov Sergey 16 Dec '11

16 Dec '11

Good day! Can not connect sobstvennye FS and asterisk. I do, as described here http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk !!!With ACLs!!! call with asterisk to FS happened, but with FS to asterisk call fail: call is made so EXECUTE sofia/internal/272@my.domain.com:5060 bridge(sofia/external/711(a)xx.xxx.xx.xx) http://pastebin.freeswitch.org/18014 and packets are not with FS in the direction of asterisk is not going [root@fs0 freeswitch]# tcpdump -nS -i venet0:0 dst xx.xxx.xx.xx where xx.xxx.xx.xx -- asterisk what could be the reason? -- Zhuravlov Sergey GTALK/JABBER:4orbit@gmail.com

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Howto Set default language for an application ?
by fsrichard＠ghz.fr 15 Dec '11

15 Dec '11

Hello, I'm trying to set up my first FreeSWITCH server but I don't know where to set the default language corresponding to the following line in conf/dialplan/default.xml : > <extension name="Local_Extension"> > ... > <action application="bridge" data="loopback/app=voicemail:default > ${domain_name} ${dialed_extension}"/> > ... > </extension> I have tried this : > <extension name="Local_Extension"> > ... > *<action application="set" data="default_language=fr"/>* > ... > <action application="bridge" data="loopback/app=voicemail:default > ${domain_name} ${dialed_extension}"/> > ... > </extension> From what I have understood it is normal that it doesn't work as the bridge to a loopback would use it's own settings, but I don't have a clue where to set it of app=voicemail Is there a way to set the default language per SIP connection ? I have got an english and a french SIP line and would like to make calls going to the english line have english instructions for leaving a message on the voicemail and the calls to the french line to have french instructions for leaving a message on the voicemail. Thank you Richard

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Logging/monitor call-quality indicators
by Fraser Redmond 15 Dec '11

15 Dec '11

How do you monitor the health of your servers/calls? I come from a web-development background where you can track things like how long it takes to generate a page, run a sql query, how many php or sql errors you get, and so on, then go back through the logs to solve problems that hadn't been reported - it'd be nice to do similar for voip, especially as there are so many more variables beyond my control. I was wondering if there are any ways to go about logging things related to call-quality like audio-latency, jitter, packet-loss, sip-errors, etc. I know about sip-traces, and the FS log, but that's too granular - I first would need to know a certain type of problem is occurring before I can investigate individual calls related to it. The only thing I can find in the wiki is PBXMate - but there isn't any mention of it in the maililng-list. Is anyone using it? Any feedback? Cheers, Fraser

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Reinstall FreeSWITCH after new kernel?
by Henrik Aagaard Sørensen 15 Dec '11

15 Dec '11

I've read a lot about the kernel timer and how it should be on 1000hz. My clean install of Ubuntu has 100hz. So I guess I have to recompile the kernel to 1000hz? If so, does I have to reinstall FreeSWITCH after the new kernel compile? And does anyone knows of any tutorial about recompiling a Ubuntu 10.04 LTS kernel to 1000hz?

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user directory variable
by Miha Zoubek 15 Dec '11

15 Dec '11

Hi, I would like to disable radius for incoming calls. This two variables disable_radius_start=true and disable_radius_stop=true are for disabling radius. I have put this in action application form in dialplan, I guess this is too late, as start packet is also send. I need to put this in user directory so that radius will be stoped in right time (and some logic that this would be only for incoming calls). Problem is that I have put this as variable (like this <variable name="disable_radius_stop" value="true" /> <variable name="disable_radius_start" value="true" /> ) in user directory and the radius packet is still send. So why this variable is not set as packets are still send? Thank for your help! Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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Re: [Freeswitch-users] FreeSWITCH doesn't get SIP registers.
by Peter Olsson 14 Dec '11

