February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

newbies is stuck / windows version of freeswitch won't start up after reboot
by Edward de Jong 19 Feb '11

19 Feb '11

I installed Freeswitch (by downloading the .EXE version for windows x64) and was running it okay on Windows 7 Professional, 64 bit edition, but after a restart, when I try to run fs_cli it gives me a socket error and fails. I am not really running anything else much on my machine, and i figure somehow that port has gotten blocked. I turned off windows firewall, but still it won't run, i can't figure out why it was running so well, but now on a clean restart it won't start up. I didn't install it as a startup program, I was manually running fs_cli and freeswitchconsole, which are the two freeswitch programs, yes? thanks in advance. edj

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Skype 2.0.72 for Linux
by Michael Collins 19 Feb '11

19 Feb '11

Does anyone know where to find this? It's the only one that the wiki says works with mod_skypopen but not even Google seems to be able to find one. (I find "2.0.72" but it links to the latest beta on the skype.com site which is not what we want.) Any suggestions are appreciated. -MC

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IVR application
by Jason b.a 19 Feb '11

19 Feb '11

aha thx MC ! Ken give me ur suggestions pls , if i need a ivr application that controlled the media server using SIP and an embedded control language !

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ESL socket outbound in C++
by Stephen Wilde 19 Feb '11

19 Feb '11

I'm learning to use ESL in C++ to do a call control and I want to share my code with the community. I had a little difficult at startup so I hope that this code can help someone. The code is: http://pastebin.freeswitch.org/15418 Using "esl_oop" module, I have implemented a basic ESL Channel class (that can be extended) so the "main" event handling loop can be very simple, for example for an unuseful esl bridge application, can be: ESLchannel legA(&connection, inbound_uuid); ESLchannel legB(&connection); legB.Originate(dialstring) while (1) { ESLevent * event = connection.recvEventTimed(1000); if (event) { const char * ename = event->getHeader("Event-Name"); const char * event_uuid = event->getHeader("unique-id"); // It's an event for LegB ? //------------------------- if (legB.Uuid() == event_uuid) { if (!strcmp(ename, "CHANNEL_PROGRESS_MEDIA")) { legA.PreAnswer(); legA.BridgeTo(legB); } else if (!strcmp(ename, "CHANNEL_ANSWER")) { legA.Answer(); legA.BridgeTo(legB); } else if (!strcmp(ename,"CHANNEL_HANGUP")) { const char * cause = event->getHeader("Hangup-Cause"); legA.Hangup(cause); } } // It's an event for LegA ? //------------------------- else if (legA.Uuid() == event_uuid) { if (!strcmp(ename,"CHANNEL_HANGUP")) { const char * cause = event->getHeader("Hangup-Cause"); legB.Hangup(cause); } } delete event; } }

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NAT issue
by Spencer Thomason 19 Feb '11

19 Feb '11

Hello all, I'm having a strange NAT problem. My Freeswitch server has a public ip and the endpoints which are Polycom IP 650s are all behind NAT with mainly Netgear SRX 5308 routers at multiple locations. The server is on a public class C network and one of the locations has its router on the same class C. I.e x.x.x.7/24-router, x.x.x.240/24-server. I'm only experiencing this with 2 of about 30 extensions. The first endpoint to register from this location gets detected as NAT but the others with the ports remapped do not. This maybe explains it better: Call-ID: 4feea796-bd96e2ed-e16d27ac(a)10.6.71.103 User: 1103(a)x.x.x.240 Contact: "user" <sip:1103@10.6.71.103;received=x.x.x. 7:5060;fs_nat=yes;fs_path=sip%3A1103%40x.x.x.7%3A5060> Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769 Status: Registered(UDP-NAT)(unknown) EXP(2011-02-18 11:15:47) EXPSECS(1143) Host: pbx10.5ninesolutions.com IP: x.x.x.7 Port: 5060 Auth-User: 1103 Auth-Realm: pbx10.5ninesolutions.com MWI-Account: 1103(a)x.x.x.x Call-ID: d21b71b2-75baa80d-768c0b08(a)10.6.71.104 User: 1101(a)x.x.x.240 Contact: "user" <sip:1101@10.6.71.104> Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769 Status: Registered(UDP)(unknown) EXP(2011-02-18 11:16:48) EXPSECS(1204) Host: pbx10.5ninesolutions.com IP: x.x.x.7 Port: 1036 Auth-User: 1101 Auth-Realm: pbx10.5ninesolutions.com MWI-Account: 1101(a)x.x.x.x Call-ID: a74e61a6-1863ea7d-789686c4(a)10.6.71.105 User: 1102(a)x.x.x.240 Contact: "user" <sip:1102@10.6.71.105> Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769 Status: Registered(UDP)(unknown) EXP(2011-02-18 11:17:23) EXPSECS(1239) Host: pbx10.5ninesolutions.com IP: x.x.x.7 Port: 1036 Auth-User: 1102 Auth-Realm: pbx10.5ninesolutions.com MWI-Account: 1102(a)x.x.x.x Call-ID: 100b5148-6a47060a-13c4-50029-c844-22fc4dd3-c844 User: 1801(a)97.67.110.240 Contact: "user" <sip:1801@10.6.71.106:5060> Agent: MP201 B 1FXS/3.0.1_build_19 Status: Registered(UDP)(unknown) EXP(2011-02-18 11:26:23) EXPSECS(1779) Host: pbx10.5ninesolutions.com IP: 97.67.110.7 Port: 1036 Auth-User: 1801 Auth-Realm: pbx10.5ninesolutions.com MWI-Account: 1801(a)x.x.x.x ------- Another Location ---- Call-ID: dca2a276-4eb94f97-c55177b0(a)10.1.10.186 User: 4204(a)x.x.x.240 Contact: "user" <sip:4204@10.1.10.186;received=x.x.x.x: 5060;fs_nat=yes;fs_path=sip%3A4204%40x.x.x.x%3A5060> Agent: PolycomSoundStationIP-SSIP_5000-UA/3.3.1.0769 Status: Registered(UDP-NAT)(unknown) EXP(2011-02-18 11:28:04) EXPSECS(1880) Host: pbx10.5ninesolutions.com IP: x.x.x.x Port: 5060 Auth-User: 4204 Auth-Realm: pbx10.5ninesolutions.com MWI-Account: 4204(a)x.x.x.x The port and IP are detected correctly but the contact is a private IP. Any clue as to why the two extension 1101 and 1102 are any different? Thanks, Spencer

