May 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

TLS not work after update
by Sergey Zhuravlov 21 May '14

21 May '14

Hi Most devices have ceased to register using TLS Happened after the upgrade: 1) Debian 6 to version 7 2) update the FS version 1.3 to 1.5 -- recompile from sources Grandstream (GXW-4008 V1.3B) device - worked perfectly before the update - ceased after. Client on csipsimple - stopped working However, continue to operate successfully Linksys spa 9xx There are some ideas about the causes of such behavior? Someone knows some ways to debug this situation? What to check? Maybe something has changed in the settings of the new versions? That's the message in the log file: tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x8726790): new connection from tls/XXX.93.XXX.2:2136/sips 2014-05-21 11:18:50.929674 [ALERT] mod_callcenter.c:519 Reuse Unused Cached DB handle db="callcenter",type="core_db" [CORE_DB] tport_tls.c:919 tls_connect() tls_connect(0x8726790): events NEGOTIATING 2014-05-21 11:18:50.929674 [ALERT] switch_core_sqldb.c:233 Dropping idle DB connection db="core",type="core_db" tport_tls.c:919 tls_connect() tls_connect(0x8726790): events NEGOTIATING tport_tls.c:1008 tls_connect() tls_connect(0x8726790): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2090 tport_close() tport_close(0x8726790): tls/XXX.93.XXX.2:2136/sips Thank you! -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com

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Reducing long gaps of silence in stock MOH
by Francis 21 May '14

21 May '14

Hi folks, I had some complaints today about callers being cut off after leaving the IVR. After testing it myself, I couldn't find anything wrong, aside from occasional periods of silence between MOH tracks, of the order of 5 to 15 seconds. I pulled the sound files into Audacity and clipped the silence down to 1/2 a second, as well as normalizing the volume on some of the quieter parts of the tracks. It's now more consistent and sounds better overall, less chance of a caller assuming they've been cut off because the track goes quiet. So I'm wondering if this would be a possible improvement to the default MOH? I'm happy to edit the files a little more cleanly than I did today. Francis

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Modifying fax header
by Peter Steinbach 21 May '14

21 May '14

Hello, we are coming from an hylafax installtion where we had our custom headers with German date and page formats. Now we are trying to use mod_spandsp fax and found the following: * The date is in english, and we would l'ike to change it to a german date or just drop it * The page# is e.g. "P.1", and we would like to change it to "Seite 1" or even "Seite 1/5" (Page 1 of 5) * The header is in bold, we would like to have it in normal letters Is there a way to change this? Where can I do that? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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dial string for more than one sip-profile to locate user
by Karsten Horsmann 21 May '14

21 May '14

Hello, i try to configure FreeSWITCH 1.2.23 with two external sip-profiles and add them to one sip-domain. The goal is that my sip-clients can use dns-srv records to get the ip of profile external1 or external2 (based on the dns-srv settings and the current client connection) for a more robust connectivity. At this time, my dialstring was* sofia/external1/username%${domain}* which didnt work of course if the user registered to the external2 profile. I also tried to change it to *username@${domain} *but that didnt work. Any hints to get the correct dialstring to locate the user on the right profile? -- Kind Regards *Karsten Horsmann*

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Successful implementation of GSMopen on Mountain Lion?
by Mervyn Yeo 21 May '14

21 May '14

Hi all, Anyone got GSMopen working on any of the Mac OSes? I'm unable to compile libcbt as there's no linux/serial.h on the Mac. Any advice would be appreciated. Thanks

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auth-calls with accept-blind-auth vs auth-messages with ????
by Luis Azedo 21 May '14

21 May '14

Hi, i have some devices that register properly (challenge/response) but can't challenge/response on INVITE and MESSAGE messages. turning accept-blind-auth on and changing some logic in mod_sofia got me to a satisfactory point concerning INVITE. i want to do the same for messages so i turned auth-messages on and was hoping that the same logic would apply when sending the message with devices that don't have to challenge/response besides register. i know what to change in sofia_presence_handle_sip_i_message but i would a session and it always comes up NULL, even with devices that have the capability to challenge/response, maybe session will only get a value for INVITE messages. what would be the best approach to make sip message perform a clean blind-auth like in INVITE without sending the proxy-authenticate back to the device ? sofia-lib / nua ? thank you.

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FreeSWITCH Week in Review(Master Branch) May 11th-17th
by Kathleen King 20 May '14

