August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

stop unwanted registration attempts
by Jungle Boogie 03 Aug '16

03 Aug '16

Hi All, How do people stop bad registration attempts to freeswitch? Is it pretty much impossible so don't worry about it as long as you have fail2ban? Using sngrep, I see lots of registration attempts like this: My actual IP has been replaced with 1.2.3.4. 2016/08/02 21:35:33.397073 195.154.48.130:5080 -> 192.168.0.137:5060 REGISTER sip:1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 195.154.48.130:5080;branch=z9hG4bK23552ce85a146013577b3912;rport From: "7612" <sip:7612@1.2.3.4:5060>;tag=23552ce8ba27 To: "7612" <sip:7612@1.2.3.4:5060> Call-ID: ce85a14-4c0e6013-577b3912(a)1.2.3.4 CSeq: 1 REGISTER Contact: "7612" <sip:7612@195.154.48.130:5080> User-Agent: VaxSIPUserAgent/3.1 Expires: 1800 Max-Forwards: 70 Content-Length: 0 I have these iptables rules: -P INPUT ACCEPT -P FORWARD ACCEPT -P OUTPUT ACCEPT -N f2b-freeswitch -A INPUT -j f2b-freeswitch -A INPUT -p tcp -m string --string "VaxSIPUserAgent/3.1" --algo bm --to 65535 -j DROP -A INPUT -p udp -m string --string "VaxSIPUserAgent/3.1" --algo bm --to 65535 -j DROP -A INPUT -p udp -m udp --dport 5080 -m string --string "sipcli" --algo bm --to 65535 -j DROP -A INPUT -p udp -m udp --dport 5080 -m string --string "friendly-scanner" --algo bm --to 65535 -j DROP -A INPUT -p udp -m udp --dport 5080 -m string --string "VaxSIPUserAgent" --algo bm --to 65535 -j DROP -A INPUT -p udp -m udp --dport 5060 -m string --string "sipcli" --algo bm --to 65535 -j DROP -A INPUT -p udp -m udp --dport 5060 -m string --string "friendly-scanner" --algo bm --to 65535 -j DROP -A INPUT -p udp -m udp --dport 5060 -m string --string "VaxSIPUserAgent" --algo bm --to 65535 -j DROP -A INPUT -j f2b-freeswitch -A f2b-freeswitch -j RETURN Are my rules not stopping this registration because it's not being recorded in any logs? How do you stop (or prevent) unwanted registration attempts, even if it's a sip scanner? Thanks!

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httapi - dial/bridge only working when it comes first
by Don Hawkins 03 Aug '16

03 Aug '16

I'm using httapi and everything is working well for the most part. The one thing I noticed is that whenever I do anything before trying to "Dial" or bridge a call to the gateway: 1. The first thing never happens, even though the console says it did. 2. The outbound call "Dial" won't work, it hangs for about 60 seconds and then returns a NORMAL_TEMPORARY_FAILURE every time. Things to note: - If I start with the dial request (B below) then it connects the call fine, it just can't come after anything else. - If I never return a request to dial (pretend B below starts a conference instead) then everything works fine, the TTS works before and the conference works after. Here is an example: A. When FS makes it's first request to my URL I return this: <document type="text/freeswitch-httapi"> <work> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabl…" name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document> B: When FS makes it's next request to my action URL (same URL with different parameters) I return the dial command and attempt to connect to the outbound gateway. <document type="text/freeswitch-httapi"> <work>  <execute application="set" data="ringback=http://blabla.com/beat.wav "></execute>  <execute application="set" data="effective_caller_id_name=My Name"></execute> <execute application="set" data="effective_caller_id_number=12144635555"></execute>  <execute application="export" data="nolocal:codec_string=G711"></execute> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabl… " application="bridge" data="sofia/gatewayAlcazar/12144635555"> <execute application="answer"/> </execute> </work> </document> What happens? It never speaks the text (although the console says it did) and when dialing out I get "NORMAL_TEMPORARY_FAILURE" and the call never connects. Keep in mind, if the second thing I return (B above) is NOT dial then everything works fine, it will speak the text, play a file, start a conference, or any other application. This is the weirdest thing to me and of course almost everything is useless if you can't connect a call at some point, if anyone has any advice I'd be highly appreciative.

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group dial disconnects before VM
by Jungle Boogie 03 Aug '16

03 Aug '16

Hi All, I have some end points in a ring group and when there's no answer, I'd like it transferred to a voicemail box. Pretty simple. What's happening is I hear, "The person at extenion--goodbye. conf/dialplan/default.xml for the group: <extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="call_timeout=20"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="group/support@${domain_name}"/> <action application="answer"/> <action application="sleep" data="2000"/> <action application="voicemail" data="default 2600"/> </condition> </extension> Logs: https://pastebin.freeswitch.org/view/b3856e93 What am I doing incorrectly? Thanks!

