December 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem with mod_sms
by peetee23＠hushmail.com 27 Dec '13

27 Dec '13

Dear All, I have a problem with mod_sms. Configuration of chatplan is as this: Here is FS log: 2013-12-26 21:14:21.841090 [INFO] mod_sms.c:336 Processing text message 1001->1004 in context public Chatplan: 1004 parsing [public->default] continue=false Chatplan: 1004(a)test.local Regex (PASS) [default] to(1004(a)test.local) =~ /^(.*)$/ break=on-false Chatplan: 1004(a)test.local Action send() 2013-12-26 21:14:21.841090 [DEBUG] sofia_presence.c:224 Can't find registered user 1004(a)test.local But user is registered: in fs_cli #sofia_contact 1004 sofia/internal/sip:1004@1.2.3.4:40348;rinstance=72ECC02F;transport=tls;fs_nat=yes;fs_path=sip%3A1004%4014.3.2.1.%3A40348%3Brinstance%3D72ECC02F%3Btransport%3Dtls Out of default configuration I have just forced TLS connection (tlsv1). Dont you have an idea where could be problem? Thanks everyone for help PT

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TLS error?
by Federico Beffa 27 Dec '13

27 Dec '13

Hi, I'm trying to setup FS as a gtalk client. I'm working on Linux Ubuntu 10.04 and have cloned the git FS repository about 1 week ago. I can compile everything with no errors. However, after configuring client.xml and starting FS I see the following error message in the freeswitch.log file: 2010-07-20 23:38:56.392853 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_ dingaling.so **/opt/freeswitch/mod/mod_dingaling.so: undefined symbol: gnutls_global_init** TLS comes from the following Ubuntu packages: libgnutls26, libgnutls-dev. I would appreciate any suggestion on how to investigate further the problem. Could it be a gnutls library version incompatibility? A bug in FS? In the ubuntu gnutls packages? Or simply an error in the configuration? Thanks for you help! Fede

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Problems about STUN and one way audio
by alex 27 Dec '13

27 Dec '13

Hi, all I have a server that has installed the FreeSwitch software, and the server has a public IP addr. My clients are android phones, the CSipSimple softphone runs on them. There are some issues when these phone call each other. When phone A calls phone B, only the phone B can listen the audio coming from phone A, but the phone A cannot. What's more, the phone B always need some delays to listen the audio from phone A. Therefore, I guess I need set a STUN server to deal with the above problem. So I read related web pages, such as http://wiki.freeswitch.org/wiki/NAT_Traversal. But, How to configure the STUN server in FreeSwitch is ambiguous. Then, I tried to make some configurations as below: vars.xml. <X-PRE-PROCESS cmd="set" data="external_rtp_ip=a.b.c.d"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=a.b.c.d"/> internal.xml <param name="ext-rtp-ip" value="$${ external_rtp_ip }"/> <param name="ext-sip-ip" value="$${ external_sip_ip }"/> external.xml <param name="manage-presence" value="true"/> <param name="ext-rtp-ip" value="$${ external_rtp_ip }"/> <param name="ext-sip-ip" value="$${ external_sip_ip }"/> But, I didn't find where could I set the port of STUN server. So I failed to launch the STUN server. Thus, Could you tell me How to solve the one-way audio problem, and How to configure the STUN server in FreeSwitch? Thanks and sorry if my English is poor. Best regards!

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Grandstream DP710 & DP715 Working Configuration
by Sean Devoy 27 Dec '13

27 Dec '13

Hi All, I didn't see any references to these GrandStream DECT phones in the archives or wiki. I was able to get them working with some minor problems. The "Primary Sip Server" and "Secondary Sip Server" fields in the setup work great with IP Addresses. However, they do not like MY DNS names as they a have underscores in them. My FS is multi-tenant so connecting by name is quite critical. GrandStream identified the problem as the underscore and says it will be fixed in the next firmware release. Anyway, here is a quick review of the phones. The DP715 is the "base station" for the DECT handsets and comes with one handset. It can support 5 handsets and 4 concurrent calls. Each handset is a unique SIP Registration. The voice quality is decent with no background noise. I was a little disappointed that the handsets have no speaker phone or headset options, but that is not so critical in home environments. The price is REALLY good, roughly $65 for DP715 and $35 for each additional DP710. I have only just started looking into these, but that is FAR cheaper than the CISCO offering. Even though the CISCO devices are much more expensive, they offer much more like an FXO and FXS port on the base. I hope to try them soon. Overall I give the DP715/DP710 3 stars for features (out of 5) and 4 stars for performance. My Directory settings for these phones are: <user id="XXXXXX">  <variables> <variable name="outbound_caller_id_name" value="XXXXXXXXXXXXXXXX"/> <variable name="outbound_caller_id_number" value="XXXXXXXXXXX"/> <variable name="internal_caller_id_name" value="XXXXXXXXXXXXXXXX"/> <variable name="internal_caller_id_number" value="221"/> <variable name="user_context" value="from-internal-XXXXXX"/> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> </variables> <params> <param name="password" value="XXXXXXXX"/> <param name="manage-shared-appearance" value-="false" />" <param name="NDLB-force-rport" value="safe"/> </params> </user>

