- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] NAT Confusion
by Bob Smith 10 Jan '12

10 Jan '12

Because by all accounts, judging by the Wiki Entry, usign a public server is not really a wise thing to do in production .... the " http://dumbme.mbit.com.au/trixbox/stun_servers.htm" link on teh Wiki is broken, and the Wiki also says "Note: stun.fwd.org is gone; also stun.freeswitch.org is never guaranteed to be up and running so use it in production at your own risk.".... so I can only assume that most public servers are similarly provided on a "best endeavours" basis and prone to disappear at any time. Further, I now have an IPsec VPN between me and the firewall, hence eliminating NAT concerns, and I am still seeing the same issues as originally described with things partially working (even with Any/Any rules applied to firewalls). >Why not pick up a public stun server and config it in your softphone, most >likely it will address your issue.

2 1

"expire" for outgoing registrations not handled correctly?
by Roland Hänel 10 Jan '12

10 Jan '12

Hi, It seems that I'm running into a timing issue with outgoing (gateway) registrations, which results in some milliseconds in that I can't send calls over that gateway. I have configured a gateway like this: <gateway name="duro"> <param name="username" value="XXXXXXX"/> <param name="from-domain" value="durosip.qsc.de"/> <param name="password" value="XXXXXXXXXXX"/> <param name="proxy" value="durosip.qsc.de"/> <param name="expire-seconds" value="120"/> <param name="register" value="true"/>^M <param name="register-transport" value="udp"/> <param name="retry-seconds" value="5"/> <param name="caller-id-in-from" value="true"/> </gateway>^M If I look at the "200 OK" messages for the SIP REGISTER packet, the server sends me an "Expires: 30", so FreeSwitch should re-REGISTER every 30 seconds. However, that's what happening: 2012-01-09 16:37:00.804865 [NOTICE] sofia_reg.c:407 Registering duro 2012-01-09 16:37:31.804880 [NOTICE] sofia_reg.c:407 Registering duro 2012-01-09 16:38:03.804894 [NOTICE] sofia_reg.c:407 Registering duro 2012-01-09 16:38:35.824877 [NOTICE] sofia_reg.c:407 Registering duro *2012-01-09 16:39:06.825090 [NOTICE] sofia_reg.c:407 Registering duro* 2012-01-09 16:39:39.824881 [NOTICE] sofia_reg.c:407 Registering duro You can see that it's *approximately* 30 seconds betwee registrations, but in fact every time it's just a little more than 30 secs. This alone makes me feel a little bit strange, since my registration might have been expired during this time window (and consequently, maybe I'm loosing incoming calls because of that). Even for outgoing calls, this seems to be an issue, let's take a look at the 16:39:06.825090 re-REGISTER. We can see that at 16:39:05 (no micro-seconds here) the gateway becomes "UNREGED": Content-Length: 542 Content-Type: text/event-plain Event-Subclass: sofia::gateway_state Event-Name: CUSTOM Core-UUID: 243269fe-d280-4f83-aca1-abb88f2ad258 FreeSWITCH-Hostname: lap597 FreeSWITCH-Switchname: lap597 FreeSWITCH-IPv4: 192.168.232.164 FreeSWITCH-IPv6: ::1 *Event-Date-Local: 2012-01-09 16:39:05* Event-Date-GMT: Mon, 09 Jan 2012 15:39:05 GMT Event-Date-Timestamp: 1326123545824882 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_fire_custom_gateway_state_event Event-Calling-Line-Number: 151 Gateway: duro State: UNREGED Ping-Status: UP But the new REGISTER is done a full second later (this is not just a display issue, I saw those messages scrolling by and there was just a second pause between them): Event-Subclass: sofia::gateway_state Event-Name: CUSTOM Core-UUID: 243269fe-d280-4f83-aca1-abb88f2ad258 FreeSWITCH-Hostname: lap597 FreeSWITCH-Switchname: lap597 FreeSWITCH-IPv4: 192.168.232.164 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2012-01-09 16:39:06 *Event-Date-GMT: Mon, 09 Jan 2012 15:39:06 GMT* Event-Date-Timestamp: 1326123546825090 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_fire_custom_gateway_state_event Event-Calling-Line-Number: 151 Gateway: duro State: TRYING Ping-Status: DOWN Then, the gateway gets re-registered in a couple of milliseconds and everything is fine again. *It seems that this behaviour effectively creates a timespan of 1-2 seconds, where the gateway is not available for outgoing (and maybe also incoming) calls.* Does anyone of you know this issue? I wonder because this looks just to easy to be wrong in this fashion? Greetings, Roland