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Need some help. Novice freeswitch setup gone wrong
by Hareem Haque 19 Feb '11

19 Feb '11

I have a little problem i setup the blue.box app it runs very well. Now i am having some problems. A. How can i stop using the xml cdr and just use text file only cdr which are dumped into a folder called /mnt/fscdr B. How can i setup the pbx as a multi tenant system and keep track of all the users we have 6 departments with 20 employees each. C. Can i setup trunks in a way that SIP trunk A can only be used by Department A. Any help is greatly appreciated. Regards Hareem Haque Best Regards Hareem. Haque

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Aastra phone registration lost
by Philippe Sultan 19 Feb '11

19 Feb '11

Hello to the FreeSWITCH users, I'm experiencing problems registering an Aastra 6731i phone to FreeSWITCH. The registration process itself works fine, but the phone never re-registers to FS after the expiration time specified in the expires paramater set in the Contact SIP Header Field. And therefore becomes unreachable from FS after some time. The phone seems to be buggy because it simply ignores the value given in the expires parameter. Testing with Asterisk, the phone remains accessible, because Asterisk adds an Expires HF in addition to the expires param in the Contact HF. Did anyone observed what's described here? I've tested several Aastra firmwares with the same result. I have also tried to set the registration expiration time on the phone itself without success. I believe I can cope with this situation by adding an Expires HF from the (SIP Express Router) proxy that stays between the phone and FS, but I'm sure there's a simpler way to get a proper registation process. I'm using FreeSWITCH 1.0.6 (that comes along with the Blue.box GUI). Kind regards, Philippe

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IVR application
by Jason b.a 19 Feb '11

19 Feb '11

i am not going to use the embedded dialplan, do the application interface has SIP API such as " onInvite " for example , or no !

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loss of audio for 30sec in middle of call
by Fraser Redmond 19 Feb '11

19 Feb '11

We've got 10 callers using our FS server (outbound call-center), and they're reporting that during some calls they lose audio for a while (it varies, but they've said half a minute is typical). It's calling from softphone -> FS -> gateway It's intermittent though - happens to all the users, but only some calls, and it happens frequently for a while, then goes away for a while. I can't think where to start looking to find the problem - still trying to arrange for them to keep a record of affected calls so I can look in the logs, but I have looked at a couple of examples already with no obvious problems there. (Besides, the logs are generally quiet during the call anyway.) Any suggestions of where to look or what else to try? In the wiki it talks about some new sofia debug commands, but doesn't explain them at all - anyone know if they'd help? http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP sofia global debug sla sofia global debug presence Or do I need to packet-trace everything, then troll through that? Cheers, Fraser

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flite build error
by envelopes envelopes 18 Feb '11

18 Feb '11

making install mod_flite making in ... making in include ... making in src ... making in src/audio ... making in src/utils ... making in src/regex ... making in src/hrg ... making in src/stats ... making in src/speech ... making in src/lexicon ... making in src/synth ... making in src/wavesynth ... making in src/cg ... making in lang ... making in lang/cmulex ... making in lang/usenglish ... making in lang/cmu_us_kal ... gcc: Internal error: Killed (program cc1) Please submit a full bug report. See <file:///usr/share/doc/gcc-4.4/README.Bugs> for instructions. make[6]: *** [../../build/armv5tel-linux-gnu/obj/lang/cmu_us_kal/cmu_us_kal_res.o] Error 1 make[5]: *** [../build/armv5tel-linux-gnu/obj/lang/.make_build_dirs] Error 2 make[4]: *** [build/armv5tel-linux-gnu/obj//.make_build_dirs] Error 2 make[3]: *** [/root/fs/freeswitch-1.0.7/libs/flite-1.3.99/build/libs/libflite_cmu_us_awb.a] Error 2 make[2]: *** [install] Error 1 make[1]: *** [mod_flite-install] Error 1 make: *** [mod_flite-install] Error 2 my system is debian. thanks!

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