- FreeSWITCH-users - FreeSWITCH Mailing Lists

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.

1 1

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

1 0

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

1 0

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

1 1

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

2 1

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

2 2

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!

6 19

Early Media & voicemail
by marino.mileti＠alice.it 24 Jul '14

24 Jul '14

Hi all, i'd like to setup a call timeout in order that calls not answered will redirect to voicemail. But in my scenario all calls are in "early_media" mode so i've noticed that all timeout not working (leg timeout, call timeout etc). I've found a bridge_answer_timeout; it works but after timeout call is killed but next line of my dialplan after "bridge" isn't executed.Any idea? Is it possible?

2 1

strange behaviour using group_confirm and originate_timeout
by Grant Bagdasarian 24 Jul '14

24 Jul '14

Hello, I'm using the enterprise originate to ring multiple destinations after the a leg has answered. These are the global parameters: <ignore_early_media=true,monitor_early_media_fail=user_busy:2:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true,originate_timeout=15> Notice the originate_timeout has been set to 15. When one of the ringing destinations answers the call, group_confirm is used to have them complete a script in order to get connected. session.execute("export", "group_confirm_key=exec"); session.execute("export", "group_confirm_file=javascript /usr/src/freeswitch/conf/scripts/confirm.js); When the confirm script is being executed, a file is being played and some digits need to be collected. Using play_and_get_digits. Audio file is about 12 seconds. But as soon as the 15 seconds timeout is hit which was set before, the destination is disconnected. What do I need to set in order for freeswitch to know the destination has answered, so it leaves that answered destination alone and does not enforce the originate_timeout? I'd expect the originate_timeout to be the time from starting to ring a destination till it has answered. Not continue enforcing it once the destination has been answered. Then again, group_confirm is required to be completed in order for freeswitch to mark the call as accepted, so this behavior might make sense. It comes down to the destinations being called not having enough time to complete the script because the originate_timeout is set to 15 seconds. Changing it to 25 gives them more time, but 15 seconds is the requirement for this application. Any ideas? because I'm lost.. Grant

3 4

mod_erlang_event
by Joel White 24 Jul '14

24 Jul '14

How do I get freeswitch to work with erlang when I installed erlang from source. When I tried to compile I got an error related to the module compilation.

2 1