February 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

AUDIO RTP REPORTS ERROR: [Bind Error! on Multipple profile
by Steve Kieu 23 Feb '15

23 Feb '15

Hi everyone, My setup is that I have * internal.xml on 192.168.0.2 * vpn.xml on 10.8.0.1 * external.xml Freeswitch behind NAT router ; wlan0 has 192.168.0.2 and 10.8.0.1 is tun0 which is the openvpn internface. Default gateway is wlan0. Thing like normal using internal.xml But for call from A registered in vpn profile I always got 2015-02-20 17:41:44.591664 [ERR] sofia_glue.c:3989 AUDIO RTP REPORTS ERROR: [Bind Error! XXX:23324] XXX is my external internet IP. The vpn.xml I set: <param name="rtp-ip" value="$${external_rtp_ip}"/>  <param name="sip-ip" value="10.8.0.1"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="ext-rtp-ip" value="auto-nat"/> no matter what I set in ext-rtp-ip (did replace auto-nat with my external IP) it still gives the errors. I want for the vpn profile, just use the vpn IP for sip for the endpoint register with this, but using correct sip rtp when sending out to endpoint registered in other profile - basically making call between profile working - Can freeswitch support that and how ? I did set all IP to use vpn address only, and call making between same profile (vpn) works. But not to other profile and not to external gateway. Same problem. Thanks a lot in advance. -- Steve Kieu

3 5

Limit max concurrent calls on gateways in lua
by Zoltán Szabó 22 Feb '15

22 Feb '15

Hi, I have a lua script where I do some business logic. After the logic I call a number on a gateway, then when the called party picked up, I call an other number on a gateway and bridge the two calls. The only problem is that I have 5 gateways and each can have maximum of 3 calls (so I can have 15 calls on this system). Here is what I have at the moment (simplified version): dial_call_1 = "{leg=1}sofia/gateway/gw-1/0035312345678"; dial_call_2 = "{leg=2}sofia/gateway/gw-2/0035387654321"; session1 = freeswitch.Session(dial_call_1); if (session1:ready()) then session2 = freeswitch.Session(dial_call_2, session1); if (session2:ready()) then freeswitch.bridge(session1, session2); if (session2:ready()) then session2:hangup(); end end if (session1:ready()) then session1:hangup(); end end How can I use the limit application here to set the max concurrent calls on my five gateway, each with 3 max calls and distribute outbound calls between them? Many thanks, Zoltan

2 5

Where to go from here?
by T Fred Farmington 22 Feb '15

22 Feb '15

Pardon my naivety, but I am fairly new to FreeSWITCH and, despite having read the FreeSWITCH 1.2 book and doing MANY web searches, I cannot seem to find the correct combination of settings to get things working. Currently I have an 'outside' SIP line number pointed to my FreeSWITCH server. When I call that number I can see its arrival via FS_CLI console and it shows in the freeswitch.log file (I can include those log file lines if it would help but it is 106 lines - I don't want to 'swamp' the posting thread) However when I attempt to establish the call the line consistently goes Busy and will not establish a connection. I have tried various recommended settings for: * ..\conf\autoload_configs\acl.conf.xml * ..\conf\sip_profiles\external.xml (and its ipv6 version) * ..\conf\sip_profiles\internal.xml (and its ipv6 version) * ..\conf\vars.xml Each time I try something, it seems like everything else quits working. So, instead of trying to 'guess' which of the configuration changes made things worse, I Uninstall FreeSWITCH and then Re-Install it. After that, my in-house, internal Softphones (Phoner, XTalk, & Voiper) all work just fine to the FreeSWITCH default extensions. But the external SIP call does not work. I'd seriously like to get beyond these initial configuration problems and begin working on actually configuring how the overall phone system is intended to operate. Where/How do I get things working? Any advice/suggestions you can offer would be greatly appreciated. Thanks TF

3 2

piped failover doesn't work in dialplan, but works from console
by Mi Ke 20 Feb '15

