January 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Debian helper script fails
by Francis 08 Jan '14

08 Jan '14

I'm getting an error message when running the debian package build helper script: mk-build-deps: Unable to find package name in control file `freeswitch/debian/control' I get this with the following environment: Debian 7.3 Wheezy 32bit Running on Virtualbox Cheers, Francis

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Multiple m-lines
by Stephen Wilde 08 Jan '14

08 Jan '14

On of my providers cannot handle INVITEs with 2 m-lines but in some cases, probably depending on what receives, my Freeswitch sends an INVITE with 2 m audio lines, one for ptime=10ms and one for ptime=20ms. Attached a screenshot. I have added the following line in dialplan: <action application="export" data="sdp_m_per_ptime=false" /> without success. Any suggestion? Stephen [image: Inline image 1]

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multiple registration + number-alias
by tilt 08 Jan '14

08 Jan '14

Hello all, I am trying to utilize number-alias and ring multiple registrations to the same endpoint. So far I have the multiple endpoints ringing, but I cant find a way to incorporate the number-alias into the dialplan xml "Local_Extension" The below works for the multiple endpoint ringing per single registration/user (http://wiki.freeswitch.org/wiki/Sofia-SIP#Multiple_Registrations) <action application="bridge" data="${sofia_contact(${dialed_extension}@${domain_name})}"/> *Problem*: How can I also incorporate the number-alias feature, so that both features work? Here is the example in wiki (http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Alphanumeric_to_nu…) <action application="bridge" data="user/${user_data(${destination_number}@${domain_name} attr id)}"/> Any help would be much appreciated. Thanks, John

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Recording during a failover recovery
by Brian Fertig 08 Jan '14

08 Jan '14

Greetings, I am working on a scenario where a leg of a call is being recorded. In the event of a failover recovery the recording persists which is great. During testing I found that when the failover takes place the failover FS server overwrites the recording originally created by the original FS server. Is there a way to cause the failover server to append to the recording and not overwrite it? Brian Fertig, CCNA, MCSE Telecom SME [Description: Description: Description: DoublePositive_Sig_Logo] bfertig(a)doublepositive.com<mailto:bfertig@doublepositive.com> Office: 410.332.0464 Toll-free: 1.888.376.7484 Headquarters: 1501 S. Clinton Street, Suite 1520 Baltimore, Maryland 21224 www.doublepositive.com<http://www.doublepositive.com> [Description: Description: Description: Description: Description: facebook_16]<http://www.facebook.com/pages/DoublePositive-Marketing-Group/73483219615> [Description: Description: Description: Description: Description: google_16] <https://plus.google.com/u/0/109428004071873072118/up/start/#109428004071873…> [Description: Description: Description: Description: Description: linkedin_16] <http://www.linkedin.com/company/90530> [Description: Description: Description: Description: Description: twitter_16] <http://twitter.com/doublepositive> [Description: Description: Description: Description: Description: youtube_16] <http://www.youtube.com/doublepositiveVideos>

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Freeswitch + TLS with a commercial certificate
by Iskren Hadzhinedev 08 Jan '14

08 Jan '14

Greetings. I'm unable to setup TLS and SRTP. I have a valid certificate from GlobalSign and my setup is currently the following: My certificate and key (merged with cat keyfile certfile > agent.pem) in /opt/freeswitch/conf/ssl/agent.pem The GlobalSign root certificate is in /opt/freeswitch/conf/ssl/cafile.pem I edited vars.xml as instructed from http://wiki.freeswitch.org/wiki/SIP_TLS#Configuration I tried running with tlsv1 and sslv23 in vars.xml, verified that FS is listening on ports 5061 and 5081 with netstat -nltp | grep freeswitch Also I get TLS listeners with "sofia status" so it should be working. Connecting to ports 5061 and 5081 with openssl s_client connect freeswitch.lan:<port> is successful, but I get a 'Verify return code: 21 (unable to verify the first certificate)'. Running nginx with the agent.pem as a certificate is working without any issues. When I try to connect to Freeswitch via TLS with Bria and Linphone 3.6.1 I get errors 408 or 503 and I don't see any output into the freeswitch console where I enabled sofia siptrace globally. What is the correct way to setup Freeswitch with a commercial certificate in order to enable TLS and SRTP ? Thank you! Kind regards, -- Iskren Hadzhinedev System Administrator The Idea Factory | 20 Mearns Street | Aberdeen | AB11 5AT | UK T: 01224 607500 VAT Reg No: 982 4936 74. Company registered in Scotland, SC237116

