July 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

WSS Certificate issue
by Oleg Stolyar 28 Jul '14

28 Jul '14

Guys, I've run into a strange problem. I cannot establish a wss connection to my FreeSWITCH from my Android Chrome or Android Firefox. I believe Firefox on a Mac is also affected although it works fine on both Chrome and Firefox on Windows. The error is "The site's security certificate is not trusted" I thought that I did not set up the cert correctly but it turns out that the FS own WebRTC demo has the same problem. If you go to this URL https://webrtc.freeswitch.org:7443 from your Android Chrome browser, you will see it. I consulted with my security team and the best we could guess was that the FS code does not check or use the cert chain from ca-bundle.crt and some browsers are not smart enough to figure it out on their own. Any ideas?

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Need help to setup verto web page
by Neo Haux 27 Jul '14

27 Jul '14

Hi all, I want to use the verto demo page in /usr/src/freeswitch/html5/verto/, but I don't know where to setup the SIP/account/password/FS's IP I tried chaging this in verto.js: function init() { cur_call = null; pop("#ext", "verto_demo_ext", "3500"); pop("#name", "verto_demo_name", "FreeSWITCH User"); pop("#cid", "verto_demo_cid", "1008"); pop("#textto", "verto_demo_textto", "1000"); pop("#login", "verto_demo_login", "1003"); pop("#passwd", "verto_demo_passwd", "1003!@#$"); pop("#hostName", "verto_demo_hostname", 192.168.1.106); pop("#wsURL", "verto_demo_wsurl", "wss://" + 192.168.1.106 + ":8082"); But doesn't work. I get the verto main page with empty fileds in Login, Password, CID name, CID number Hostname, Websocket URL Even after filling them manually and clicking on the button "Log In" don't trigger any action. I can see that the port 8082 is open by FS after following the instruction on confluence's verto page. Any help is welcome ! Thanks

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Outbound Call Recording Delay Problem
by Amin Dhamani Askhanuman 26 Jul '14

26 Jul '14

Hi, We are using Freeswitch. Currently we are facing Voice Recording Delay issues in some files We already tested recorded files in .mp3 and .wav but delays in both. Any Solution? -- Best regards, Amin Dhamani Amin Dhamani | System Administrator | +66 911167159 | Krits Tower, 1032/1-5, 14 Rama 4, Bangkok 10120 *www.askhanuman.co.th <http://www.askhanuman.co.th/>* | www.facebook.com/AskHanuman | amin(a)askhanuman.com This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. ______________________________________________________________________________ -- This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. ______________________________________________________________________________

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Routing/NAT issues with head?
by sangdrax8 25 Jul '14

25 Jul '14

I am trying to get the current git pull to work using the default configurations that are provided. I am having issues that seem related to routing, or possibly NAT. Let me try and explain my issue, and see if someone knows there is a setting I am missing or if this is actually some strange bug. Basic setup: I am using the default configuration provided, only changing the default password for clients. My test machine has 2 interfaces, eth0 on the internet (no NAT), and eth1 on an internal RFC address. With the default configuration, the "internal" profile is on the internet address from eth0 on port 5060. The "external" profile is on the same eth0 ip, on port 5080. Problem: So everything looks normal, and I can register two soft clients (Groundwire, one android one iPhone) to the internal profile. When I place a call between 1000 and 1001, the SIP packets all work exactly as expected, but I get no audio. Using these same to clients on an older build (still 1.5.x) in my production environment works just fine. So I would think the clients settings are fairly standard. Looking at the server, I found that the freeswitch is sending the audio packets to the RFC address of my clients (192.168.x.x) and is deciding to send these packets out my eth1 interface. Now I do route all RFC's out eth1, so this test server can be connected to from the inside. It doesn't have an interface in the same subnet as the clients though, just a route for all RFC's internally. Testing a theory, I removed the route to the 192.168.x.x range which the phones happen to be in (wifi network gets NAT to the internet when connecting to this machine as most clients probably do). Once I do this, and with no changes to my freeswitch, the audio works. With out this route on the base server, freeswitch sends the UDP packets out eth0, back to the correct Internet IP my clients connected from. I thought maybe this had something to do with nat.auto, or localnet.auto, but using the acl cli command, it would seem these things all check out correctly. When I ask it if the 192.168 net is in the nat.auto it returns true, so it would see to know it SHOULD be natting for that network. The only false I get is for the subnet that my eth0 is in. If, by chance, I had been testing with a range that my server didn't have a route to (if I didn't route all RFC's internally) I probably wouldn't have seen this.

