- FreeSWITCH-users - FreeSWITCH Mailing Lists

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

2 1

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

11 27

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 !

3 2

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

2 1

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!

1 1

FS on Public IP adrress
by lloyd thomas 18 Feb '11

18 Feb '11

I have put my FS box on a public IP address so that it can serve phones at remote sites. I have not set up any providers on it yet. I would like to make an internal call between two extensions, but I get the following error when I try to call freeswitch@internal> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Content-Type] = [log/data] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Content-Length] = [121] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Level] = [7] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Text-Channel] = [0] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-File] = [sofia.c] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Func] = [sofia_handle_sip_i_invite] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Line] = [6081] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [User-Data] = [] [DEBUG] libs/esl/src/esl.c:1111 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: log/data Content-Length: 121 Log-Level: 7 Text-Channel: 0 Log-File: sofia.c Log-Func: sofia_handle_sip_i_invite Log-Line: 6081 User-Data: _undef_ 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected by acl "domains". Falling back to Digest auth. 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected by acl "domains". Falling back to Digest auth. I am using two polycom 550's to test this. Any Ideas? Thanks Lloydie T

4 4

big audio delay in bridged conference
by George Niculae 18 Feb '11

18 Feb '11

Hi All, I have two conferences set in my dialplan - one is bridged and is using mod_loopback (as advised in http://jira.freeswitch.org/browse/FS-2741 ) the other is a normal one. Extensions in dialplan looks like: <extension name="3202"> <condition field="destination_number" expression="^202-conference$"> <action application="bridge" data="loopback/app=conference:202-conference@3202"/> </condition> </extension> <extension name="3203"> <condition field="destination_number" expression="^203-conference$"> <action application="conference" data="203-conference@3203"/> </condition> </extension> With the bridged conference I hit weird behaviour: there is a delay in audio that grows in time and can even touch 9 seconds in about 10 minutes. The other conference works just fine tested in same conditions as the first one. I have monitored FS console during both conferences but noticed nothing special. (running HEAD, commit 49a5effcdf2cea9e0ddcf146cf3fe85d1872e654) Any pointer greatly appreciated, Thanks, George

2 2

CallManager trunk to FS
by Mitch Johnson 18 Feb '11

18 Feb '11

I'm having a few issues where the freeswitch 1.6 is not registering with the callmanager 8. [ERR] sofia_reg c:1765 CM8 Registration Failed with status DNS Error [503]. <include> <gateway name="CM8"> <param name="username" value="xxxx"/> <param name="password" value="xxxx"/> <param name="realm" value="CM8.localhost.local"/> </gateway> </include> I added the CM8.localhost.local to the hosts file as the CM is on the same network and doesn't need a DNS entry. Can I replace the value in the realm field with an IP Address? I did check the box MTP required on the callmanager. I've reloaded the xml file using the sofia profile external ... Thanks so much, Mitch

1 0

Which CPU to select
by Vitalii Colosov 18 Feb '11

18 Feb '11

Hello list, Could you please suggest, based on your experience, which of the CPU will be enough to handle 1000 simultaneous calls using G711 codec on all end points? Is it really needed to buy Xeon with 4 cores? or Core 2 Duo or Quad will be perfectly enough? Assuming CPU frequency is ~2.5 Ghz The following page seems like not too much updated: http://wiki.freeswitch.org/wiki/Real-world_results Thank you, Vitalie

2 1

ASR from Freeswitch to Loquendo - No Audio passed to MRCP Server
by Michel Habib 18 Feb '11

18 Feb '11

Dears, I am seeking expert advice on setting up Freeswitch to Loquendo MRCP Connection. I have freeswitch on one computer, configured for MRCP V1 - Loquendo Speech Suite 7. I have Loquendo set-up on the other computer - with MRCP V1 Profile. Using a simple lua script driving my extension, I can successfully:- - establish the connection to Loquendo Server. - define grammar - compile grammar successfully. - Begin Recognition Then it waits forever until i hangup the call. Looking at the Wireshark Traces, the RTP packets are transferred succesfully from softphone to freeswitch [i could hear my voice on the playback] But, when playing back the RTP packets between freeswitch and Loquendo, there is only deep silence [same thing with audio dump created by Loquendo]. Can you direct me where to troubleshoot this on the Freeswitch side? I will be more than happy to provide any logs you need to resolve this. Thank you, Michel.

2 1