- FreeSWITCH-users - FreeSWITCH Mailing Lists

XML_CURL max connection limit
by Muhammad Shahzad 13 Jun '13

13 Jun '13

Hi, Is there any connection limit perhaps hard coded anywhere in xml_curl module? I am load testing FS with xml_curl to provide config, directory and dial-plan. I observed i start getting HTTP ERROR 0 when i have around 1600 concurrent calls on machine with media. Although i see no sign of system overload either on FS (load average < 1.8) or Apache (load average < 4,75). Both have plenty of RAM and bandwidth free. Is there anyway to get rid of this HTTP ERROR 0? It must be generated by FS internally, since i don't see anything like this error in Apache logs. Below 1,638 concurrent calls @ 50 CPS, i get success rate of 99.999% but as soon as it crosses this mark, success rate drastically drops to 60% and even below (when concurrent calls go beyond 2,000). Here is error console log. ################################################### 2013-06-05 18:09:13.591229 [ERR] mod_xml_curl.c:310 Received HTTP error 0 trying to fetch http://172.16.171.61/vm-users.php data: [hostname=vms01&section=directory&tag_name=domain&key_name=name&key_value=10.1.3.2&Event-Name=GENERAL&Core-UUID=84c917dc-cdde-11e2-9bb2-8b779d6a4e97&FreeSWITCH-Hostname=vms01&FreeSWITCH-Switchname=vms01&FreeSWITCH-IPv4=10.1.3.2&FreeSWITCH-IPv6=%3A%3A1&Event-Date-Local=2013-06-05%2018%3A09%3A03&Event-Date-GMT=Wed,%2005%20Jun%202013%2013%3A09%3A03%20GMT&Event-Date-Timestamp=1370437743591210&Event-Calling-File=mod_voicemail.c&Event-Calling-Function=resolve_id&Event-Calling-Line-Number=1350&Event-Sequence=1776930&action=message-count&key=id&user=989000049598&domain=10.1.3.2] ################################################### Although this setup is more then sufficient for my requirement but i just want to know if i can extend it beyond, since i still have lot of resources free. OS: Debian 6.0.7 x64 Apache: 2.2.16 FreeSWITCH: 1.5.1b+git~20130424T005233Z~66d1755b06 Both FS and Apache are running on separate dedicated machines on same LAN. Thank you. -- Mit freundlichen Grüßen Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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Gateway Call Limits
by Alex Ynema 12 Jun '13

12 Jun '13

Hi, I'm trying to figure out how to limit the number of calls a Gateway is allowed to use. Our Sip provider has provided up with 200 which I need to set within the system somehow. What's the best way to handle it for an outgoing only system. I've been trying to figure out how how to configure 'Rate limiting concurrent outgoing calls via a provider' which is mentioned in the wiki a bit but nothing specific on what to actually do. *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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sending 183 WITHOUT SDP from dialplan
by Matthew Meek 12 Jun '13

12 Jun '13

I am attempting to setup FS as an B2BUA SBC between a telco platform and a telco SIP trunk. Everything is fine except for one call-flow scenario: We need to sometimes "reject" calls by sending reason code 17, which is quite easy by sending a SIP 486. The issue is that the far-end first receives early media and then the reason code 17, and the far-end switch is bonding the B-leg media for 55 seconds instead of moving on in the dial-plan. When comparing this to our PRI build-out, the difference is that in the PRI-buildout, no early-media is sent. In the SIP world, we are sending a 183 with SDP to setup the early media... If I don't send early media, the telco will inject their own early media path... so I need a work-around. The telco is asking that we send specifically a 183 WITHOUT SDP, which I am having trouble figuring out how to do. I know how to send the 180 WITHOUT SDP, a 183 WITH SDP, but I cannot find the solution to sending 183 WITHOUT SDP... I know it is unusual to send 183 without SDP, but it is still RFC compliant... Thoughts? Matthew Meek

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Is this kind of whisper possible?
by Jeff Bernhardt 12 Jun '13

