August 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

PD: SIP -> XMPP calls
by Muszalski Michał - Korpo TP 12 Aug '14

12 Aug '14

> Hello, > > We are trying to make some kind of a gateway between SIP and XMPP domain. > We have an environment with FreeSWITCH and OpenFire (FreeSWITCH is > registered as a component in OF). Calls from XMPP to SIP are working fine. > The problem is with calls from SIP to XMPP. The caller (SIP) has a ringing > tone, a the callee (XMPP) has a 'connecting...' message after answer the > call. > Do you have any experience in making calls between SIP and XMPP users? Is > it possible to establish the video calls also? > > Best Regards, > > Michał Muszalski > Senior Specialist > > Orange Labs Poland > Obrzeżna 7, 02-691 Warsaw >

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OpenSSL security advisory
by Michael Jerris 11 Aug '14

11 Aug '14

Please note the release of a new OpenSSL earlier today. We recommend you upgrade to the latest release immediately as a number of the addressed issues will effect freeswitch https://www.openssl.org/news/secadv_20140806.txt

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mod_conf : play tmp/uuid.wav!userleft on member exit.
by Stephen Dame 11 Aug '14

11 Aug '14

I've successfully been able to modify dialplan to prompt caller for name, record it and announce in conference per wiki. solution works great. Adding a few lines of code makes sure they can only enter existing conferences. Now that I have their name in temp wav file, trying to decide best approach to play name!" is leaving conference" into conference when they leave. Before heading down a path. Wanted to seek some advice on some possible approaches. 1) Thought I may be able to set exit sound in conf profile with UUID but not sure variables are expanded at run time. 2) The other approach is to monitor ESL conference events and play back the UUID file when its detected that they have left (separate process) 3) Modify source of mod_conf to hardcode playing of /tmp/uuid when member leaves only if the file exists. Is there an easier way I'm missing? Regards, Stephen

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Presence + Transfers
by Chris Tunbridge 11 Aug '14

11 Aug '14

Currently i have Aastra and Cisco having this behavior, working on getting a polycom lab setup. FreeSWITCH Version 1.4.7+git~20140724T230506Z~9d1c0f7f3d~64bit (git 9d1c0f7 2014-07-24 23:05:06Z 64bit) I have my extensions 101, 102 and 103 So if i call in, and answer on 101, 102 and 103 will successfully show that 101 is on the phone. If i then transfer the call from 101 to 102, 102 and 103 will show 101 get off the phone, however 101, 103 never show that 102 is on the phone. further transferring the call from 102 to 101 or 103 also never shows that they are on the phone *conf/sip_profile/internal.xml:* http://pastebin.freeswitch.com/23020 *conf/autoload_configs/presence_map.conf.xml:* http://pastebin.freeswitch.com/23021 Dial String is as follows: <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(*/${dialed_user}@ ${dialed_domain}):transfer_fallback_extension=${dialed_user}}"/> This behavior is similar on Jitsi, but because i set it to do peer mode, it checks every 30 seconds so after 30 seconds it shows that the person (in this case 102) is on the phone.

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Dingaling In Component Mode "How To Call JingleTo SIP"
by Ali Jawad 11 Aug '14

11 Aug '14

Dear All This might be a duplicate post, I am sorry if it is. Normally I get direct feedback from the list. But this time I did not get any reply. So I was afraid it might not have been delivered. First I would like to state that dingaling is working in client mode with gtalk perfectly. I can call from gtalk to my sip extension "1000". I have setup component mode now I am using FS with Openfire, open does show the following: 1 fs.jabber2.mydomain.net LibDingaLing gateway However stats on the jabber server are Session Statistics: Packets Received/Sent: 1/1 So the component is REGISTERED, so far so good my problem now is how to initiate a call from XMPP to a SIP extension. Using client mode I would do the following: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@mydomain to the jabber client "Jabbin" does not popup an "add contact" on XTEN. Anything I missed here ? Thanks for any hints or suggestions The Freeswitch server logs are below: 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" to="fs" version="1.0"></stream:stream> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" from="fs.jabber2.MyDomain" id="69fcaa7b"></stream:stream> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <handshake>72dac833190326bb82dddedae558b2b794df5e1b</handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <handshake></handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <iq type="get" id="970-504" from="component.jabber2.MyDomain" to="fs.jabber2.MyDomain"> <query xmlns="http://jabber.org/protocol/disco#info"></query> </iq> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <iq from="fs" to="component.jabber2.MyDomain" id="970-504" type="result"> <query xmlns="http://jabber.org/protocol/disco#info"> <identity category="gateway" name="LibDingaLing"></identity> <feature var="http://jabber.org/protocol/disco"></feature> <feature var="http://jabber.org/protocol/disco#info"></feature> <feature var="jabber:iq:version"></feature> <feature var="vcard-temp"></feature> <feature var="http://www.google.com/xmpp/protocol/voice/v1"></feature> <feature var="jabber:iq:last"></feature> </query> </iq>

