December 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Caller ID
by Joel White 09 Dec '14

09 Dec '14

I have several installations of FreeSWITCH. I have managed to get one to dynamically create the user directory from PostgreSQL and it properly sets the outbound caller id. I have another system, very similar in how it was configured. The second installation also dynamically generates the user directory from PostgreSQL as well. I am however having an issue with setting the caller id on this other system. I have went through with a fine tooth comb and am having a hard time locating the discrepancy between the two systems. Has anyone had an issue similar to this?

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fs_cli hangs
by akhil garg 09 Dec '14

09 Dec '14

freeswitch is running on 192.168.1.1 by command freeswitch. file "freeswitch/autoload_configs/event_socket.conf.xml" has been edited <param name="listen-ip" value="172.26.1.47"/> running "fs_cli -H 192.168.1.1 -d 7" from other machine e.g. 192.168.1.2 on lan is successfull. running "fs_cli -H 192.168.1.1 -d 7" from same machine i.e. 192.168.1.1 hangs. Regards, Akhil

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Billing software
by Luis Daniel Lucio Quiroz 09 Dec '14

09 Dec '14

What other options for FS compatible software other than vBilling do you recommend me?

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packaging FS debs using existing source tree
by Bruce Lefko 09 Dec '14

09 Dec '14

I'd like to try making some changes to FS and then use the debian/util.sh to create deb packages, however it looks like that script re-checkouts the source to create the packages. Is there a way to make that script use the existing source tree that's already been checked out? Thanks!

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Log-auth-failures question
by Vitaly 09 Dec '14

09 Dec '14

Hi, list! In sip-profiles: "log-auth-failures" - write log entries ( Warning ) on authentication failures ( Registration & Invite ) in /usr/local/freeswitch/log/freeswitch.log files. Is it possible to write these entries in other file of in other directory? Thanks in advance. -- Best regards, Vitaly.

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FreeSWITCH 1.4.14 Released
by Ken Rice 08 Dec '14