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Determine When Bridge Attempt is Complete
by Colin Morelli 03 Aug '16

03 Aug '16

Hey, I'm looking to determine when a bridge request is complete. I'm using mod_amqp to receive events from Freeswitch and using that data to update a separate database. The problem is that, so far, I've only been able to determine when a bridge attempt fails in a channel_execute_complete event. I'm really trying to avoid sending that event over mod_amqp because there are *a ton* of those events generated for every call. Is there any other way to determine when a bridge attempt fails? We've got CHANNEL_BRIDGE & CHANNEL_ANSWER to determine when it's successful - would be great if anyone knows for the other case. Best, Colin

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Calls without registration
by Paul Zillmann 03 Aug '16

03 Aug '16

Hello, due to an open firewall some robots could get into my network and start some calls (11.000!) over my PSTN gateway. I looked through the data and those robots didn't registered themselve, they just placed the call on the internal profile and succesfully got through the dialplan. How is it possible that they can make calls on the internal profile (Port 5060) without registration and is there a variable for sofia to don't allow those calls. Is there anything else I can do appart from checking the source IP Address in the dialplan? Thanks, Paul

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fusionpbx error
by uzair iltifat 03 Aug '16

03 Aug '16

Hi, I have recently installed fusionPBX, when i try to access it with the URL i get this error ----- SQLSTATE[28000] [1045] Access denied for user 'fusionpbx'@'localhost' (using password: YES) ----- waiting for the fastest response, thanks! http://uzair.ahmed.netcv.com/http://uzair.ahmed.netcv.com/ Uzair Ahmed [image: http://]about.me/uzairIltifat <http://about.me/uzairIltifat?promo=email_sig> <http://about.me/uzairIltifat?promo=email_sig> http://uzair.ahmed.netcv.com/

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Recording with a beep please
by Don Hawkins 03 Aug '16

03 Aug '16

Using httapi, the code below plays the beep BUT will not begin the recording until I press the terminator key, * in this case. How can I make it start recording right away and have * terminate the recording? <document type="text/freeswitch-httapi"> <work> <speak name="speak" loop="1" engine="flite" voice="kal" text="Please leave a message after the beep." language="en"></speak> <execute application="playback" data="tone_stream://%(100,15000,800);loops=-1"/> <execute action="redacted" application="record" data="/usr/share/freeswitch/htdocs/recordings/z0jnu8h6-898958.wav 25 20" data="RECORD_STEREO=true" data="playback_terminators=*"> </execute> </work> </document>

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Implementing One Way audio through Freeswitch(Default Mode)
by Saurabh Verma 03 Aug '16

03 Aug '16

Hi I want to implement One Way Audio through Freeswitch in Default mode. *SoftClient1 ---------> FREESWITCH ----------> SoftClient2* *a=sendonly (Default Mode) * For this i have hardcoded the media attribute for audio as sendonly in Softclient1.Everytime Softclient1 will initiate the call to softclient2.Now when i make a call,Softclient1 sends *sendonly as media attribute in SDP of INVITE packet*.Freeswitch replies with recvonly in the 200OK.When Freeswitch forwards the Invite to Softclient2 ,*we can't see any media attribute* parameter in it and Softclient2 replies to FREESWITCH with *sendrecv in 200OK *and hence Both way media is flowing instead of one way. Generally if a SIP client receives sendonly in Invite, it will send recvonly in 200OK by default.But here Freeswitch is modifying it and not forwarding sendonly to SoftClient2. Why Freeswitch is not forwarding the sendonly in invite to SoftClient2? How can we implement this using Default mode? Can somebody help me with this? -- *With Warm Regards:* *Saurabh Kumar Verma*

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issues with buidling 32bit Debian repo
by Peter 03 Aug '16

03 Aug '16

I am still facing difficulties with creating the 32bit FreeSWITCH packages for Debian 8: I have been building the packages using Stanislav's script on two 32 bit machines, one with 512MiB RAM and the other 2GiB RAM. On the 512MiB machine I had to modify the invoked |./debian/rules script and lower the xs compression level in order to avoid memory problems.| |On the 2GiB I could run the build script unmodified.| | | |I published the repo to my vps, but whichever of both builds I publish, if I do a | |apt-get update, I get stuck at the same error:| |"Failed to fetch gzip:/var/lib/apt/lists/partial/xxxx.com_debian_freeswitch-unstable-i386_dists_jessie_main_i18n_Translation-en%5fUS.gz Invalid file format" | |So, what could be wrong with the gzipped Translation-en package? | |%5f underscore in filename ?| | | All I would like is to upgrade my small FreeSWITCH 1.2 system, running happily for many years on a 32bit platform (now Debian 8) Moving to a 64bit platform is currently not an option for me.

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[Weird FreeSwitch Duplicate UUID Error]
by Bilal Abbasi 02 Aug '16

02 Aug '16

Hi Users, I am getting some weird error on Freeswitch, that is coming on calling random extensions, I have a dialplan for 3 digit extensions , so when i can 194 it works perfect and when i call 197 it gaves me below error, Duplicate UUID! unable to create channel. What i have in my dial plan is, <action application="set" data="b-leg-uuid=${create_uuid()}"/> <action application="bridge" data="{origination_uuid=${b-leg-uuid},api_on_answer='uuid_audio ${b-leg-uuid} start write mute -4'}user/${destination_number}@ ${domain_name}"/> Can someone please guide me what i am doing wrong in this, P.S: some times calling 197 works perfect and it comes on 194 WEIRD. Regards Bilal Abbasi

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