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FreeSWITCH Friday BugHunt Reminder!
by Ken Rice 25 Jul '14

25 Jul '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday BugHunt Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken http://ift.tt/vGSsGm http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Sched broadcast issues
by Chris Martineau 25 Jul '14

25 Jul '14

Hi, I have just recently started getting issues with sched_broadcast. I have not changed versions and as far as I am aware nothing else has changed. The problem is also intermittent whereby the broadcast kicks in very late. We are setup so that on connection a jingle is played to show that you are connected in case early media takes a while to arrive. The jingle is played on a sched_broadcast 1 second after connection to allow the media paths to be established. Here is the command... <action application='sched_broadcast' data="+1 playback::/path/to/jingle.wav"/> Basically, occasionally you get ringing and then 5-10 seconds later you get the jingle even though it should trigger after 1 second? The version we have been on for a while is FreeSWITCH version: 1.2.23+git~20140508T161622Z~df733a7fde~64bit (git df733a7 2014-05-08 16:16:22Z 64bit) Has anyone come across this issue? Regards Chris

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Testing mod_verto, a few ideas
by Antonio Silva 25 Jul '14

25 Jul '14

Hi,. i was testing mod_verto. I love this new endpoint. :) During the tests, i saw a few details that i think could be nice to have comparing to the other endpoints. - dbname - we can't force the path where to store json.db for presence, right now it creats a json.db in the directory, freeswitch/db, This db is used for presence, no? - profile restart / relaod configuration without dropping the active calls, right now the command i use is reload mod_verto which drops all the active calls - implement new option to disable login from websocket, like in sofia "sip-forbid-register", i solved by setting a random jsonrpc-password - cannot force caller id from directory with variables, i set the variable effective_caller_id_name and effective_caller_id_number in directory.xml but they are not used... possible other variables could be nice. - missing internal events, like in sofia, for example, sofia:register, quite useful to monitor the registered verto endpoints with esl. - without setting the parameter mcast_ip, sometimes i have the error repeated multiple times: 2014-07-24 16:59:25.973074 [ERR] mod_verto.c:3247 MCAST INVALID READ -1 Since this is a resume of my testing i didn't report them on jira, If you wish i can open separated jira's to each point. Thanks, António

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Ricoh MP C2800 Fax SIP Registration
by Sean Devoy 25 Jul '14

25 Jul '14

Hi, I have a Ricoh MP C2800 VOIP Fax that has VOIP capabilities. Unfortunately, the user interface is quite limited and it fails to register to FS. The SIP reply to the initial AUTH with the NONCE never gets back to the FAX. There is nowhere to specify external IP and PORT options so I have limited room to work on things on the FAX end. Does anyone have one of these working behind NAT? Any tips? I have tried NDLB options, but welcome suggestions. Thanks, Sean

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feature idea - "system" api call, but with call data passed in?
by Nathan Neulinger 25 Jul '14

25 Jul '14

Would be nice to have an equivalent of the 'system' command, but which would pass (as json) the current call data - such as would be output by 'info' application - on stdin. (bg_system as well) -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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MultiNAT
by Kurtis Heimerl 25 Jul '14

25 Jul '14

Hey Users, I have an interesting NAT setup. I'm running FS on the inside of our network as a router/proxy between some SIP phones and DID providers. However, each DID provider is behind a *different* NAT (a property of our VPN setups for them). For instance: DID1 is at IP 192.168.1.1 and DID2 is at 10.0.0.1. I have calls working for each of them when I set the following in my external profile: <param name="ext-rtp-ip" value="10.0.0.2"/> <param name="ext-sip-ip" value="10.0.0.2"/> However, I need to dynamically route between *both* of them. I need a mechanism for setting ext-rtp-ip and ext-sip-ip in the dialplan itself! Is there a set way to do this? Thanks!

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