08 Dec '14

New Post on freeswitch.org from krice387 check it out at http://ift.tt/1zmebiE FreeSWITCH 1.4.14 Released FreeSWITCH 1.4.14 has been released! This is routine maintenance release. Quick highlight of improvements: support for logging full timestamps with dialplan (defaults to old behavior unless requested), allow mod_http_cache to S3 services other than Amazon, and new configuration parameter add-variables-to-offer, addition of `connect-timeout` option for curl API command, allow setting CURL timeout from curl API command , respecting umask when creating new files in mod_sndfile , support for additional curl auth methods in mod_curl , fifo_position macro, improvement of accuracy in timerfd implementation , and remove support SSLv2/3 from mod_rayo due to security concerns. Tarballs and Packages available at the usual locations. New features that were added: 7c0cf50 Set managedlist to return the value to the API stream and the log in mod_managed 4037e78 Support per-module references directory in mod_managed 10ebeba Add console log level in mod_managed 33cb950 Added pure CreateStateHandlerDelegate in ManagedSession for native api usage in mod_managed 889b678 Added GetPtr to Util class for internal pointers extraction (very useful when using native api) in mod_managed 8478874 FS-6831 Addition of unset app for the chatplan and while you can already unset by calling set with var=, same with set in dialplan, this is a convenience function similar to our unset in dialplan in mod_sms [Jira: http://ift.tt/1Hwa0Xw] bc767bb Adding rfc6598.auto and adding rfc6598 space to nat.auto acl. This is the NAT444 carrier grade nat space. 1190e59 FS-6965 Set the Accept-Language into a variable when received in mod_sofia [Jira: http://ift.tt/1uLG7M3] 94278b5 FS-5159 Allow enter and exit sounds to interrupt the MOH in a wait_mod conference 1944f9a FS-6968 Changes to mod_fifo.c to add outbound_per_cycle_min in mod_fifo [Jira: http://ift.tt/1Hwa0XA] 0f2816d Add command to compile non-minified js file for testing in mod_verto f175c71 FS-6805 add support for logging full timestamps with dialplan, defaults to old behavior unless requested. Jira: http://ift.tt/11E7kpC fada4b8 FS-6977 Do not create freeswitch.serial if zrtp not enabled Jira: http://ift.tt/11E7kpG dd629c1 Add external_video_source to media handle and expose switch_core_media_start_video_thread() to start the core video thread for non-rtp based media 0eefdca FS-6947 Opus RTP payload fmtp settings ( maxaveragebitrate / maxplaybackrate ) Jira: http://ift.tt/11M58xc dd61232 FS-6979 Allow mod_http_cache to S3 services other than Amazon Jira: http://ift.tt/11E7i0R 826d428 FS-6992 Global configuration or maxplaybackrate and maxaveragebitrate from opus.conf.xml Jira: http://ift.tt/11M58xe e1c0ef5 New configuration parameter, add-variables-to-offer (default=false) (When true, all channel variables are included in the offer to rayo client in mod_rayo). 1ee325d Add `connect-timeout` option for curl API command f1df8d6 Allow setting CURL timeout from curl API command c73afe1 FS-7004 Respect unmasked when creating new files in mod_sndfile [Jira: http://ift.tt/1pgbVcN] 6f660c3 Updated mod_rayo to no longer support SSLv2/3 f198d82 FS-5666 Add ignore_connect_fail config setting to not kill the call when redis is down when using redis back-end for limit [Jira: http://ift.tt/1pgbVcP] 250234d FS-5800 Add support for additional curl auth methods in mod_curl [Jira: http://ift.tt/1pgbVcR] 1e92619 FS-6097 – added fifo_position macro [Jira: http://ift.tt/1FkkFCz] da6043f Improve timerfd implementation to be more accurate df423b8 Improve timerfd implementation to be more accurate Improvements in cross platform build supports: 0f8b993 Fix mod_say_es_ar Makefile.am 07030c6 Fix compiler warning on unmatched return type 5dee5ce FS-6953 Fix for build issues with OS X [Jira: http://ift.tt/1uLG9Um] a17be38 Add reconf target 300b8d8 FS-6973 Fix for changed Opal svn url breaking build of mod_opal [Jira: http://ift.tt/1Hwa1dY] 9673cf0 Fix for apr build issue FS-6848 freebsd arm 62a2e10 Remove hack breaking some cross compile builds. if you really need this, you should be using new enough glibc anyways. The following bugs were squashed: f87c335 Fix for issue with tracking agent availability 8f3c157 FS-6957 Fix for muting issues upon joining a conference in mod_conference [Jira: http://ift.tt/1uLG9Ue] 831832c FS-6890 Fix for FreeSWITCH forgets to send RE-INVITES [Jira: http://ift.tt/1uLG9Ug] 4eb5b38 Fix bug where re-invites needlessly re-init the codec and jb a497169 FS-6890 Additional work for FreeSWITCH forgetting to send RE-INVITES in mod_sofia [Jira: http://ift.tt/1uLG9Ug] 9c1e603 FS-6954 #comment Fix for side effect of another fix for this bug. Please test this patch. [Jira: http://ift.tt/1yzQC5V] f66f2ca FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia [Jira: http://ift.tt/1uLG9Ug] 415f82f FS-6954 Work toward fix for Freeswitch adding additional m=audio on t.38 call in mod_sofia [Jira: http://ift.tt/1yzQC5V] 5ce5199 FS-6969 Fix for unwanted profile shutdown in mod_verto [Jira: http://ift.tt/1Hwa0XE] In terms of stability these were the use cases that were fixed: f3d089a Fix a crash when trying to remove a shadow directory in mod_managed 2f1b12f OPENZAP-232 Check for digits received on sangoma isdn stack to avoid delaying moving to the ring state if all digits are received at once in overlap dialing mode. 6b8d5b2 Fix release guard timer check in freetdm 34cf3b9 FS-6980 #resolve Fix for crash when using native recording on recordstop Jira: http://ift.tt/11E7i0U ab24bde FS-5533 fix issue with busy signal being sent back to all shared lines instead of just the calling device Jira: http://ift.tt/11M58xi a279bf3 FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia Jira: http://ift.tt/1uLG9Ug 0c68bb6 FS-6957 Fix for muting issues in mod_conference Jira: http://ift.tt/1uLG9Ue 33d37ce PLIV-13 Disable SSLv3 in libcurl for mod_httapi dbc5571 FS-6983 Wrap new curl TLS macro usage with ifdefs Jira: http://ift.tt/11E7i0W 0699ea8 Fixes for various issues 87a4670 FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia Jira: http://ift.tt/1uLG9Ug 82aa331 FS-6531: #resolve set to tag on auto answer notify Jira: http://ift.tt/11M58xk 75473a7 FS-6531: #resolve set to tag on uuid_phone_event notify Jira: http://ift.tt/11M58xk 07c5cc1 FS-7003 Fix infinite loop when output sent to server without SSML configured and repeat-times=0 Jira: http://ift.tt/11E7kG0 0cf770a FS-6996: #resolve fix define change as of glibc 2.20 for _BSD_SOURCE -> _DEFAULT_SOURCE [Jira: http://ift.tt/1pgbVcT] 424df19 FS-Fix for build failure on mips[Jira: http://ift.tt/1pgbU8z] 8eaaa08 FS-6622 Set buffer size for streams based on the number of channels to avoid buffer starvation in mod_shout [Jira: http://ift.tt/1pgbVtb] 5127b64 FS-7014 Don’t touch the tech_pvt when a call has just ended, leaving us with a null tech_pvt in mod_sofia [Jira: http://ift.tt/1FkkFCF] 8330336 FS-6450 Update library [apr] Backport APR_RING_FOREACH and APR_RING_FOREACH_SAFE macros to APR for unimrcp compatibility.[Jira: http://ift.tt/1pgbVtf] c645ab3 FS-6848 #resolve #comment slight tweak to fix the logic and avoid unbalanced parens [Jira: http://ift.tt/1FkkE1B] 79de78a FS-7021 Remove extra loops when using originate_retries and the caller hangs up.[Jira: http://ift.tt/1pgbUp3]