2 3

running freeswitch binary on more than one computer
by Arnuld Uttre (Phonologies) 10 Jan '12

10 Jan '12

I have several computers and all of them have exactly same OS, CentOS 5.5 . I built freeswitch on one computer with some --prefix=/home/arnuld/foo . After "make install" was done, I copied the installed directory to other machine. When I try to run it says "cwitch_xml.c couldn't find /home/arnuld/foo" Is it not feasbile to move the binary around ? Why freeswitch hardcodes the --prefix into binary ? -- Arnuld Uttre Systems Software Engineer arnuld(a)Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992

2 1

Xfer Call Back
by Carlos Cesario 10 Jan '12

10 Jan '12

Hello, In Att Xfer exists any method to call return to original call if called dont attend the call ? Eg: A transfer to B and hold phone, if B dont attend call the call back to caller . Grets Carlos

1 1

ignoring 180 and 183 w/o sdp
by Manjiri Deshpande 10 Jan '12

10 Jan '12

Hi, To avoid unnecessary ring back,I want to avoid 180 and 183 w/o SDP from a particular m/c address.\ I tried using ignore_early_media and sip_ignore_183nosdp.But still FS is not ignoring 180 and forwarding 180 to inbound leg. What happens is sip_ignore_183nosdp is ignoring 183,but sending 180 instead of this. Please help me to find out any solution for this. Manjiri

1 0

FreeSWITCH hangs at start up
by rajkumar k 10 Jan '12

10 Jan '12

I installed freeswitch and tried to start, but it hangs while running check_ip task. Does anyone know about this? # freeswitch 2012-01-10 10:26:31.984000 [INFO] switch_event.c:631 Activate Eventing Engine. 2012-01-10 10:26:32.060000 [DEBUG] switch_event.c:610 Create event dispatch thread 0 2012-01-10 10:26:47.768000 [INFO] switch_nat.c:419 Scanning for NAT 2012-01-10 10:26:47.772000 [DEBUG] switch_nat.c:169 Checking for PMP 1/5 2012-01-10 10:26:47.780000 [ERR] switch_nat.c:200 Error checking for PMP [general error] 2012-01-10 10:26:47.780000 [DEBUG] switch_nat.c:424 Checking for UPnP 2012-01-10 10:26:59.796000 [INFO] switch_nat.c:440 No PMP or UPnP NAT devices detected! 2012-01-10 10:26:59.912000 [INFO] switch_core_sqldb.c:1898 Opening DB 2012-01-10 10:27:01.848000 [NOTICE] switch_scheduler.c:166 Starting task thread 2012-01-10 10:27:01.884000 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1326171421 2012-01-10 10:27:01.884000 [DEBUG] switch_scheduler.c:214 Added task 2 check_ip (core) to run at 1326171421 -- View this message in context: http://old.nabble.com/FreeSWITCH-hangs-at-start-up-tp33108439p33108439.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