12 Jun '13

I'm wondering if the following is possible: Caller A is on a call with caller B. Caller C (maybe a secretary) has an urgent message for caller A and wants to barge into the call so that only caller A can hear him/her. This is called "whisper" or "agent coaching," which I believe is possible... BUT, is it possible for caller B to then push a key combination and whisper back to caller C without caller A hearing? For instance, the secretary whispers into the boss, "Hey, I have Mr. Jones on another call and he says it's urgent. Do you want to take it?" and the boss wants to respond to the secretary right there without putting the current caller on hold, so he hits *500 or something and tells the secretary, "Tell Mr. Jones I'll call him back" without disrupting the current caller babbling on. Is this possible? Jeff

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FreeSWITCH Community Conference Call Today
by Michael Collins 12 Jun '13

12 Jun '13

Hello FreeSWITCHers, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_12 On this discussion we're going to look at the CudaTel. We are interested in showing the community at large what the FS team has been up to. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Timer Not Configured
by Cesar Bermudez 12 Jun '13

12 Jun '13

Hi all. I m testing the conference, and i want to record it, but i get this: 2013-06-11 10:35:56.794857 [ERR] mod_conference.c:3942 Error Opening File [/opt/freeswitch/recordings/adn_open/adn_open_2013-06-11-10-35-56.wav] 2013-06-11 10:35:56.794857 [ERR] switch_core_timer.c:117 Timer is not properly configured. 2013-06-11 10:35:56.794857 [DEBUG] mod_conference.c:4061 Recording of /opt/freeswitch/recordings/adn_open/adn_open_2013-06-11-10-35-56.wav Stopped 2013-06-11 10:35:56.814847 [DEBUG] mod_conference.c:2114 Auto recording file: /opt/freeswitch/recordings/adn_open/adn_open_2013-06-11-10-35-56.wav 2013-06-11 10:35:56.814847 [ERR] mod_sndfile.c:202 Error Opening File [/opt/freeswitch/recordings/adn_open/adn_open_2013-06-11-10-35-56.wav] [System error : No such file or directory.] 2013-06-11 10:35:56.814847 [ERR] mod_conference.c:3942 Error Opening File [/opt/freeswitch/recordings/adn_open/adn_open_2013-06-11-10-35-56.wav] 2013-06-11 10:35:56.814847 [ERR] switch_core_timer.c:117 Timer is not properly configured. But the folder have perms, its owned by FS and have 777. this is the version that i m using. FreeSWITCH Version 1.5.2b+git~20130607T172942Z~a433b97375 (git a433b97 2013-06-07 17:29:42Z) But i m running on a Vmware Sphere VM machine, ( last Centos is the SO ) any recomendations? Regards.

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Re: [Freeswitch-users] ./configure --enable-core-pgsql-support and mod_xml_scgi
by kellon.langdon＠sis-grenada.com 12 Jun '13

12 Jun '13

Hi All I have attempted to compile freeswitch with support for pgsql but the compile fails at mod_xml_scgi Prior to compiling with --enable-core-pgsql-support we have compiled freeswitch without any configuration options and the freeswitch server has comiled successfully. We have also compiled freeswitch with --enable-core-odbc-support configure option and that also compiles successfully. Does anyone have or know of a fix for this problem. i have trolled the Freeswitch User Forum and found a number of suggested fixes, from removing -Werror in Makefiles and removing spidermonkey. None of these have fixed the problem. We have used various git downloads and also download version 1.2.10 bz2 version. Still no success. Has this problem been fixed or is it still an outstanding issue?? Thanks Kellon

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Re: [Freeswitch-users] Cisco phone registration
by andpe 12 Jun '13

