August 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

Wiki editing account
by David McNab 25 Aug '15

25 Aug '15

Hi guys, How do I go about applying for an editing account on the wiki? In my learning process, I've been figuring out some things about Python scripting which might be able to help others who are Python-inclined. I'm asking if it might be possible to get edit access on the wiki, so I can publish some walk-through examples, together with practical tips and caveats. Cheers David

2 1