14 Dec '11

CentOS enables it by default I think. Try 'service firewall stop' if it helps. Wireshark might detect packets even if they are filtered later on. /Peter ----- Reply message ----- Från: "Henrik Aagaard Sørensen" <henrikaagaardsorensen(a)gmail.com> Datum: ons, dec 14, 2011 18:56 Rubrik: [Freeswitch-users] Ang.: FreeSWITCH doesn't get SIP registers. Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> But I'm able to see incoming SIP on port 5060 with wireshark. And I haven't installed any firewall? On 14/12/2011, at 12.47, Peter Olsson <peter.olsson(a)visionutveckling.se> wrote: > Check the firewall settings on the box, or deactivate the firewall, that's the most probable cause. > > /Peter > > ----- Reply message ----- > Från: "Henrik Aagaard Sørensen" <henrikaagaardsorensen(a)gmail.com> > Datum: ons, dec 14, 2011 18:26 > Rubrik: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. > Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> > > I've installed FreeSWITCH on a clean CentOS 6 server. No errors etc. > It startup fine, again without any errors. > > But when I try to connect a SIP client to the sample extension 1000 (or any other) I get a timeout. > > When I run netstat -tnulp | grep "freeswitch" I get: > tcp 0 0 83.221.133.18:5080<http://83.221.133.18:5080> 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 83.221.133.18:5060<http://83.221.133.18:5060> 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 127.0.0.1:8021<http://127.0.0.1:8021> 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 2001:1448:246:9f47:216:5060 :::* LISTEN 1167/freeswitch > udp 0 0 83.221.133.18:5080<http://83.221.133.18:5080> 0.0.0.0:* 1167/freeswitch > udp 0 0 83.221.133.18:5060<http://83.221.133.18:5060> 0.0.0.0:* 1167/freeswitch > udp 0 0 2001:1448:246:9f47:5060 :::* 1167/freeswitch > > In fs_cli, when I run sofia status profile external, I get: > Name external > Domain Name N/A > Auto-NAT false > DBName sofia_reg_external > Pres Hosts > Dialplan XML > Context public > Challenge Realm auto_to > RTP-IP 83.221.133.18 > SIP-IP 83.221.133.18 > URL sip:mod_sofia@83.221.133.18:5080<http://sip:mod_sofia@83.221.133.18:5080> > BIND-URL sip:mod_sofia@83.221.133.18:5080<http://sip:mod_sofia@83.221.133.18:5080> > HOLD-MUSIC local_stream://moh > OUTBOUND-PROXY N/A > CODECS IN G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM > CODECS OUT PCMU,PCMA,GSM > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG false > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > CALLS-IN 0 > FAILED-CALLS-IN 0 > CALLS-OUT 0 > FAILED-CALLS-OUT 0 > > When I try to connect and run tshark -f "port 5060" I get: > 0.000000 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 > 3.997431 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 > ... > > I've setup FreeSWITCH on other CentOS 6 distro's and Ubuntu, without any problem. This one I cannot figure out though. > > Can anyone help? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ee8e13532768713110985!

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"302 Moved Temporarily" invokes ${hold_music} instead of ${transfer_ringback}
by Christian Benke 14 Dec '11

14 Dec '11

Hi! I've been tinkering with ringback/transfer_ringback and moh lately and i'm a bit confused about the behaviour in certain cases: I've configured ringback, transfer_ringback and hold_music with three different local-streams(ringback and transfer_ringback are set in the bridging extension, moh is set as a sip_profile-parameter). What i expect to get is hearing "ringback" before the call is answered, "transfer_ringback" when i transfer the answered call from b-leg-endpoint and "moh" wenn the call is set "on hold". Unfortunately this only works when the transfer is initiated via the transfer-command in FreeSWITCH(Default features.xml dx). If the answered calls is transfered by the b-leg-endpoint with "302 Moved Temporarily", FreeSWITCH plays "hold_music". Is there a way to change this behaviour? Best regards Christian

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Re: [Freeswitch-users] FreeSWITCH doesn't get SIP registers.
by Peter Olsson 14 Dec '11

