July 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

possible bug with mod_callcenter
by Nathan Neulinger 29 Jul '14

29 Jul '14

I had an agent on a call, then set it's status to "Available (On Demand)". Called into queue from another extension. When the first call finished, this user was immediately handed another call, but only ONE call, almost like it got handed the call before the automatic transition to Idle had a chance to take effect. This is with (relatively recent) master. Anyone else seen this? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Newbie, Video question - with 2 users, can they show each other?
by Mark Berkeland 29 Jul '14

29 Jul '14

I have freeswitch working great with several SIP video clients, but my question is, if I only have 2 video users, I would like them to each see the other, NOT have them both follow the speaker (voice switching). As it currently stands with my system, if one person is talking, he only sees himself (and the far end sees him as well). I would like the speaker to see the other person, as it is with point to point video solutions. Of course, when a third user joins, the voice switching is fine. But is there a way to have it NOT do that when there are only two participants? I have spent some time searching the lists and the freeswitch docs, but have not stumbled upon the right search terms to help me find an answer. Thanks! -Mark Berkeland

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Displaying user friendly caller name
by Malay Thakershi 28 Jul '14

28 Jul '14

Hello, I use originate to make outbound calls. originate {origination_caller_id_name='CH calling',origination_caller_ id_number=1XXXXXXXXXX,origination_uuid=XXXXXXXX,absolute_codec_string=PCMU, ignore_early_media=true,originate_timeout=45,}sofia/gateway/chsbc01/1XXXXXXXXXX 1056 XML default Can I display "CH calling" as caller name when people receive our calls? It is not displaying that. It only display the number. Thanks. In public.xml dial plan: <extension name="provider-inbound"> <condition field="destination_number" expression="^XXXXXXXXXX|XXXXXXXXXX$"> <action application="pre_answer" /> <action application="sleep" data="1000" /> <action application="transfer" data="1056 XML default" /> </condition> </extension> In default.xml dial plan: <extension name="CHPhone"> <condition field="destination_number" expression="^1056$">  <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 1980 Factory Records, Inc."/> <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/> <action application="set" data="RECORD_ARTIST=CH"/> <action application="set" data="RECORD_COMMENT=CH call record"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="recording_follow_transfer=true"/> <action application="sleep" data="2000" /> <action application="managed" data="clsAsmtApp" /> </condition> </extension>

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Ring before answer
by Malay Thakershi 28 Jul '14

28 Jul '14

Hello, I have my inbound extension set as following. However, when people dial in, they don't hear any ring as one would normally expect. They directly hear first voice prompt after call is connected. How can I make it ring for a while say 2-3 rings? Thanks. In public.xml dial plan: <extension name="provider-inbound"> <condition field="destination_number" expression="^XXXXXXXXXX|XXXXXXXXXX$"> <action application="pre_answer" /> <action application="sleep" data="1000" /> <action application="transfer" data="1056 XML default" /> </condition> </extension> In default.xml dial plan: <extension name="CHPhone"> <condition field="destination_number" expression="^1056$">  <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 1980 Factory Records, Inc."/> <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/> <action application="set" data="RECORD_ARTIST=CH"/> <action application="set" data="RECORD_COMMENT=CH call record"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="recording_follow_transfer=true"/> <action application="sleep" data="2000" /> <action application="managed" data="clsAsmtApp" /> </condition> </extension>

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Latest master seems to consume more CPU for conferences
by Oleg Stolyar 28 Jul '14

28 Jul '14

Guys, I upgraded to the latest master and it seems that it is using 30%-50% more CPU per active conference compared to the previous master from late May 2014. Is this expected or am I missing something?

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Call Waiting + FIFO
by Chris Tunbridge 28 Jul '14

28 Jul '14

I have a script that creates fifo's on the fly, and at one point i had it worked with call waiting, but i don't remember what i did at that time, nor what changed (months and months ago) however, i cannot seem to get Calls in FIFO to trigger call waiting on the phones that are set as Members of that FIFO Queue. Basically if i have 2 calls come in, and i answer one, i want the 2nd call to ring through to my phone (as well as others) even though i am already on a call, Call Waiting currently works if i dial directly to the device (say extension to extension calling) or even by routing a call straight to an extension, then trying a second call to that extension. I have tested this with the same results on the following Aastra 6757i (Firmware 2.6.2019 and 3.3.something) Cisco SPA525G2 Polycom SoundPoint IP 550 Jitsi SoftClient X-Lite Soft Client. The behavior is the same regardless of the endpoint, i even tried creating loopbacks in order to trick the system, but from what i can tell, the server isn't even attempting to call the phone that's currently busy. The problem is that we have clients who get off one call, and have to wait 30+ seconds (fifo_member_timeout) before they can answer the next call, I've tried adjusting lag and it seems to have no impact. I'm working on getting a wireshark of this behavior, but i just wanted to reach out and see if the current behavior is in fact the correct one, and if not, or if there's a solution what i can do about it. Have i overlooked something ridiculously small? Here's what my fifo list <id> looks like http://pastebin.freeswitch.com/22943 Please let me know if you need anymore information.

