April 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call is interrupted in 30 seconds, configuring NAT on Amazon EC2
by Стас Тельнов 11 Apr '16

11 Apr '16

Hello. When using a call or conference through sip — freeswitch with external provider there is a problem – the call is interrupted in 30 seconds. Though the sound goes all right. I think that it caused by the NAT settings for freeswitch, but I don't understand how to adjust it correctly. At start of freeswitch I see the following mistakes in the tracking data: 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP 1/5 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP 2/5 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP 3/5 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP 4/5 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP 5/5 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for PMP [general error] 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT devices detected! Despite of this mistake, conference communication between two internal users works normally. The problem arises at a call through external provider. We have the following architecture: In a cloud of Amazon EC2 there are 2 servers – opensips and freeswitch, both for NAT for external clients, but have an opportunity to work with each other directly. opensips has the internal address 172.31.0.169 and external 52. *.*.177 freeswitch has the internal address 172.31.22.124 and external 52. *.*.198 In fact, freeswitch acts only for conferences, and is ready for use of a remote DB on opensips. The auto-nat settings by default didn't work. The problem is in the external profile settings as far as I understand. I have filled and created the following configuration: vars.xml <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto”/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198”/> <!— public freeswitch ip —> <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198”/> <!— public freeswitch ip —>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/tls"/> sip_profile/external.xml <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value=“52.*.*.198”/> <!— public freeswitch ip —> <param name="ext-sip-ip" value=“52.*.*.198”/> <!— public freeswitch ip —> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server (that would be logical), but in that case conferences didn't work at all and errors below appeared: [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ... Also I tried to put such configuration: <param name="rtp-ip" value="auto"/> <param name="sip-ip" value="52.*.*.198”/> but it also hasn't helped to solve the problem. autoload_configs/switch.conf.xml <param name="rtp-start-port" value="16384"/> <param name="rtp-end-port" value="32768"/> "sofia status" looks as follows: Name Type Data State ================================================================================================= 172.31.22.124 alias internal ALIASED external profile sip:mod_sofia@52.*.*.198:5060 RUNNING (0) external profile sip:mod_sofia@52.*.*.198:5061 RUNNING (0) (TLS) external::*********.com gateway sip:USER@*********.com REGED internal profile sip:mod_sofia@52.*.*.198:5080 RUNNING (0) internal profile sip:mod_sofia@52.*.*.198:5081 RUNNING (0) (TLS) ================================================================================================= 2 profiles 1 alias "sofia status profile external" looks as follows: ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 172.31.22.124 Ext-RTP-IP 52.*.*.198 SIP-IP 172.31.22.124 Ext-SIP-IP 52.*.*.198 URL sip:mod_sofia@52.*.*.198:5060 BIND-URL sip:mod_sofia@52. *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp TLS-URL sip:mod_sofia@52.*.*.198:5061 TLS-BIND-URL sips:mod_sofia@52. *.*.198:5061;maddr=172.31.22.124;transport=tls HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA CODECS OUT PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0 What do I adjust wrong? Whether there is some opportunity, to tell freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?

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Re: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
by jelena＠misticnabica.hr 10 Apr '16

10 Apr '16

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freeswitch with postgres setup process?
by jungle Boogie 10 Apr '16