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Welcome to 2014!! FreeSWITCH Weekly News and Notes
by Ken Rice 08 Jan '14

08 Jan '14

Hey guys, Hope you all had a good holiday break! Don¹t forget today at 1PM EST (10AM PST) will be ClueCon Weekly at sip:888@conference.freeswitch.org for PSTN dial in info see http://fs0.us/call888 Todays topic is community scrum and Q&A Come join the FS Dev Team and ask your questions! ClueCon is moving forward! If you want to sponsor this year, contact the ClueCon team via marketing(a)freeswitch.org Have a Great Week! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Codec trans coding
by Shahzad Bhatti 08 Jan '14

08 Jan '14

i have a case that freeswitch receive calls from different sip phones supporting different codecs i want to trans code and then pass to leg b in a way that every codec is trans code to PCMA / PCMU like iLBC -> PCMU Speex -> PCMA Silk -> PCMU but if leg A offer G729 then it passes to leg B with out trans coding what is the best way to do that regards Shahzad Bhatti

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How to get the network address of registered Endpoint (NAT Problem)
by Mitja 08 Jan '14

08 Jan '14

Hello there, Im trying to solve NAT related Problem which is caused by missbehaving SIP phone. Up Front: I searched the wiki and the web for all the NAT related sites/mailling list. I think its a error of the invoxia IPhone docking station (the missbehaving SIP phone), thus I did not open a jira for it. We have a freeswitch server with internet accessibility and a Phone behind a natting router (Fritz!Box). FreeSWITCH (1.2.3.4) <-> (10.11.12.13) NAT (192.168.0.1) <-> (192.168.0.2) SIP Phone The Phone is able to register and initiate calls. But when someone else to call that endpoint, its unreachable. I did some research and the problem seems to be that the phone registers with a Contact containing the private IP Adress (192.168.0.2). freeswitch@1.2.3.4@internal> sofia status profile internal reg Registrations: ================================================================================================= Call-ID: aeac7f28-a6a0-4ba1-9d7c-30934924726d User: 270(a)1.2.3.4 Contact: "user" <sip:270@192.168.0.2:52767;ob> Agent: invoxia-lemonvoice-v6.17.4 Status: Registered(UDP)(unknown) EXP(2014-01-06 13:06:35) EXPSECS(134) Host: ippbx-border IP: 10.11.12.13 Port: 52767 Auth-User: 270 Auth-Realm: 1.2.3.4 MWI-Account: 270(a)1.2.3.4 Other devices in the same Setup have their NAT IP [+Port] (10.11.12.13:52767) in their Contact Header which then works fine. In this setup however the FreeSWITCH tries to send the INVITE Message to 192.168.0.2 and even when a NAT Table on the firewall manages to map it to 10.11.12.13:52767 the answer from 10.11.12.13 is discarded. Normally I use bridge with user/270(a)1.2.3.4 with the dialstring "{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(internal/${dialed_user}@${dialed_domain})}". I tried to force freeswitch to use the NAT IP address by calling bridge with "sofia/internal/sip:270@10.11.12.13:52767;ob" which worked fine. The IP address is of course not static but if I would be able to extract the IP information which is stored in sofia registry (see above) I could arrange my dialplan in a way that would fix that phones missbehaviour. So after all that describing text (sorry for that), heres my question: Is there a way (an API or Dialplan command) to get the IP Information which is stored the sofia registry. Thanks in advance Regards, Mitja

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how to list users registered from xml_curl
by Jimmy Chang 08 Jan '14

08 Jan '14

Hi, Is there any way that I can list users registered from xml_curl? Thanks. Jimmy --- 由於 avast! 防毒軟體已開啟防護，所以此封電子郵件將不受病毒及惡意軟體的侵害。 http://www.avast.com

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how to get the gateway name which the call is using
by gggyniidt 08 Jan '14

08 Jan '14

Hello My dialplan like this: <extension name="test"> <condition field="destination_number" expresion="^(100)$"> <action application="bridge" data="sofia/gateway/gw/100|sofia/gateway/gw1/100"/> </condition> </extension> I want to know which gateway was using by the call, gw or gw1? is there any variables to use? Thanks!

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