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FreeSWITCH Week in Review (Master Branch) November 30th-December 6th
by Ken Rice 08 Dec '14

08 Dec '14

New Post on freeswitch.org from kathleen check it out at http://ift.tt/1AWUUpw FreeSWITCH Week in Review (Master Branch) November 30th-December 6th Hello, again. This week in the FreeSWITCH master branch we had 27 commits. The features this week are: timezone support for mod_say_{de,es,ja,nl,th,zh}, and the addition of a drop_dtmf_masking_tone channel variable which, if set, would replace all DTMF tones by the single tone specified in the variable. New features that were added: e55aee1 FS-7025 Add drop_dtmf_masking_tone channel_variable [Jira: http://ift.tt/1yJc1Mn] a8c5a0c FS-7048 Add timezone support to mod_say_{de,es,ja,nl,th,zh} Improvements in cross platform build supports: dc9e904 FS-7025 fix compiler warning introduced from e55aee14 [Jira: http://ift.tt/1yJc1Mn] b69c93e FS-7030 More work toward fixing FS build on Windows Visual Studio 2012 [Jira: http://ift.tt/1AWUUFW] db66cdb Fix mrcp libraries to build correctly c327455 FS-7030 More work toward getting FS to build on Windows Visual Studio 2012 [Jira: http://ift.tt/1AWUUFW] b341ff7 FS-7046: fix data types and casting on some vars to silence windows build warnings in mod_verto [Jira: http://ift.tt/1yJc4rg] 7ce5171 FS-7046 follow up on type change in mod_verto [Jira: http://ift.tt/1yJc4rg] The following bugs were squashed: 46adbec FS-7030 #comment [unimrcp] restore visual studio 2010/2012 project files added by FS project [Jira: http://ift.tt/1AWUUFW] bad5dc3 FS-7037 Fix for T38 fax break started by commit 5bbef7f1e50 [Jira: http://ift.tt/1AWUXBF] 72c3df5 FS-6891 FS-6713 #comment revert 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira: http://ift.tt/1yJc22F] 2a7b022 FS-6980 #resolve don’t crash when using native recording on recordstop the redo [Jira: http://ift.tt/11E7i0U] 35ba6a3 FS-6766 Fix verto caller ringback missing on conference bridge in mod_verto [Jira: http://ift.tt/1AWUUG2] e8cf9c7 FS-7045 Guarantee that dialed call can be joined when answered event is sent in mod_rayo [Jira: http://ift.tt/1yJc4rj] 4be6290 FS-7052 Moving jb queue swap operation out of the debug block. [Jira: http://ift.tt/1AWUXS0] 843e495 FS-7051 Preserve the annexb=no/yes status in mod_sangoma_codec [Jira: http://ift.tt/1yJc4rl] 158c1f2 FS-7002 Fix for recorded audio being choppy when diferent ptimes present and record session starts on bleg [Jira: http://ift.tt/1AWUXS3] The complete list of commits can be found here:2014_11_30-2014_12_7

