- FreeSWITCH-users - FreeSWITCH Mailing Lists

Is it possible to use parameters from vars.xml in auto_config/*xml ?
by Yehavi Bourvine 27 Aug '15

27 Aug '15

Hi, I am using ODBC for Freeswitch's core, and have to define it multiple times in the configuration files under auto_config directory. Is it possible to use values set in vars.xml inside these configs, so I'll have to set the db's parameters only once? Thanks, __Yehavi:

2 2

Freeswitch IAX2 support
by Bharat Lalcheta 27 Aug '15

27 Aug '15

Hello All, I am trying to enable IAX2 support for freeswitch. I successfully compiled and install PTLIB and OPAL. Also recompiled freeswitch with mod_opal successfully. Below is opal.conf. <configuration name="opal.conf" description="Opal Endpoints"> <settings> <param name="trace-level" value="25"/> <param name="context" value="default"/> <param name="dialplan" value="XML"/> <param name="dtmf-type" value="signal"/>  <param name="jitter-size" value="40,100"/>  <param name="codec-prefs" value="PCMU,PCMA"/>    <param name="gk-address" value="*"/>  <param name="gk-identifer" value=""/>  <param name="gk-interface" value="<My Public IP>"/>  </settings> <listeners> <listener name="default"> <param name="h323-ip" value="<My Public IP>"/> <param name="h323-port" value="1720"/> </listener> </listeners> </configuration> On netstat for 4569 (IAX) port, below is the result udp 0 0 0.0.0.0:4569 0.0.0.0:* 16533/freeswitch I have one DIDWW DID configured with IAX2 protocol and routed to this server. When i am dialing this DID, i am getting calls on server which i checked in ngrep. However, there is no event on freeswitch. Just for your information, I configured mod_opal on both latest 1.2 and 1.4 version and found same result for both. Can anyone please help me what i need to do additonally to get IAX2 calls on freeswitch? Do i need to configure anything for IAX2 ? Is there any this i can test to check it is configured and working properly ? Thanks & Regards, Bharat Lalcheta

2 1

Unable to get channel variables from directory to dialplan.
by Dave Horton 26 Aug '15

26 Aug '15

I think you may need to access it as ${variable_user_context}

1 0

Issue (possibly with stale nonce) when redirecting INVITE to another FS
by Dave Horton 26 Aug '15

26 Aug '15

I have a setup where a session border controller is distributing calls across multiple FS servers. Calls come from device —> SBC —> one of the FS servers, where they are challenged, and subsequently established. The problem I have is that in some cases (when I determine a device already has an active call on a different FS server) I respond to the INVITE with a redirect request so that the SBC will redirect the INVITE to the server with the already-active call for that device. The problem is that before the redirect is sent by FS #1 it first sends a 407 Proxy Authorization - that goes back to the device which responds with the appropriate credentials. My FS app then sends a 302 back to the intermediary SBC, which generates a new INVITE to FS #2. That INVITE contains the same Proxy-Authorization header that was used to authenticate on FS #1, but FS #2 rejects it thusly: 2015-08-25 11:02:43.059928 [DEBUG] sofia_reg.c:1335 Send challenge for [5125265790(a)10.124.48.104] 2015-08-25 11:02:43.059928 [WARNING] sofia_reg.c:1339 SIP auth challenge (INVITE) on sofia profile 'device' for [5125265790(a)10.124.48.104] from ip 10.124.48.171 The second 407 goes back to the phone — this time from FS #2 — and the phone gives up at this point, and the call dies. I actually wanted to bypass authentication on FS #2 entirely, so I redirected the call to a different profile (one used only for these special purpose redirections), and in that profile I set auth-calls=“false” and 'apply-inbound-acl’ to include the address of the SBC. It seems, though, that by virtue of the Proxy-Authorization header being included, FS #2 is still trying to authenticate. And then authentication fails, I believe because the nonce is designed to be a one-time thing (I see the sip_authentication.last_nc column in the table of nonces seems designed to declare the nonce stale if used more than once). This seems a relatively common use case (redirection via an intermediary SBC or proxy) so I figured there must be a solution here somewhere? Thanks in advance for any and all advice. (Note: I am using an older version of FS: FreeSWITCH Version 1.0.head (git-4192195 2011-08-16 19-39-06 -0400) Dave

