January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call recovery - Multi-primary / multi-backup scenario
by Carlo Dimaggio 26 Jan '12

26 Jan '12

Hi all, I would like to know if the track-calls feature could be used in a multi-primary - multi-backup scenario. What I think is an environment with Kamailio dispatcher to N freeswitch boxes with a (multi or) single freeswitch backup. In case of failure, one primary should be replaced with a backup (spare) box that will take its IP and active calls. If all active freeswitch track call state in the same table/database, how the backup box will know what calls should be recovered? Anyone could explain the call recovery algorithm? Best Regards,

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FreeSWITCH load CDR CSV to Mysql, theoretical problem?
by Henrik Aagaard Sørensen 26 Jan '12

26 Jan '12

I'm using the script on: http://convergence.pk/freeswitch-load-cdr-csv-to-mysql.html which also can be found on the http://wiki.freeswitch.org/wiki/Mod_cdr_csv. I'm having in running every 5 minute. And everything works. But I do have some concern. What if the job suddenly takes more than 5 minutes to complete and another job start, doing the same thing. Couldn't that be a problem and can it be fixed?

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Registration Failed with status DNS Error
by Antonio 26 Jan '12

26 Jan '12

Hi I have a problem registering a gateway, the return error is DNS error [503]. My DNS is OK, i can dig to the destination domain in question, and even when o do sofia_dig in fresswith it resolves the domain without problems. how does freeswitch resolves the destination domain? if you need more information, please let me know. Btw i just compile the last git head. Thanks, António ******** Log register. 2012-01-25 17:20:42.637192 [NOTICE] sofia_reg.c:414 Registering sol 2012-01-25 17:20:42.737189 [ERR] sofia_reg.c:2004 sol Registration Failed with status DNS Error [503]. failure #1 2012-01-25 17:20:43.637195 [WARNING] sofia_reg.c:473 sol Failed Registration [0], setting retry to 30 seconds. ******** sofia status gateway sol: freeswitch@internal> sofia status gateway sol ================================================================================================= Name sol Profile 192.168.10.25 Scheme Digest Realm mobile.commsmundi.com:5060 Username 2203 Password yes From <sip:2203@mobile.commsmundi.com:5060;transport=udp> Contact <sip:gw+sol@192.168.10.25:5060;transport=udp;gw=sol> Exten 2203 To sip:2203@mobile.commsmundi.com:5060 Proxy sip:mobile.commsmundi.com:5060 Context incoming Expires 1800 Freq 1800 Ping 1327508466 PingFreq 25 PingState -1/0/1 State FAIL_WAIT Status DOWN CallsIN 0 CallsOUT 0 FailedCallsIN 0 FailedCallsOUT 0 ================================================================================================= ******** sofia_dig: freeswitch@internal> sofia_dig mobile.commsmundi.com Preference Weight Transport Port Address ================================================================================ 1 0.200 udp 5060 192.168.10.1

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Gateway Configuration with No Authentication
by Vishal Kakkar 26 Jan '12

26 Jan '12

Hi all, We are trying to use FS to handle incoming call from SIP provider. Provider doesnt have any authentication because its all inbound traffic. They have configured their SIP trunk to point to our FreeSwitch Box. We have also configured one gateway in eternal/provider.xml with the IP of provider and disable the registration as provider is not expecting the same(they dont have any user configured for us, as they are using their box in trunk mode). Issue is that when an incoming call reaches to provider, we are not getting any INVITE from them and its being dropped at the IP that we have added as gateway. Reason shared is that they are expecting us to have heartbeat with their IP. When we turn ping option, it sends OPTION message but username is getting appended to TO and FROM uri's so that cause 404 No User found response. Please help to configure FS to send option something like this- *OPTIONS sip:10.129.43.154:5060;ttl=0 SIP/2.0* Via: SIP/2.0/UDP 10.129.43.23:5060; rport;branch=z9hG4bK- 6df4-1156281802-19999-423 Call-ID: 5aa9-1e61-7222006212322-chiloe-0(a)10.129.43.23 CSeq: 1 OPTIONS Max-Forwards: 70 *To: <sip:10.129.43.154;ttl=0>* *From: <sip:10.129.43.23>* ;tag=95ffcd055e0f78f7d5d397020e8 9288db5f2 User-Agent: Dialogic-SIP/10.3.2.57 chiloe 0 Contact: <sip:10.129.43.23:5060> Accept: application/ sdp Content-Length: 0 i.e. They want OPTION with only IP address not user@IP. So that on recieving any call they can transfer it to our FS. Please Help. Thanks -Manav

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MESSAGE Body Encoding
by John Rose 25 Jan '12

25 Jan '12

Does anyone know if the MESSAGE payload can have two encodings in one string when multi-byte text is present? Example: UTF-8 and CP-1252 ? Or is this dependent on the UA or API that populates the body? John

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Noobie question
by Craig Smith 25 Jan '12

25 Jan '12

How do I do a "dialplan show" in FreeSWITCH? I'm trying to troubleshoot why calls aren't going out through a gateway. Thanks in advance.

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Dialplan help
by Elliott Vogel 25 Jan '12

25 Jan '12

Hello, I'm trying to create a dial plan where a user is not registered (online) or an error sip message (480,486,500) is received; I want the call be sent to another sip address. I want to use within this context: <extension name="15618351100"> <condition field="destination_number" expression="^\+(15618351100)$" /> <condition field="caller_id_number" expression="^\+(\d+)$"> <action application="ring_ready" /> <action application="set" data="ignore_early_media=true"/> <action application="bridge" data="user/elliott(a)zoogmedia.com"/<mailto:user/elliott@zoogmedia.com"/>> </condition> </extension>

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Opus codec passthru (or terminate)
by Frank Rosengart 25 Jan '12

25 Jan '12

Hi, how do I have to configure my global codec prefs to pass thru opus codec calls? I guess with mod_opus it's already possible to terminate these calls, right? But anyway, pass thru would be ok. I'm now using baresip as useragent on both ends. Thanks! Frank

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FreeSWITCH ICE Support
by Kristian Kielhofner 25 Jan '12

25 Jan '12

Hello, Has anyone ever investigated adding ICE functionality to FreeSWITCH? While FreeSWITCH happily ignores ICE candidates in the SDP now it would be "really neat" if it could participate in ICE negotiation to find optimal media paths with other ICE implementations. -- Kristian Kielhofner

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Sendmail segfaulting before second message is sent
by Rob Morin 25 Jan '12

25 Jan '12

For some extensions, I've got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. Does anyone have any suggestions as to how I can troubleshoot this? (Running CentOS 5.7, sendmail 8.13). Thank you, Rob

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