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Logging Time After Transfer
by Jim Hughes 08 Dec '14

08 Dec '14

We would like our Call Detail Records from FreeSwitch to log the time after a call is transferred. I read entries in the forums that if we “enable legs” we might get that info, but it didn’t seem to work. We got lots more information, but not the time after transfer. For example: An agent receives a call and talks to somebody for 1 minute, then transfers the call to a 3rd party and that call continues for 5 minutes. We’d like the Call Detail Record to contain both pieces of information: Time Before Transfer: 1 Minute Time After Transfer: 5 Minutes Has somebody already done this so we don’t have to re-invent the wheel? Thanks, Jim.

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Building FS with mod_java support
by Victor Medina 08 Dec '14

08 Dec '14

Hi all! I´ve been trying to build Freeswitch 1.4 with Java support. It compiles just fine, installation fails with: making install mod_java make[4]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make install-am make[5]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[6]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[6]: Nothing to be done for `install-exec-am'. cp freeswitch.jar /opt/CloudVoice-vPBX/fs/scripts cp: cannot stat ‘freeswitch.jar’: No such file or directory make[6]: *** [install-data-local] Error 1 make[6]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[5]: *** [install-am] Error 2 make[5]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[4]: *** [install] Error 2 make[4]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[3]: *** [mod_java-install] Error 1 make[3]: Leaving directory `/home/administrador/freeswitch/src/mod' make[2]: *** [install-recursive] Error 1 make[2]: Leaving directory `/home/administrador/freeswitch/src' Can someone help? Sin mas a que hacer referencia, Victor Medina

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Re: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI
by Notify Me 08 Dec '14