1 1

De-bridging via ESL
by David McNab 26 Aug '15

26 Aug '15

Hi folks, Is there a way to de-bridge a session purely via ESL? For example, Alice, Bob and Charlie are in conference. I'm looking for a way for Charlie to depart the conference, ring Douglas, talk to him, then patch Charlie back into the conference, and bridge Douglas in as well. Cheers David

2 1

Change uri inside Proxy-Authorization (sip_auth_uri)
by Gonzalo Gasca Meza 26 Aug '15

26 Aug '15

In a SIP TLS call initial SIP INVITE with proxy authentication uri is populated with transport=tls, which seems to be causing issues with far end. How can I modify this field? I need to be (uri=" sip:+14089234300@tls.itsp.com) like this: Proxy-Authorization: Digest username="gonzalo", realm="sip.itsp.com", nonce="76202e080586a818939c7b1292e61317", cnonce="CoLCw8Y1EjOpXQqrj4sqmw", opaque="b4728d445b067bf647afd6d7028881f6", algorithm=MD5, uri=" sip:+14089234300@tls.itsp.com", response="d901a524745a6124c50e6dac1c08a502", qop=auth, nc=000000 Instead of sip:+14089234300@gonzalotls.itsp.com;transport=tls: Proxy-Authorization: Digest username="gonzalo", realm="sip.itsp.com", nonce="5c99ec833921f9e31eb50d27c6230b33", cnonce="0sNq8sY2EjOpXQqrj4sqmw", opaque="c85f5b8d3a7ac674837b72594bf3c7b8", algorithm=MD5, uri=" sip:+14089234300@tls.itsp.com;transport=tls", response="d17b81a6cbbb1538d5c9a03e35d6bdde", qop=auth, nc=00000001 This is my configuration: <include> <gateway name="itsp-tls"> <param name="tls-version" value="tlsv1"/> <param name="register-transport" value="tls"/> <param name="username" value="gonzalo*****"/> <param name="password" value="password****"/> <param name="realm" value="itsp-tls"/> <param name="proxy" value="tls.itsp.com"/>  <param name="register" value="false"/> </gateway> </include> <extension name="us_itsp_secure"> <action application="log" data="INFO Processing REGEX"/> <condition field="destination_number" expression="^80(\d{11})$"> <action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=false"/> <action application="export" data="zrtp_secure_media=false"/> <action application="export" data="nolocal:rtp_secure_media=mandatory:AES_CM_128_HMAC_SHA1_80"/> <action application="set" data="dialed_number=+$1"/> <action application="log" data="INFO Processing US call to ITSP Secure ${dialed_number}"/> <action application="bridge" data="sofia/gateway/itsp-tls/${dialed_number}"/> </condition> </extension> Thanks

1 0

RTP to SRTP Freeswitch 1.4
by Gonzalo Gasca Meza 26 Aug '15

26 Aug '15

I have the following scenario: SIP Phone (RTP/SIP UDP) ---> Freeswitch ---> SRTP/TLS --> ITSP Currently when I place a call Freeswitch is not sending SRTP SDES in SDP. It does the TLS connection but SDP contains only AVP codecs. I have used the rtp_secure_media set to true and also export but crypto caps are not attached. I can see them listed, but nothing happens. Logs: http://pastebin.com/DgZ1Lejp sofia_status external::itsp-tls gateway sip:gonzalo@tls.itsp.com:5061:5061 NOREG dialplan/public.xml <extension name="public_itsp_tls"> <condition field="destination_number" expression="^(80\d{11})$"> <action application="set" data="rtp_secure_media=true"/> <action application="export" data="rtp_secure_media=true"/> <action application="log" data="INFO Processing call to itsp"/> <action application="transfer" data="$1 XML default"/> </condition> </extension> dialplan/default.xml <extension name="US_ITSP_Secure"> <action application="log" data="INFO Processing REGEX"/> <condition field="destination_number" expression="^80(\d{11})$"> <action application="set" data="dialed_number=+$1"/> <action application="log" data="INFO Processing US call to ITSP Secure ${dialed_number}"/> <action application="bridge" data="sofia/gateway/itsp-tls/${dialed_number}"/> </condition> </extension> sip_profiles/external/sipsitspTrunk.xml <include> <gateway name="itsp-tls"> <param name="tls-version" value="tlsv1"/> <param name="register-transport" value="tls"/> <param name="contact-params" value="transport=tls"/> <param name="realm" value="itsp-tls"/> <param name="proxy" value="tls.itsp.com:5061"/>  <param name="register" value="false"/> </gateway> </include>