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Re: [Freeswitch-users] Skinny shared reg and load balancing
by Nathan Neulinger 28 Jul '14

28 Jul '14

Yeah, this definitely isn't something supported by the code at this point. About the only way with the current code that I could think you _might_ be able to accomplish this would be coming up with something like a fallback dial of extension - configure each skinny extension that does the: <action application="bridge" data="skinny/internal/NNNNNNNNNNN"> to try that, and if the bridge fails, try a sip bridge via the other server. i.e. on server 1 something like: <action application="set" data="continue_on_fail=whatever-the-code-is" /> <action application="bridge" data="skinny/internal/AAABBBCCCC" /> <action application="bridge" data="sofia/gateway/server2/AAABBBCCCC" /> and on server 2: <action application="set" data="continue_on_fail=whatever-the-code-is" /> <action application="bridge" data="skinny/internal/AAABBBCCCC" /> <action application="bridge" data="sofia/gateway/server1/AAABBBCCCC" /> and then just have the port 2000 connection randomly selected for those phones. (Or use one of the sequential syntaxes like: <action application="bridge" data="skinny/internal/AAABBBCCCC|sofia/gateway/server1/AAABBBCCCC" /> Note - I haven't tried either of these approaches at this point (and going just on throwing together config idea), but if you do try it out and have success, please respond to the list with a summary of your resulting configuration in case it's of use to others. -- Nathan On 07/25/2014 04:51 AM, Nick Zaitsev wrote: > > Nathan, good day to you. > Thank you for your reply. > The thing is: Now i have opensips(dispathcer and load balancer modules) in front of multiple freeswitch servers to > balance register and invite requests,moreover i have shared mod_sofia between multiple fs,so,in that way i could make > calls between all of the nodes.So i would like to introduce skinny the same way. Could you advice how can i do that? > Because sharing reg in the database does not work with skinny, ext on different nodes can not reach each other. And,in > addition, i do not know how i can balance skinny to get redundancy. > Thank you for your time. > Best regards > > -- > Nick Zaitsev -- ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Re: [Freeswitch-users] multiple languages
by Neo Haux 28 Jul '14

28 Jul '14

Hi Darcy, I am trying to configure my voicemail to be in French too without success. Where did you put the line : <action application="set" data="default_language=fr"/> ? I added it in the dialplan/default.xml (extension "Local_Extension" that has the line <condition field="destination_number" expression="^(10[01][0-9])$">) <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> *<action application="set" data="default_language=fr"/>* <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> But I still get the voicemail in english Thanks, ForwardedMessage.eml Subject: Re: [Freeswitch-users] multiple languages From: Chris Tunbridge <blasterjr(a)gmail.com> Date: 07/28/2014 01:27 AM To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Glad i could point you in the right direction. On Sun, Jul 27, 2014 at 9:13 PM, Darcy Primrose <fs(a)voice2net.ca <mailto:fs@voice2net.ca>> wrote: Chris, ignore the last message, I had an english message forced, it is working correctly. Thank you again for your assistance. Darcy ----- Original Message ----- *From:* Darcy Primrose <mailto:fs@voice2net.ca> *To:* FreeSWITCH Users Help <mailto:freeswitch-users@lists.freeswitch.org> *Sent:* Sunday, July 27, 2014 11:03 PM *Subject:* Re: [Freeswitch-users] multiple languages Chris, thank you, I now get french or english on the voicemail depending on the endpoint setting. However, I still do not get French on the IVR, If I can solve that, I am in great shape, any ideas what I might be missing. Your help with this is very much appreciated. Darcy

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FreeSWITCH SLEEP
by Andre 28 Jul '14

28 Jul '14

After a call I have transfer after bridge on. When the call goes back to the dial plan I want to sleep the call do some stuff and hang-up. On the original bridge I do have hangup after bridge on. What is the correct syntax for me to sleep then hangup? Do I have to Answer the call for sleep to work? Example: After transfer go to : //do I need to Answer again for sleep to work? <action application="sleep" data="5000"/> // run some scripts not related to freeswitch <action application="hangup"/>

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Cisco 7906 + Freeswitch
by Nick Zaitsev 28 Jul '14

28 Jul '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 and the lots of NOTIFY from freeswitch, Could anybody help with this issue please? Best regards -- Nick Zaitsev

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