08 Dec '14

The delay is very short, they send a100 Trying and then a 480 temporarily Unavailable back, as I can see in tcpdump. After they send the fail and its acknowledged, they end it and send an SMS back. This is a full dump 12:41:27.169928 IP (tos 0x0, ttl 64, id 34092, offset 0, flags [none], proto UDP (17), length 1146) 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 1118 INVITE sip:sina.nowahala@sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q Max-Forwards: 5 From: "234diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=04tDF6pXQU4Ng To: <sip:sina.nowahala@sip.othersipgw.com;+234diallednumber> Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 INVITE Contact: <sip:mod_sofia@my.public.ip.address:5080> User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-FS-Support: update_display,send_info Remote-Party-ID: "234diallednumber" <sip:+234diallednumber@my.public.ip.address>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1417930197 1417930198 IN IP4 my.public.ip.address s=FreeSWITCH c=IN IP4 my.public.ip.address t=0 0 m=audio 22290 RTP/AVP 8 0 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 12:41:27.385419 IP (tos 0x0, ttl 53, id 5736, offset 0, flags [none], proto UDP (17), length 384) othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 356 SIP/2.0 100 Trying Via: SIP/2.0/UDP my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q From: "234diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=04tDF6pXQU4Ng To: <sip:sina.nowahala@sip.othersipgw.com;+234diallednumber> Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.14 Content-Length: 0 12:41:27.424422 IP (tos 0x0, ttl 53, id 5737, offset 0, flags [none], proto UDP (17), length 1015) othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 987 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q Max-Forwards: 4 From: "234diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=04tDF6pXQU4Ng To: <sip:sina.nowahala@sip.othersipgw.com;+234diallednumber>;tag=8c8DZK1aK9mKN Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.14 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 X-FS-Display-Name: sina.nowahala X-FS-Display-Number: sip:sina.nowahala@sip.othersipgw.com Remote-Party-ID: "sina.nowahala" <sip:sina.nowahala@sip.othersipgw.com>;party=calling;privacy=off;screen=no 12:41:27.424789 IP (tos 0x0, ttl 64, id 34093, offset 0, flags [none], proto UDP (17), length 412) 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 384 ACK sip:sina.nowahala@sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q Max-Forwards: 5 From: "234diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=04tDF6pXQU4Ng To: <sip:sina.nowahala@sip.othersipgw.com;+234diallednumber>;tag=8c8DZK1aK9mKN Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 ACK Content-Length: 0 So you recommend a transfer on fail wav to be played? I have been tearing my hair out trying to play sounds before the call is transferred to them, so that the dialler hears something. On Sun, Dec 7, 2014 at 4:44 AM, Avi Marcus <avi(a)avimarcus.net> wrote: > How long is the delay? I assume they return a normal_clearing or fail 4XX? > If the delay is short and they have a "real" status return, then.. for > success, you can just queue a "success.wav" to play after. > And for fails, you can use > https://wiki.freeswitch.org/wiki/Variable_transfer_on_fail and transfer to > an extension that plays an error message. > > This SMS sending sounds like a normal API call.. which if it was, could be > done with curl or inside a lua/js script. > > -Avi > > On Fri, Dec 5, 2014 at 6:47 PM, Notify Me <notify.sina(a)gmail.com> wrote: >> >> Hi Avi! >> >> Thanks for the reply, I did figure it out, thanks! This is what I did, >> >> <include> >> <extension name="firstsipgw-inbound"> >> <condition field="destination_number" expression="^0123450(\d{2})$"/> >> <condition field="caller_id_number" expression="^0(\d{10})$" >> require-nested="true"> >> <action application="set" data="effective_caller_id_number=+234$1"/> >> <action application="set" >> data="caller_id_number=+234${effective_caller_id_number}"/> >> <action application="set" data="transfer_ringback=$${us-ring}"/> >> <action application="set" >> data="transfer_ringback=file_string://$${hold_music}"/> >> <action application="set" >> data="transfer_ringback=local_stream://connecting"/> >> <action application="bridge" >> >> data="sofia/external/sip:user.name@sip.othersipgw.com;${effective_caller_id_number}"/> >> </condition> >> </extension> >> </include> >> >> I can see from the logs that dials the othersipgw.com URI, and I can >> see tcpdumps that corresponds to the traffic being sent, and the >> other side gets it. The other side immediately drops the call and does >> some processing that returns an SMS to the dialler. >> >> I dont know if I can trouble you for more help.. there is no ringing >> when the transaction happens, and the dialler is unsure what is >> happening, if the call actually connected. As you might have guessed >> from the above I am trying to make it ring for a few seconds before >> the call drops. Is this possible? I would like for the dialler to be >> able to hear it ring a few times before the call is cut. >> >> Any help gratefully accepted. >> >> On Tue, Dec 2, 2014 at 2:22 PM, Avi Marcus <avi(a)avimarcus.net> wrote: >> > Hi - did you figure this out yet? >> > >> > One comment: >> > <action application="bridge" data="sofia/gateway/othersipgw/+234$1"/> >> > -- I >> > don't know if $1 is available anymore. You might want to just set that >> > as >> > part of the actual number to route, e.g. add a 1212 prefix and match it >> > again or set it as a channel variable. You can see in the logs if the $1 >> > is >> > resolving correctly. >> > >> > -Avi >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting(a)freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org > >

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