14 Dec '11

Try service iptables stop. My mistake... /Peter ----- Reply message ----- Från: "Henrik Aagaard Sørensen" <henrikaagaardsorensen(a)gmail.com> Datum: ons, dec 14, 2011 19:28 Rubrik: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> I just get this: service firewall stop firewall: unrecognized service On Wed, Dec 14, 2011 at 7:11 PM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: CentOS enables it by default I think. Try 'service firewall stop' if it helps. Wireshark might detect packets even if they are filtered later on. /Peter ----- Reply message ----- Från: "Henrik Aagaard Sørensen" <henrikaagaardsorensen(a)gmail.com<mailto:henrikaagaardsorensen@gmail.com>> Datum: ons, dec 14, 2011 18:56 Rubrik: [Freeswitch-users] Ang.: FreeSWITCH doesn't get SIP registers. Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> But I'm able to see incoming SIP on port 5060 with wireshark. And I haven't installed any firewall? On 14/12/2011, at 12.47, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: > Check the firewall settings on the box, or deactivate the firewall, that's the most probable cause. > > /Peter > > ----- Reply message ----- > Från: "Henrik Aagaard Sørensen" <henrikaagaardsorensen(a)gmail.com<mailto:henrikaagaardsorensen@gmail.com>> > Datum: ons, dec 14, 2011 18:26 > Rubrik: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. > Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> > > I've installed FreeSWITCH on a clean CentOS 6 server. No errors etc. > It startup fine, again without any errors. > > But when I try to connect a SIP client to the sample extension 1000 (or any other) I get a timeout. > > When I run netstat -tnulp | grep "freeswitch" I get: > tcp 0 0 83.221.133.18:5080<http://83.221.133.18:5080><http://83.221.133.18:5080> 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 83.221.133.18:5060<http://83.221.133.18:5060><http://83.221.133.18:5060> 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 127.0.0.1:8021<http://127.0.0.1:8021><http://127.0.0.1:8021> 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 2001:1448:246:9f47:216:5060 :::* LISTEN 1167/freeswitch > udp 0 0 83.221.133.18:5080<http://83.221.133.18:5080><http://83.221.133.18:5080> 0.0.0.0:* 1167/freeswitch > udp 0 0 83.221.133.18:5060<http://83.221.133.18:5060><http://83.221.133.18:5060> 0.0.0.0:* 1167/freeswitch > udp 0 0 2001:1448:246:9f47:5060 :::* 1167/freeswitch > > In fs_cli, when I run sofia status profile external, I get: > Name external > Domain Name N/A > Auto-NAT false > DBName sofia_reg_external > Pres Hosts > Dialplan XML > Context public > Challenge Realm auto_to > RTP-IP 83.221.133.18<tel:83.221.133.18> > SIP-IP 83.221.133.18<tel:83.221.133.18> > URL sip:mod_sofia@83.221.133.18:5080<http://sip:mod_sofia@83.221.133.18:5080><http://sip:mod_sofia@83.221.133.18:5080> > BIND-URL sip:mod_sofia@83.221.133.18:5080<http://sip:mod_sofia@83.221.133.18:5080><http://sip:mod_sofia@83.221.133.18:5080> > HOLD-MUSIC local_stream://moh > OUTBOUND-PROXY N/A > CODECS IN G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM > CODECS OUT PCMU,PCMA,GSM > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG false > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > CALLS-IN 0 > FAILED-CALLS-IN 0 > CALLS-OUT 0 > FAILED-CALLS-OUT 0 > > When I try to connect and run tshark -f "port 5060" I get: > 0.000000 96.224.14.164 -> 83.221.133.18<tel:83.221.133.18> SIP Request: REGISTER sip:83.221.133.18<tel:83.221.133.18> > 3.997431 96.224.14.164 -> 83.221.133.18<tel:83.221.133.18> SIP Request: REGISTER sip:83.221.133.18<tel:83.221.133.18> > ... > > I've setup FreeSWITCH on other CentOS 6 distro's and Ubuntu, without any problem. This one I cannot figure out though. > > Can anyone help? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ee8e8b132761321042113!

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How to set voicemail default language (leaving message not listening to message)?
by Richard 14 Dec '11

14 Dec '11

Hello, I'm trying to get the voice that talks on the voicemail before leaving a message (dial # or stop taking to end your message) to french language but cannot find out where to set this ! I havent been able to find any documentation about where to set application languages when called in a bridge loopback like in the following line : <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> I added the default language line to the extension that contains this line (default.xml : Local_Extension) but it does not affect the loopback. Full extension : <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9]|1100)$"> *<action application="set" data="default_language=fr"/>* <action application="export" data="dialed_extension=$1"/>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/>  <action application="set" data="hangup_after_bridge=true"/>  <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>  <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> I have got my French language working when I call the 4000 extension : <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^4000$|^\*98$"> *<action application="set" data="default_language=fr"/>* <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name}"/> </condition> </extension> If I change the path of the audio files in conf/lang/en/en.xml everything works in French. However I want to have one extension with English language and another with French langauge so I need to be able to set the default language to fr when bridge/loopbacking to app=voicemail. It's the application="bridge" data="loopback/app= that baffles me ! I presume the loopback passes the call onto the voicemail application but I haven't found where to set the default language for this application when using a bridge and loopback ! Thanks in advance, Richard

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FreeSWITCH Logger
by arnuld＠Phonologies.com 14 Dec '11

14 Dec '11

Currently FreeSWITCH logs to files on disk and you can customize the behavior a bit using logfile.conf.xml. I am working on a project where I need to direct FreeSWITCH to write logs to a socket (like a TCP client). I have written a TCP Server listening for logs on a particular socket. I thought I will change the source code of freeswitch/src/switch_log.c and will add TCP client code to it, then recomplile it. Is there any better way ?

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