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fisheye shows No activity found for a few days
by Seven Du 26 Dec '13

26 Dec '13

Just found Fisheye shows No activity found. for a few days, any problem on the back end? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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ORIGINATOR_CANCEL and dialplan proccessing
by Anton Kvashenkin 26 Dec '13

26 Dec '13

Hello guys, I need to run system command after getting ORIGINATOR_CANCEL status message. Judging by http://jira.freeswitch.org/browse/FS-3561 i can get it working by adding <action application="set_zombie_exec"/> to my dialplan. But it proceedes anyway even after successfully hanging up call. My snippet of dialplan: http://pastebin.freeswitch.org/21771

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TLS Audiocodes
by Szasz Szabolcs 26 Dec '13

26 Dec '13

Hi, Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to work TLS between freeswitch and a softphone (phonerlite), but I have problem with Audiocodes during the TLS authentication. I've loaded the certification but it still doesn't work. Can I debug the tls in freeswitch? Please help me. Thank you in advance. Szasz Szabolcs

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Client TLS certificate setup
by adahary 26 Dec '13

26 Dec '13

I have already installed the CA certificate in the FS conf/ssl (following FS/TLS wiki) and managed to get TLS registering with PhonerLite so I know that basically it should work with other sip/tls clients as well.I'm trying to setup other softphones with TLS/SIPS connection to Freeswitch and I'm failing to register with CSimpleSip/android and Jitsi/windows. I get this:tport_type_tls.c:607 tport_tls_accept() tport_tls_accept(0xb6e02268): new connection from tls/1.2.3.4:13886/sipstport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATINGtport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATINGtport_tls.c:962 tls_connect() tls_connect(0xb6e02268): TLS setup failed (error:00000001:lib(0):func(0):reason(1))tport.c:2092 tport_close() tport_close(0xb6e02268): tls/1.2.3.4:13886/sipsI will appriciate any help/instruction/tip on how to setup working TLS connection with CSimpleSip/android and Jitsi/windows clients.Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Client-TLS-certificate-setup-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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TLS problem
by Dror Lupu 26 Dec '13

26 Dec '13

Hi Guys, I've done the steps for TLS setup as shown on the Wiki (using the server's IP address in CN and ALT), but I can't get TLS to work (UDP/TCP works fine). All certificates seem to be in place and generated correctly. I get this error, when using Linphone (I installed the root CA on my machine). 1. tport.c:2745 tport_wakeup_pri() tport_wakeup_pri(0xb5062a60): events IN 2. tport.c:869 tport_alloc_secondary() tport_alloc_secondary(0xb5062a60): new secondary tport 0xb67c0410 3. tport_type_tls.c:607 tport_tls_accept() tport_tls_accept(0xb67c0410): new connection from tls/172.16.111.211:61942/sips 4. tport_tls.c:873 tls_connect() tls_connect(0xb67c0410): events NEGOTIATING 5. tport_tls.c:873 tls_connect() tls_connect(0xb67c0410): events NEGOTIATING 6. tport_tls.c:962 tls_connect() tls_connect(0xb67c0410): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) 7. tport.c:2092 tport_close() tport_close(0xb67c0410): tls/172.16.111.211:61942/sips 8. Any ideas? Thanks in advance, Dror.

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How to register sip phone in VOIP Disabled Areas
by Shahzad Bhatti 25 Dec '13

25 Dec '13

Hello everybody, i want to know that is there any possibility that i can register on Free Switch from a location where port 5060 is not allowed from ISP like i am in pakistan and here i m unable to register on FreeSwitch Server using DSL Broadband Service. Regards Shahzad Bhatti

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