1 0

Overlap dialing problem
by Štefan Čudai 10 Jan '12

10 Jan '12

Hi, does FreeSwitch support for overlap dialing or did somebody try to use it? I am using Sangoma card A102 under Windows 7 with FreeTDM. I have configured my freetdm.conf.xml with this two lines: <param name="overlap" value="yes"/> <param name="min_digits" value="0"/> But when I'm dialing number (which has 14 digits) I hear disconnect tone after dialing first 5 digits. I think that reason is because FreeTDM module is waiting number with minimum digits equals to "0" (see my min_digits). I have also try set min_digits to 20 and then dial same number with 14 digits but result was same. So, I looked to log file and found that FreeTDM module got in thist case STATUS CONFIRM message but he was still waiting for next digits and his time expired. Can you help me please? Thanks a lot. Stefan

3 2

Calling multiple phones at once
by Ryan How 10 Jan '12

10 Jan '12

Hi, How do I set it up in freeswitch to call mulitple phones at once (internal and via a gateway) that works with call_timeout. I can't seem to find the appropriate wiki page. Call groups kinda work, but then the phones need to register. My setup with bridge the timeout doesn't work <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="user/1000(a)$${domain},sofia/gateway/sip.pennytel.com/0447123456"/> <action application="answer"/> <action application="bridge" data="loopback/app=voicemail:default $${domain} 1000"/> <action application="hangup"/> After 30 seconds all the phones start ringing again. And when the external number hangs up it immediately starts ringing again. With the call groups it seems to work correctly. Also, using the user/1000@domain notation, multiple registrations doesn't work, only a single device rings. Thanks!

1 1

CISCO 2811 Freeswitch IVR
by Oliver Schenk 10 Jan '12

10 Jan '12

I've been battling while creating an IVR using FreeSWITCH mod_managed and connecting through a CISCO 2811. Most things now work quite well, but I am having a few issues with the way the system answers calls (or doesn't answer calls...). I have FreeSWITCH running as a windows service on Windows Server 2008, which is connected via LAN to a CISCO 2811 with a 4 port FXO card, which is then connected to a POTS phone line. Take the following scenario: 1. Managed .NET application creates a call string and uses ESL to talk to freeswitch and originate a call: string callstring = "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567(a)192.168.x.x '&managed(ivrAppName)'"; eslConnection.API("originate", callstring); where 192.168.x.x is the CISCO IP. 2. The CISCO sees that the phone number (1091234567) starts with a 1 so it uses FXO port 1 and strips the 1 and uses the remaining phone number (091234567) to make the call. 3. My phone rings, I pick up and I can hear my IVR playing. These are my current problems: - IVR starts playing before I even pick up the phone. This means that if the system calls a mobile phone and the person doesn't pick up, the IVR will start playing and eventually the mobile phone will divert to voice mail. Obviously I then get a missed call and an sms saying I have a new voice mail, which is annoying. Instead I would like it to KNOW that no one has picked up, but I don't know how to do this. Somehow the CISCO needs to be able to tell FreeSWITCH that the call has not yet been answered. For some reason however as soon as the CISCO starts calling FreeSWITCH thinks the call is already connected. It doesn't know that the CISCO is actually still ringing. Maybe I'm doing originate the wrong way or something ... - The phone only rings for about 10 seconds before hanging up. I've tried "call_timeout", "bridge_answer_timeout". I've also tried setting CISCO "ring number". Nothing works, my phone still only rings for about 10 seconds. I don't know if this is a FreeSWITCH issue or a CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just starts playing even if no one answers the phone. CISCO Config for relevant FXO port: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service h450.2 no supplementary-service h450.3 supplementary-service h450.12 no supplementary-service sip moved-temporarily no supplementary-service sip refer fax protocol cisco sip registrar server expires max 3600 min 3600 no update-callerid no call service stop voice-port 0/3/2 output attenuation -3 no comfort-noise cptone AU impedance complex1 caller-id enable ! dial-peer voice 100 pots preference 1 destination-pattern 1T port 0/3/2 ! Many Thanks, Oliver

4 20

(no subject)
by bwibowo＠gmail.com 10 Jan '12

10 Jan '12

1 0