20 May '14

Hello, again. This passed week in the FreeSWITCH master branch we had 60 commits. The majority of the commits continued work toward fixing bugs reported by Coverity. Coverity is a static code analysis tool used to find bugs in source code. It is currently being used to identify and fix various issues in the FS source code and the CID numbers at the beginning of some of the commit oneline messages reference a Coverity ticket. https://scan.coverity.com/ The following bugs were squashed: cbe6f74 FS-6285 fixed issue with ANY, RR not working properly in mod_skypopen Jira: http://jira.freeswitch.org/browse/FS-6285 9a1c4f6 fixed passing a large structure by its value in mod_voicemail 3480720 Chinese say now say "one" before "ten" for numbers >99 in mod_say_zh 2b93912 Fix raw GSM AT command execution not returning the proper token count in freetdm 7dbab7b FS-6517 fixed mod_skypopen not using configured context with mod_sms chatplan with PATCH. Thanks Russell Treleaven. Jira: http://jira.freeswitch.org/browse/FS-6517 New features that were added: 3666029 add auth-subscriptions flag. sofia ua can now optionally challenge subscription requests Improvements in cross platform build supports: c006db6 fix test for pkg-config dee864e added modules.inc to .gitignore b263201 Avoid polluting CFLAGS for mod_snmp e488834 Prevent opus lib from re-configuring on every make 94ec447 Cleanup makefile library dependencies of mod_snmp Coverity fixes: ecde431 CID:1025850 Dereference before null check in mod_cidlookup e1f8e19 CID:1087632 fixed a resource leak 7d5434f CID: 1214223-1214230 If sent an invalid registration authentication header with multiple params of the same name, then only use the first of the params instead of leaking memory in mod_sofia d055f5b CID 1214263: mod_ssml - remove unused pointer value 68825ff CID:1214220 fixed resource leak in mod_dptools 619b31e CID:1214140 Buffer not null terminated, make sure hostname is always null terminated 8fa7398 CID:1025868 Simplified a function arguments never called 07b6bba CID:1214209 Resource leak, don't leak event populated by switch_channel_get_variables in mod_fsk 1286013 CID:1214255 Unused pointer value removed from mod_conference dc83348 Fix build for 31b8b55d0b31dfd58b5e6f8166a228ab2f456d3d 0469db0 CID:1214200 Dereference null return value in mod_httapi bdbefdc CID:1087631 In the unlikely case the sprintf fails, free the memory instead of leak it 6b90242 CID:1214189 Division or modulo by float zero in libteletone 452d7a3 CID:1214163 Unchecked return value, check return value of switch_ivr_play_file and bail out if it fails in mod_fifo c1f7887 CID:1024580 removed logically dead code from mod_commands a997331 CID:1214125 fixed the size of the the destination buffer in mod_voicemail_ivr 455ed10 CID:1214130 Buffer not null terminated in mod_xml_curl and mod_xml_scgi c19aa58 CID: 1214222 fixed a resource leak in mod_sofia 386000d CID:1024418 fixed unintentional integer overflow in mod_dptools 92c6f3a CID:1214192 Dereference after null check, don't crash when no file name passed on file open in mod_httapi a67389a CID:1025879 removed logically dead code in mod_cidlookup 3e18f08 clean up Coverity warnings and improve error checking in mod_voicemail 21a5d16 CID:1024581 removed logically dead code from mod_commands b10c865 CID:1025849 Dereference before null check in mod_cidlookup b7b1420 CID 1214210: mod_ssml - resource leak when parsing SSML ebc0a99 CID:1024250 Dereference before null check. CID:1024251 Dereference before null check, we don't need to check these, as they are already used above in all cases bf8b88c CID: 1210583 removed dead code reported by Coverity 269db09 CID:1214232 fixed a resource leak reported by Coverity in switch_ivr d8d58fd CID:1214141 Buffer not null terminated in mod_unimrcp 6a8002d CID:1024582 removed logically dead code from mod_commands 622cd21 CID:1214231 don't leak when using invalid syntax 73c1e09 CID:1024579 removed logically dead code from mod_commands 44401e8 CID:1023936 Unused pointer value 31b8b55 CID:1214216 fixed a resource leak in mod_dingaling 179e07c CID:1214219 fixed resource leak in mod_hash 65d9cf6 CID:1214237 String not null terminated 1680518 CID:1214256 Unused pointer value removed from mod_conference 4238339 CID:1023970 removed unused pointer value from mod_dptools 7c72e94 CID:1214158 Buffer not null terminated in mod_fifo d2884ac CD:1214211 fixed resource leak in mod_sms 54fc2f5 CID:1023975 Unused pointer value in mod_commands b58bbd1 CID:1214233 Pointer to local outside scope 9b1ebff CID:1024270 Dereference before null check in mod_commands 4653d78 CID:1087387 Unused pointer value 29e2263 CID:1210789 Make return value of a function available from the another calling function and add small status check in mod_callcenter 4e888a2 CID:1214196 Dereference null return value, if we hit this condition, something very bad has happened, we should never have the private not set if we get to the hangup hook, log error instead of crashing in mod_spy 388e119 CID: 1214221,1214222 fixed a resource leak in mod_sofia 1b0bdaf CID:1214212 Resource leak in voicemail_list in mod_voicemail 19f3ab5 CID:1214218 fix memory leak in conference_event_channel_handler Feedback is welcome at kathleen.king(a)quentustech.com and the commits for this week are in the attached file along with the corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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change sip contact uri options
by Nuno Reis 20 May '14

20 May '14

Hi! I've been looking at FS source code and realized that at least the invite_contact can be changed by declaring the contact-user, ext-sip-ip and ext-sip-port profile variables. I would like to know what other option do i have to change contact(all the sip_uri string) in the different stages of a call: REGISTER/INVITE/SESSION_IN_PROGRESS, ACK, OK, etc... I would also like to be able to change contact in runtime dynamically, what option do i have in this case? Thanks. -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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TEST
by Brian West 20 May '14

20 May '14

IGNORE -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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OS X Wiki updated for 10.9.3 and new ldns prereq
by Mario G 19 May '14

19 May '14

For those interested: https://confluence.freeswitch.org/display/FREESWITCH/Installation+and+Setup… OS X 10.9.3 tested 10.6 through 10.9 tested with new ldns prerequisites, also tested with updated brew packages The OS X FreeSWITCH installer was updated to include ldns: https://confluence.freeswitch.org/display/FREESWITCH/Installation+on+OS+X+u… Links to git clone 1.2 were corrected (odd since I got them from the Linux page and they worked when tested) now using the same as Linux page git clone for 1.2. Mario G

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