12 Jun '13

Hi I've seen once your article :-) It is very interesting. However, C 7931 phone behaves as described. I hope this is a configuration error. Without the correct value "realm" I can not use multitenant in FreeSWITCH. I send you the configuration. In wireshark I see that TO and FROM fields contain the IP and not the realm (domain name). No matter how I set to challenge-realm in FreeSWITCH (auto_from or auto_to). Anyway, this is understandable when the phone sends the wrong as I had expected. In FreeSWITCH console, I get the message: [WARNING] sofia_reg.c: 2515 Can not find user [1000 @ abcd] from abcf You must define a domain called 'ABCD' in your directory and add a user with the id = "1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. config: <device> <fullConfig>true</fullConfig> <deviceProtocol>SIP</deviceProtocol> <sshUserId>xxx</sshUserId> <sshPassword>xxx</sshPassword> <devicePool> <dateTimeSetting> <dateTemplate>M/D/Y</dateTemplate> <timeZone>your timezone</timeZone> <olsonTimeZone>your timezone</olsonTimeZone> <ntps> <ntp> <name>x.x.x.x</name> <ntpMode>Unicast</ntpMode> </ntp> </ntps> </dateTimeSetting> <callManagerGroup> <tftpDefault>true</tftpDefault> <members> <member priority="0"> <callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> </ports> <processNodeName>mysip.server.domain</processNodeName> </callManager> </member> </members> </callManagerGroup> </devicePool> <commonProfile> <phonePassword/> <backgroundImageAccess>true</backgroundImageAccess> <callLogBlfEnabled>0</callLogBlfEnabled> </commonProfile> <loadInformation>SIP31.9-3-1-1S</loadInformation> <vendorConfig> <disableSpeaker>false</disableSpeaker> <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset> <pcPort>1</pcPort> <settingsAccess>2</settingsAccess> <garp>0</garp> <voiceVlanAccess>0</voiceVlanAccess> <videoCapability>0</videoCapability> <autoSelectLineEnable>0</autoSelectLineEnable> <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive> <displayOnTime>10:00</displayOnTime> <displayOnDuration>00:01</displayOnDuration> <displayIdleTimeout>00:05</displayIdleTimeout> <webAccess>0</webAccess> <spanToPCPort>1</spanToPCPort> <loggingDisplay>1</loggingDisplay> <loadServer/> </vendorConfig> <deviceSecurityMode>1</deviceSecurityMode> <authenticationURL/> <directoryURL/> <idleTimeout>10</idleTimeout> <idleURL/> <informationURL/> <messagesURL/> <proxyServerURL>mysip.server.domain</proxyServerURL> <servicesURL></servicesURL> <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig> <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> <dscpForCm2Dvce>96</dscpForCm2Dvce> <transportLayerProtocol>4</transportLayerProtocol> <capfAuthMode>0</capfAuthMode> <capfList> <capf> <phonePort>3804</phonePort> </capf> </capfList> <certHash/> <encrConfig>false</encrConfig> <sipProfile> <sipProxies> <backupProxy></backupProxy> <backupProxyPort>5060</backupProxyPort> <emergencyProxy/><emergencyProxyPort/> <outboundProxy/>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>true</rfc2543Hold> <callHoldRingback>2</callHoldRingback> <localCfwdEnable>true</localCfwdEnable> <semiAttendedTransfer>true</semiAttendedTransfer> <anonymousCallBlock>2</anonymousCallBlock> <callerIdBlocking>0</callerIdBlocking> <dndControl>0</dndControl> <remoteCcEnable>true</remoteCcEnable> </sipCallFeatures> <sipStack> <sipInviteRetx>6</sipInviteRetx> <sipRetx>10</sipRetx> <timerInviteExpires>180</timerInviteExpires> <timerRegisterExpires>600</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>true</transferOnhookEnabled> <enableVad>false</enableVad> <preferredCodec>g729</preferredCodec> <dtmfAvtPayload>101</dtmfAvtPayload> <dtmfDbLevel>3</dtmfDbLevel> <dtmfOutofBand>avt</dtmfOutofBand> <alwaysUsePrimeLine>false</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <stutterMsgWaiting>1</stutterMsgWaiting> <callStats>false</callStats> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> <startMediaPort>16384</startMediaPort> <stopMediaPort>16399</stopMediaPort> <voipControlPort>5060</voipControlPort> <dscpForAudio>184</dscpForAudio> <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> <dialTemplate>dialplan.xml</dialTemplate> <phoneLabel>LABEL</phoneLabel> <sipLines> <line button="1"> <featureID>9</featureID> <featureLabel>1000</featureLabel> <proxy>USECALLMANAGER</proxy> <port>5060</port> <name>1000</name> <displayName>1000</displayName> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>3</callWaiting> <authName>1000</authName> <authPassword>xxxx</authPassword> <contact>1000</contact> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>1</messageWaitingLampPolicy> <messagesNumber>121</messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>false</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> </line> </sipLines> <softKeyFile>SoftKey.xml</softKeyFile> </sipProfile> </device> Andy W dniu 2013-06-11 15:28:14 użytkownik Cal Leeming [Simplicity Media Ltd] <cal.leeming(a)simplicitymedialtd.co.uk> napisał: Hello, The phone shouldn't be sending the IP address, it should be using the hostname you specified. I actually did an article, albeit about a NAT issue, on the Cisco 7940 which goes into detail about the SIP packets going to and from the server [1]. As you can see from this article, the phone is sending the hostname in the To field. Therefore, this could either be a dodgy firmware or some config option you have set wrong. Can you please give us the following; * FS SIP trace logs showing the Cisco phone during registration and/or call state * Cisco phone config (this can be extracted using the telnet server if you are not using TFTP) Hope this helps Cal [1] http://blog.simplicitymedialtd.co.uk/476/analysis-of-cisco-7940-sip-alg-and… On Tue, Jun 11, 2013 at 9:30 AM, andpe <andpe(a)poczta.onet.pl> wrote: Hi I have a problem with the registration of the Cisco 79xx phones. The SIP Message Header (MH) which sends the phone, in the fields "TO" and "FROM", places the IP address instead of the domain name. The configuration of the phone is set to the domain name (xx.yy.com.) Is there any way to set the phone to send the domain name in the field instead of an IP address? Is there a firmware version that sends MH correctly? Andy _________________________________________________________________________ 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