20 Feb '15

4 15

sip_route_uri with simplify ( uuid_simplify & auto simplify )
by jay binks 20 Feb '15

20 Feb '15

Im bringing up a call ( "Freeswitch node A" ) where im using sip_route_uri to send the call via a specific node in my network "Freeswitch node B". but I also want to get this "Freeswitch Node A" out of the call path after the call is answerd. "simplify" works great for this. however when I try and simplify a call, that was brought up with a sip_route_uri my simplify fails ! :( The Refer-To: sent by "FS node A" contains the SIP URI for my call , but has no reference to the details used in sip_route_uri meaning nothing was sent to "Freeswitch node B" is this a bug, or am I missing something in the use case of these 2 features ?? -- Sincerely Jay

3 9

rfc2833 problems
by Jason Holden 20 Feb '15

20 Feb '15

Running fs 1.2.11 Experiencing problems with DTMF rfc2833. Packet capture shows in the internal profile all the DTMF but on the external profile 3 quarters of the DTMF is missing. Any recommendations how to address. Experiencing the problem with a Grandstream analog gateway.

2 1

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 20 Feb '15

20 Feb '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

1 0

Can you queue external number?
by Shai Perelman 20 Feb '15

20 Feb '15

HI, in our old system there is a queue to an internal extension say 199 and 199 has folow up to an external number say 972544933444 I was told that in freeswitch this is not possible/// is it possible in some way to queue an external number in fs? thanks Shai

3 3

B leg ringing when Caller hangup before answer
by Charles Wang 19 Feb '15

19 Feb '15

Hi all, I have a server with freeswitch 1.4.15 + freetdm(FXS/FXO). I think there is a bug in inbound call via the freetdm FXO device. When I try to make call from FXO and it bridges to SIP device named 1234 via the following dialplan. <extension name="my call test"> <condition field="destination_number" expression="^(1234)$"> <action application="set" data="call_timeout=30"/> <action application="bridge" data="user/1234"/> </condition> </extension> Before SIP 1234 answers the call, the caller (FXO) hangup call before 1234 answered. But the callee (SIP 1234) is still ringing and stop ring after about 30 seconds. It is the same condition if the callee is FXS device. I can find two channels during ringing (A leg & B leg). After the caller(FXO) onhook, the A leg is still alive and A leg will not be hangup before the callee (FXS or SIP) stop ring ( call-timeout ). There are two legs (A leg & B leg) after the caller FXO had hangup. uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context 53ce4fe9-9511-4625-87ae-1448421c9810,inbound,2015-02-12 17:33:11,1423733591,FreeTDM/2:2/1234,CS_EXECUTE,unknown,unknown,,1234,bridge,freetdm/FXS1/1,XML,TEST,PCMU,8000,64000,PCMU,8000,64000,,charles,,,RINGING,,,,,,,unknown,unknown,,1234,XML,default cf5a7ee4-cbfd-48e4-ab4e-4d757216712c,outbound,2015-02-12 17:33:12,1423733592,FreeTDM/1:1/,CS_CONSUME_MEDIA,unknown,unknown,,1,,,XML,default,,,,,,,,charles,,,RINGING,Outbound Call,1,,53ce4fe9-9511-4625-87ae-1448421c9810,,,unknown,unknown,,1,XML,default Can anyone help me to solve it or tell me why? -- Best Regards Charles

4 11

How to remove caller_id_name?
by Sergey Safarov 19 Feb '15

19 Feb '15

I receive INVITE with bad caller_id_name. How I can remove bad header? Bad INVITE example INVITE sip:2496@10.21.7.30 SIP/2.0 Via: SIP/2.0/UDP 10.21.7.30;rport;branch=z9hG4bKcX1B77t32Z5pe Max-Forwards: 69 From: "#127.024.031" <sip:2036@company.com>;tag=BmHamt2NK4y0r

3 9