10 Apr '16

Hello All, I have freeswitch installed from source on Debian 8.3 64 bit and Postgresql 9.4 installed from the Postgres packages[1]. What's the best way to have Freeswitch use a database that I'll setup and create? Do I need to recompile freeswitch with certain modules enabled? Do I need to enable to config flags? I've completed these things: install unixodbc-dev install odbc-postgresql update /etc/odbc.ini: [freeswitch] ; WARNING: The old psql odbc driver psqlodbc.so is now renamed psqlodbcw.so ; in version 08.x. Note that the library can also be installed under an other ; path than /usr/local/lib/ following your installation. Driver = /usr/lib/x86_64-linux-gnu/odbc/psqlodbcw.so Description=Connection to LDAP/POSTGRESQL Servername=localhost Port=5432 Protocol=6.4 FetchBufferSize=99 Username=freeswitch Password=xxx Database=freeswitch ReadOnly=no Debug=1 CommLog=1 view etc/odbcinst.ini [PostgreSQL ANSI] Description=PostgreSQL ODBC driver (ANSI version) Driver=psqlodbca.so Setup=libodbcpsqlS.so Debug=0 CommLog=1 UsageCount=1 [PostgreSQL Unicode] Description=PostgreSQL ODBC driver (Unicode version) Driver=psqlodbcw.so Setup=libodbcpsqlS.so Debug=0 CommLog=1 UsageCount=1 And finally, test the connection: # isql -v freeswitch +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ [1] http://www.postgresql.org/download/linux/debian/ -- ------- inum: 883510009027723 sip: jungleboogie(a)sip2sip.info xmpp: jungle-boogie(a)jit.si

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Re: [Freeswitch-users] freeswitch with postgres setup process?
by jelena＠misticnabica.hr 10 Apr '16

10 Apr '16

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FreeSWITCH in cluster
by Dmitry Mordovin 10 Apr '16

10 Apr '16

Hi, I want configure two FreeSWITCH servers as cluster. As I understand, I need: - Configure one core DB (PostgreSQL) for each FreeSWITCH instances. - Add two A records in DNS to resolve domain name and load balancing. *Scheme, 2 FS servers and one DB.* FS-1 server 192.168.0.1 Core-DB server 192.168.0.100 FS-2 server 192.168.0.2 *Add two A records in DNS* mydomain.com. 43200 IN A 192.168.0.1 mydomain.com. 43200 IN A 192.168.0.2 I expect, FreeSWITCH instances will mirror each other in this configuration. So, if Bob register on 192.168.0.1 server, and Alice register on 192.168.0.2, they can call each other like local users. Also, in this case, SIP messages from Bob during the call session will sending to any of nodes, depend on DNS resolve. Am I right? Is this correct way to build cluster? Please direct me in right way. BR, Dmitry

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FreeSWITCH library for SRTP/DTLS
by saurabh verrma 09 Apr '16

09 Apr '16

Hi All, I want to use FreeSWITCH library for DTLS/SRTP support. I want to know in freeswitch which library has the support of these features(SRTP/DTLS) ? Is there any application available based on freeswitch library ? Any help would be appreciable. -- *With Warm Regards:* *Saurabh Kumar Verma*

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Mastering FreeSWITCH Book
by Anthony Minessale 09 Apr '16

09 Apr '16

Hi, We've been asked by packt to produce a 3rd book called "Mastering FreeSWITCH" Would anyone be interested in working on the authoring of that book and get their name in print? We need help with creating the content as well as formatting the text into the word templates they require etc. Please contact consulting(a)freeswitch.org if you are interested. -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900

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Scenario : registering phase
by amani mansour 09 Apr '16

09 Apr '16

Good morning , Can you help me please ?, I need to configure a number X in my soft phone . I want to do a scenario in the regestering phase which must be like this : soft --------------------register--------------------------------> server soft <-----------------unauthorized (first 401)----------------server soft ---------------------register (with authorization header computed with the nonce of the 401)----------------------------------------------------->server soft <--------------------unauthorized 401 (with stale ==False) ---->server i did this in dialplan/default.xml: <extension name="fail register"> <condition field="caller_id_number" expression="^400$"> <action application="respond" data="401"/> <action application="hangup"/> </condition> </extension> i have only one unauthorized then i have ok ,but how can i do to have this result . thank you with best regards , amani

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Re: [Freeswitch-users] detecting tone
by jelena＠misticnabica.hr 09 Apr '16

09 Apr '16

1 0

detecting tone
by Babak Yakhchali 09 Apr '16

09 Apr '16

Hi I want to detect the tone that is attached to this email with spandsp. what descriptor should I use. I'm trying 49 but it is not working. thanks

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