3 3

Can Stereo Placement in Conferences be bypassed
by Chris Mandra 26 Aug '15

26 Aug '15

Hi guys. Is there a way to turn stereo placement OFF for a single user user for a stereo conf call (eg. 3500, web-rtc) I don't want to turn stereo off - just the automatic placement heard by a single user. Thanks! chris --

1 0

RE- Gateway failover question - SIP INVITE still be sent to the failed gateway after failover within the dialplan takes place
by Andrew Keil 26 Aug '15

26 Aug '15

To FreeSWITCH Users, I wondered if someone could assist me regarding failover when a gateway goes down. FreeSWITCH version 1.4.20 (current production release). The requirement from my SIP Trunk provider is to failover in about 4 to 5 seconds. I have tried the following inside my dialplan: <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="call_timeout=4"/> <action application="bridge" data="sofia/gateway/SIPprovider1/xxxxxxxxxx;user=phone|sofia/gateway/ SIPprovider2/xxxxxxxxxx;user=phone"/> <action application="sleep" data="1000"/> <action application="bridge" data="sofia/gateway/SIPprovider2/xxxxxxxxxx;user=phone"/> Where xxxxxxxxxx is a valid phone number. SIPprovider1 would be down (ie. SBC switched off) and SIPprovider2 would be working OK. Obviously these are setup within the FreeSWITCH /conf/sip_profiles/external/myprovider.xml. I have experimented with other various combinations: <action application="set" data="call_timeout=4"/> <action application="bridge" data="sofia/gateway/SIPprovider1/+44xxxxxxxxxx;user=phone|sofia/gateway/ SIPprovider2/+44xxxxxxxxxx;user=phone"/> <action application="sleep" data="1000"/> <action application="bridge" data="sofia/gateway/SIPprovider2/+44xxxxxxxxxx;user=phone"/> or <action application="set" data="call_timeout=4"/> <action application="bridge" data="sofia/gateway/SIPprovider1/+44xxxxxxxxxx;user=phone|sofia/gateway/ SIPprovider2/+44xxxxxxxxxx;user=phone"/> or <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="call_timeout=4"/> <action application="bridge" data="sofia/gateway/SIPprovider1/+44xxxxxxxxxx;user=phone"/> <action application="sleep" data="1000"/> <action application="bridge" data="sofia/gateway/SIPprovider2/+44xxxxxxxxxx;user=phone"/> The issue I am having is the 4 to 5 second switch to SIPprovider2 does work, however for some reason the SIP INVITE messages are still being sent to SIPprovider1 after the 4 seconds. I need the SIP INVITE messages to STOP after 4 seconds to SIPprovider1 (which is down). Any ideas would be most appreciated. Kind Regards, Andrew Keil Visytel Pty Ltd

1 0

bridge unexpectedly disconnects after 5-6 sec
by Artur Mega 25 Aug '15

25 Aug '15

i set up freeswitch, and almost all configs are default (i made some little changes in directory and dialplan). I can call anywhere, for example to my cell phone, but call longs approximetely only 5 sec... In siptrace i found nothing. No timeouts like call_timeout or leg_timeout. Also i thought maybe the reason in the gateway registerations... Server has private (static) ip. log is here: http://pastebin.com/mLrkGz3c -- Arthur

2 1