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Ubiquiti UniFi-VoIP and FS
by Kurt Albershardt 11 Aug '14

11 Aug '14

According to http://www.nojitter.com/post/240168789/ubiquiti-networks-enters-the-voip-ma…, Ubiquiti is using FS in its USG http://www.ubnt.com/unifi-switching-routing/usg/ Anyone have any details? thanks~

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Re: [Freeswitch-users] Originating calls
by Michael Jerris 11 Aug '14

11 Aug '14

https://wiki.freeswitch.org/wiki/Loopback On Aug 2, 2014, at 11:19 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > Hello Freeswitch users. > > Currently when I originate a call in my IVR application, I execute the API originate command like this: > api originate {variable1=data,variable2=data}sofia/gateway/outbound/<number> &park() xml default > So yes I have a gateway called outbound and this originates a call by dialling the specified number on that gateway and parking it. I get the UUID and can then do stuff with it. > > But what I really want to do is originate a call by putting the specified phone number through the dialplan so Freeswitch can work out which endpoint to use. I am sure this must be possible but I can't figure out how to do it. > > Just to summarise, I want to originate a call to a number which might be a local extension or it might be a number that is reached through a gateway. Why can't I do something like: > api originate {variable1=data,variable2=data}<number> &park() xml default > I want Freeswitch to work out which Sofia profile and endpoint to use. I want Freeswitch to park the call in the same way as above and give me the UUID. Is this possible? > > Can anyone help with this please? >

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Another ClueCon Successfully Completed!
by Ken Rice 11 Aug '14

11 Aug '14

Hey ClueCon¹ers! I want to thank all of you for making ClueCon 2014 a HUGE Success! There are too many people to name in a single email that went to helping make ClueCon a success, but I wanted to call out a few people. John Boteler AKA Boteman Thanks for helping me with the A/V! William King aka quentusrex_ Thanks for helping copy, backup, and keep track of all the media we recorded at ClueCon Anthony Minessale, Brian West and Mike Jerris Thanks for developing ClueCon over the last 10 years! With you guys it just wouldn¹t happen! Adam Kalsey Thanks for MC¹ing again this year. Kathleen King and Kristin King Thanks for help with on the fly graphics, registration, and all around help! Sharon White Thanks for helping us all stay on schedule! Lets not forget our wonderful Sponsors. These guys made it all possible! RangeNetworks, Sangoma, CorvisaCloud, FlowRoute, Simwood, Truphone, Fairwaves, OnSip, 2600HZ, Ziron, Marchex, Yealink, Ooma, OrecX, VoiceTel, and ThinkGeek! (visit http://cluecon.com/sponsors for links) Now the real fun begins! We get to start planning ClueCon 2015! We already know the dates will be Aug 3-7 2015 so block that week out! Again thanks to everyone! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Cisco 7906 + freeswitch (not registering)
by Nick Zaitsev 11 Aug '14

