February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

big thanks to all freeswitch developers and contributing users
by Christian Löschenkohl 24 Feb '10

24 Feb '10

i want to say a big THANKY YOU to all contributing freeswitch community members. over one year has passed since i did fall in love with this project. it is getting better every day, one get's help and advices if needed. the admins do care about nearly every problem - no matter if it's big or small. i also did manage an opensource project and i wish i had done it with that much heart and intense power that i see here. i also hope that i can contribute back enough (questions, bug reports, wiki enhancements). wishing you all the best br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

3 2

Snom 300. Any good?
by Matthew Law 24 Feb '10

24 Feb '10

I am very new to VOIP in general and after spending some time getting a simple FS installation running on a small Xen instance, I am looking to buy my first VOIP phone. I don't need anything too fancy. I have looked at quite a few and the Snom 300 looks the most favourable so far. I need something which has a reasonably priced headset option and will allow me to make and answer calls 'as' my two businesses from my home office to the FS VM which is out on the internet. Do people on the list have experience of this handset or could you recommend another with similar features and headset available? Many thanks, Matt.

2 1

mod_fax undefined symbol
by TTNC - Technical 24 Feb '10

24 Feb '10

Hi Guys I'm having trouble getting mod_fax to load. Running on Debian testing (squeeze). Everything is installed as per - http://wiki.freeswitch.org/wiki/Installation_Guide. (dpkg-buildpackage etc) When trying to load the fax module I get: 2010-02-19 01:48:17.554935 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_fax.so **/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFDefaultStripSize** And when a fax is sent, I'm getting: 2010-02-19 01:04:57.355330 [ERR] switch_core_session.c:1490 Invalid Application rxfax I guess because mod_fax isn't loaded. I've got libtiff4 and libtiff4-dev installed: ii libtiff4 3.9.2-2 Tag Image File Format (TIFF) library ii libtiff4-dev 3.9.2-2 Tag Image File Format library (TIFF), development files ii libtiffxx0c2 3.9.2-2 Tag Image File Format (TIFF) library -- C++ interface Just tried updating to the latest svn trunk (16700M) and it hasn't made any difference. From googling, it suggests that it could be because the module is complied against a different one currently running on the system, however I'm not sure how this can be the case, there is only the one version installed. Any suggestions as to what I can try? Any help appreciated Russ

8 27

Two Ethernet Ports, One FS Instance?
by Jerry Richards 24 Feb '10

24 Feb '10

If my server has two ethernet ports, do I need two FS instances? Or can a single FS instance send/receive messages through both ports using two different sip_profiles? Best Regards, Jerry

2 1

freeswitch minimum install
by Madovsky 24 Feb '10

24 Feb '10

Hi, Is there a way to install freeswitch from source with the strict minimum xml necassary in the conf dir to run freeswitch ? Thanks Franck

3 2

SIP provider recommendation for US termination
by Rob Forman 24 Feb '10

24 Feb '10

Hey all, I'm having on-going sporadic issues with one of my SIP providers (call quality, delayed or lost DTMFs, high random PPD). Does anyone have some good experiences (for US termination) in terms of both quality and support? There are so many bad ones out there; I don't want to switch blindly. I don't know if we're supposed to share commercial endorsements on here. If you have advice, would you mind dropping me a note off-list at rob4manhere (at) gmail.com. Many thanks, Rob