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Transfer and scoping confusion
by admin 12 Jun '13

12 Jun '13

Hi Guys, I am climbing the learning curve, but I am at a standstill on this. I have searched both books and online for this, and have experimented with limited success. I have setup the standard FS box, a little behind at v.1.2.5.3 and I have managed to fumble my way though TDM connections, dialplan extensions, and even have HylaFax functioning. I am now "stretching" myself by adding some requested user features. The first is answering a call and sending directly to voicemail instead of the default rerun through the default local extension. >From the 40,000 ft view, the incoming, internal (Local_Extension) call, a-leg, has been answered and bridged to an internal phone, b-leg. After some communication with the caller, the person on the b-leg elects to send the caller directly to voicemail. I have successfully implemented a "bind_meta_app" to "execute_extension" to a new extension I called direct_to_vmail. Here I "read" the four digit phone number entered. At this point all is well, at least according to the logs. At this point I need to "unbridge" the b-leg and bridge the a-leg to voicemail. Can't find an unbridge, break looks wrong in this... Transfer appears to be what I need, as it says it "transfers the call into the new context" and seems to infer that an "unbridge" will occur. OK so far? Now, if I transfer, it needs to go to a specific extension that understands the request, so transfer direcly out to another extension by extension name. Logs confirm this does happen. Here is my question about variable scoping. When I "read" the four digit phone number entered I first assumed it created a channel variable, and the log seems to indicate this and seems to follow the transfer used in the features.xml in "is_transfer". This example uses the "read" variable to transfer back into the dialplan. I need to transfer into a particular extension and use this channel variable to know the voicemail requested. I test the channel variable prior to the transfer statement and after entry into the called extension. Initial tests show assignment before transfer, no assignment after which tends to indicate to me a scoping problem. I have tried import and export statements with no success. I defined the channel variable back in the Local_Extension extension before the "execute_extension" to see if it was a variable being declared local later in the call stack... No luck as assignment appears to take place, but not later in the call chain. I know this is long and "wordy", I just think this is a problem with understanding the basics of FS and I don't want to get bogged-down with specific XML statements until I know I understand the basics. Any Ideas? What am I missing? Thanks James

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FS as a REDIRECT server
by adahary 12 Jun '13

12 Jun '13

I'm planning to have several FS in few regions and would like that each client will be able to register with the nearest FS in his current region location - regardless of the client's @domain (using multiple clients doamins). I would like to use the following registration procedure: 1. A client is turned ON (new location) and sends REGISTER request to the REDIRECT server at 'redirect.fs.com' (all clients will be provisioned with the same registrar url); 2. The REDIRECT server looks for the nearest FS server and sends 302-redirect to the client 3. The client receives the 302 and re-registers to the nearest FS server. 4. The targeted FS update the LOCATION DB with the current client FS server location for incoming call routing . Is it possible to setup a FS as a REDIRECT server? How? *if there is a better approch then please advise Regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-as-a-REDIRECT-server-tp759… Sent from the freeswitch-users mailing list archive at Nabble.com.

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