11 Aug '14

Good day to you. I have cisco7906,i ve upgrade it to use SIP,but,when it sends only REGISTER,and after it recieves OK,it does not send anything Here is the config for cisco 7906 <device> <fullConfig>true</fullConfig> <deviceProtocol>SIP</deviceProtocol> <devicePool> <dateTimeSetting> <dateTemplate>D.M.Y</dateTemplate> <timeZone>Ekaterinburg Standard Time</timeZone> <ntps> <ntp> <name>$NTP</name> <ntpMode>Unicast</ntpMode> </ntp> </ntps> </dateTimeSetting> <callManagerGroup> <tftpDefault>true</tftpDefault> <members> <member priority="0"> <callManager> <name>$Freeswitch</name> <description>CallManager 5.0</description> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> </ports> <processNodeName>$Freeswitch</processNodeName> </callManager> </member> </members> </callManagerGroup> </devicePool> <commonProfile> <phonePassword></phonePassword> <backgroundImageAccess>true</backgroundImageAccess> <callLogBlfEnabled>0</callLogBlfEnabled> </commonProfile> <loadInformation>SIP42.8-5-4S</loadInformation> <loadInformation434 model="Cisco 7942">SIP42.8-5-4S</loadInformation434> <vendorConfig> <disableSpeaker>false</disableSpeaker> <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset> <pcPort>0</pcPort> <settingsAccess>1</settingsAccess> <garp>0</garp> <voiceVlanAccess>0</voiceVlanAccess> <videoCapability>0</videoCapability> <autoSelectLineEnable>0</autoSelectLineEnable> <daysDisplayNotActive>1,7</daysDisplayNotActive> <displayOnTime>10:30</displayOnTime> <displayOnDuration>06:05</displayOnDuration> <displayIdleTimeout>00:05</displayIdleTimeout> <webAccess>1</webAccess> <spanToPCPort>1</spanToPCPort> <loggingDisplay>1</loggingDisplay> <loadServer></loadServer> </vendorConfig> <deviceSecurityMode>1</deviceSecurityMode> <idleTimeout>0</idleTimeout> <directoryURL></directoryURL> <servicesURL>$SERVICESURL</servicesURL> <idleURL></idleURL> <messagesURL></messagesURL> <proxyServerURL></proxyServerURL> <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig> <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> <dscpForCm2Dvce>96</dscpForCm2Dvce> <transportLayerProtocol>2</transportLayerProtocol> <capfAuthMode>0</capfAuthMode> <capfList> <capf> <phonePort>3804</phonePort> </capf> </capfList> <certHash></certHash> <encrConfig>false</encrConfig> <sipProfile> <sipProxies> <backupProxy>$Freeswitch</backupProxy> <backupProxyPort>5060</backupProxyPort> <emergencyProxy>$Freeswitch</emergencyProxy> <emergencyProxyPort>5060</emergencyProxyPort> <outboundProxy>$Freeswitch</outboundProxy> <outboundProxyPort>5060</outboundProxyPort> <registerWithProxy>true</registerWithProxy> </sipProxies> <sipCallFeatures> <cnfJoinEnabled>true</cnfJoinEnabled> <callForwardURI>x--serviceuri-cfwdall</callForwardURI> <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI> <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI> <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI> <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI> <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI> <rfc2543Hold>false</rfc2543Hold> <callHoldRingback>2</callHoldRingback> <localCfwdEnable>true</localCfwdEnable> <semiAttendedTransfer>true</semiAttendedTransfer> <anonymousCallBlock>2</anonymousCallBlock> <callerIdBlocking>2</callerIdBlocking> <dndControl>0</dndControl> <remoteCcEnable>true</remoteCcEnable> </sipCallFeatures> <sipStack> <sipInviteRetx>6</sipInviteRetx> <sipRetx>10</sipRetx> <timerInviteExpires>180</timerInviteExpires> <timerRegisterExpires>3600</timerRegisterExpires> <timerRegisterDelta>5</timerRegisterDelta> <timerKeepAliveExpires>120</timerKeepAliveExpires> <timerSubscribeExpires>120</timerSubscribeExpires> <timerSubscribeDelta>5</timerSubscribeDelta> <timerT1>500</timerT1> <timerT2>4000</timerT2> <maxRedirects>70</maxRedirects> <remotePartyID>false</remotePartyID> <userInfo>None</userInfo> </sipStack> <autoAnswerTimer>1</autoAnswerTimer> <autoAnswerAltBehavior>false</autoAnswerAltBehavior> <autoAnswerOverride>true</autoAnswerOverride> <transferOnhookEnabled>false</transferOnhookEnabled> <enableVad>false</enableVad> <preferredCodec>g711alaw</preferredCodec> <dtmfAvtPayload>101</dtmfAvtPayload> <dtmfDbLevel>3</dtmfDbLevel> <dtmfOutofBand>avt</dtmfOutofBand> <alwaysUsePrimeLine>false</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <stutterMsgWaiting>1</stutterMsgWaiting> <callStats>true</callStats> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig> <startMediaPort>10100</startMediaPort> <stopMediaPort>10300</stopMediaPort> <voipControlPort>5060</voipControlPort> <dscpForAudio>184</dscpForAudio> <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> <dialTemplate>dialplan.xml</dialTemplate> <phoneLabel>Cisco</phoneLabel> <natReceivedProcessing>false</natReceivedProcessing> <natEnabled>false</natEnabled> <natAddress></natAddress> <sipLines> <line button="1"> <featureID>9</featureID> <featureLabel>$ACCOUNT</featureLabel> <proxy>$Freeswitch</proxy> <port>5060</port> <name>$ACCOUNT</name> <displayName>$ACCOUNT</displayName> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>3</callWaiting> <authName>$ACCOUNT</authName> <authPassword>$ACCOUNT_PASS</authPassword> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>3</messageWaitingLampPolicy> <messagesNumber></messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <contact>$ACCOUNT</contact> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>false</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> </line> <line button="2"> <featureID></featureID> <featureLabel></featureLabel> <speedDialNumber></speedDialNumber> </line> </sipLines> </sipProfile> </device> Here is tcpdump 10:44:13.076447 IP (tos 0x0, ttl 63, id 22, offset 0, flags [none], proto UDP (17), length 601) 10.4.33.18.49156 > fsw [no cksum] SIP, length: 573 REGISTER sip:10.160.1.17 SIP/2.0 10:44:13.078747 IP (tos 0x0, ttl 64, id 22120, offset 0, flags [none], proto UDP (17), length 666) fsw.sip > 10.4.33.18.sip: [udp sum ok] SIP, length: 638 SIP/2.0 401 Unauthorized 10:44:13.086086 IP (tos 0x0, ttl 63, id 23, offset 0, flags [none], proto UDP (17), length 824) 10.4.33.18.49156 > fsw.sip: [no cksum] SIP, length: 796 REGISTER sip:10.160.1.17 SIP/2.0 10:44:13.093733 IP (tos 0x0, ttl 64, id 22121, offset 0, flags [none], proto UDP (17), length 638) fsw.sip > 10.4.33.18.sip: [udp sum ok] SIP, length: 610 SIP/2.0 200 OK here is the reg status from fs sofia Call-ID: 00270dbd-978d0002-fc125432-60241d47(a)x.x.x.x User: 1005(a)x.x.x.x Contact: "" <sip:1005@10.160.1.59:5060;transport=udp> Agent: Cisco-CP7906G/8.5.3 Status: Registered(UDP)(unknown) EXP(2014-08-11 14:15:01) EXPSECS(3441) Host: fsw1 Port: 49154 Auth-User: 1005 Call-ID: m7uDzd2ojPUAn-5SmfJWhvJ43l3I6-9e User: 1002(a)x.x.x.x Contact: "1002" <sip:1002@10.132.144.133:53706;ob> Agent: CSipSimple_GT-I9000-10/r2330 Status: Registered(UDP)(unknown) EXP(2014-08-11 13:33:38) EXPSECS(958) Host: fsw1 Port: 5060 Auth-User: 1002 Total items returned: 2 and the lots of NOTIFY from freeswitch, Could anybody help with this issue please? Best regards -- Nick Zaitsev

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keeping leg b up and reconnect to it
by ik 11 Aug '14

11 Aug '14

Hello, I have a SIP user that is working under cellular 3g data. Sometimes it losing it's IP and have it assigned new IP, other times, roaming disconnect the data and it takes up to 45 seconds to reconnect (and again, gaining new IP). While the client is idle, there is no issue, however when the client is connected to a call, freeswitch (and any other sip servers for that matter) does not know about it (and that's okay). But when the client return to be available, and try to reconnect to the call, the call itself disconnect, and making hangup on both sides. I have the following dialplan: <?xml version="1.0" encoding="utf-8"?> <include> <context name="testing"> <extension name="bridge loopback"> <condition field="destination_number" expression="^(300\d)$"> <action application="set" data="hangup_after_bridge=false" /> <action application="bridge" data="{hangup_after_bridge=false}user/$1" /> </condition> </extension> </context> </include> Is there a way to keep the other leg (for example leg b) up, and attempting to reconnect leg-a to it ? Thanks, Ido

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