3 2

SDP With T.38 in INVITE Problem
by ivdreg ivdreg 24 Feb '10

24 Feb '10

Hi all, Dose someone have a problem that if there T.38 in coming from gateway FreeSwitch drops the call because of media error ? As I see from log only T.38 port is zero and SDP has also media port. Is it possible to configure FS to do not break a call but if media is OK. 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel sofia/backend/XXXXXXXXXX@10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc] 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/ XXXXXXXXXX@10.10.1.110:7065) State Change CS_NEW -> CS_INIT 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX@10.10.1.110:7065 [BREAK] 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX@10.10.1.110:7065) Running State Change CS_INIT 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338 (sofia/backend/XXXXXXXXXX@10.10.1.110:7065) State INIT 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/ XXXXXXXXXX@10.10.1.110:7065 SOFIA INIT 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/ XXXXXXXXXX@10.10.1.110:7065 Patched SDP --- v=0 o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212 s=session t=0 0 m=audio 21108 RTP/AVP 18 4 8 0 c=IN IP4 10.10.1.110 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 m=image 21108 udptl t38 c=IN IP4 10.10.1.110 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxRateManagement:transferredTCF +++ v=0 o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212 s=session t=0 0 m=audio 17058 RTP/AVP 18 4 8 0 c=IN IP4 10.10.1.100 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 m=image 17058 udptl t38 c=IN IP4 10.10.1.100 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxRateManagement:transferredTCF 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/ XXXXXXXXXX@10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING ...... 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP: v=0 o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110 s=FreeSWITCH c=IN IP4 10.10.1.110 t=0 0 *m=audio 26850 RTP/AVP 8* a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 *m=image 0 udptl 19* 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/ XXXXXXXXXX@10.10.1.110:7065 entering state [ready][200] 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal sofia/backend/YYYYYYYYYY(a)10.10.1.110 [BREAK] 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/ XXXXXXXXXX@10.10.1.110:7065] has been answered 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec sofia/backend/XXXXXXXXXX@10.10.1.110:7065 PROXY/8000 20 ms 160 samples *2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP [sofia/backend/XXXXXXXXXX@10.10.1.110:7065] 10.10.1.100:17058->10.10.1.110:0codec: 0 ms: 20 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR: [Missing remote port] 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/ XXXXXXXXXX@10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]* 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal sofia/backend/XXXXXXXXXX@10.10.1.110:7065 [KILL] 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX@10.10.1.110:7065 [BREAK] 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX@10.10.1.110:7065) Running State Change CS_HANGUP 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX@10.10.1.110:7065) State HANGUP 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/ XXXXXXXXXX@10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to sofia/backend/XXXXXXXXXX@10.10.1.110:7065 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46 sofia/backend/XXXXXXXXXX@10.10.1.110:7065 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX@10.10.1.110:7065) State HANGUP going to sleep 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333 (sofia/backend/XXXXXXXXXX@10.10.1.110:7065) State Change CS_HANGUP -> CS_REPORTING 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX@10.10.1.110:7065 [BREAK] 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX@10.10.1.110:7065) Running State Change CS_REPORTING 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590 (sofia/backend/XXXXXXXXXX@10.10.1.110:7065) State REPORTING Thanks

3 11

Issue with making calls through FSComm and Build issues with Freeswitch 1.0.5 latest updated on 24-Feb-2010 04:05
by Suneel Papineni 24 Feb '10

24 Feb '10

Hi, I am trying to make FSComm work with Freeswitch (1.0.5 latest), but failed to make any calls. FSComm is getting registered properly and I can see SIP messages at Freeswitch. When a call is tried to make from one FSComm to another, there is no INVITE message seen in Freeswitch and also in wireshark traces as well. (OS environment is Windows XP 32-bit ). Could someone let me know if there is any specific config changes to be made to FSComm to work. (I am trying initially with FSComm pre-build binary version. Results are same). As per previous suggestions I tried to work with the latest version, downloaded Freeswitch 1.0.5 Latest updated on 24th February 2010 at 4:00am from "http://latest.freeswitch.org/". When I tried to build this, it is giving 4 errors. So build is failing. Errors are as follows: Error 201 error C2491: 'spandsp_stop_inband_dtmf_session' : definition of dllimport function not allowed d:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\src \mod\applications\mod_fax\mod_fax.c 812 mod_fax Error 202 error C2491: 'spandsp_inband_dtmf_session' : definition of dllimport function not allowed d:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\src \mod\applications\mod_fax\mod_fax.c 825 mod_fax Error 213 error LNK2019: unresolved external symbol _sip_dig_function referenced in function _mod_sofia_load mod_sofia.obj mod_sofia Error 214 fatal error LNK1120: 1 unresolved externals D:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\Deb ug\mod\mod_sofia.dll mod_sofia I am trying to build Freeswitch for a windows 32-bit system. Tried to build both Debug and Release versions but failed with the above errors. Could someone let me know from where I need to download the latest version (possibly without errors). Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

1 0

How to debug time-based routing?
by Matthew Law 24 Feb '10

24 Feb '10

After some playing around I now have a working config but it appears to be routing calls straight to voicemail based on time of day. I did see the example of this in the default config, commented it out and reloaded but I cannot see anything in the log output to verify this. Looking at the log output, I can see it hit the regex for the outside number, match that and then get sent to extension 1000 (which is correct) but at that point it is sent straight to voicemail and I'm stumped. Can someone please explain how to go about debugging this? I have the log level set to 7, if that helps. Many thanks, Matt.

2 2

Streaming conference audio to a website
by Bekele Martins 24 Feb '10

24 Feb '10

Hello. I'm new to Freeswitch, but I have a question. Is it possible to have a conference call and then stream the audio of the conference to my website so people can listen to it over the Internet? I read about mod_conference, but I couldn't find the answer to my question. Thank you.

4 4