- FreeSWITCH-users - FreeSWITCH Mailing Lists

Caller ID prefix missing
by Paul Atreides 01 Apr '15

01 Apr '15

Hi, when I place a call over the gateway to the PTSN I see the dialed number without the leading 0 prefix in the history of my Grandstream GXP2140. Is there a way to add the 0 back again? Thank you <include> <extension name="isdn-outbound"> <condition field="destination_number" expression="^0(\d+)$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/isdn/$1"/> </condition> </extension> </include>

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OPUS at 48 khz vs OPUS at 8 khz
by doancea 01 Apr '15

01 Apr '15

Hi All, We've noticed a high load on our transcoding servers when doing OPUS<->PCMA transcoding. All our client software is using opus@48khz, but inside the audio payload we have narrowband audio (forced with OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND) ) . Questions : 1. How can we avoid resampling ? We've noticed this log line showing twice for each call: 29128b92-d842-11e4-877f-9bd23fac7445 2015-04-01 07:39:14.798759 [NOTICE] switch_core_io.c:1287 Activating write resampler If we modify the client software to use opus@8khz and leave freeswitch as it we have very bad audio (nothing intelligible). But If we modify the client software to use opus@8khz and change mod_opus.c in freeswitch so it would use only OPUS@8khz then we have proper audio. 2. We have noticed that there is a "samplerate" param in fmtp (which is not in this RFC: https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03 ) it's used only for configuring the decoder. Why only for the decoder ? If "samplerate" is set then it's passed to opus_encoder_create() , but not to opus_decoder_create(). 3. How about using sprop-maxcapturerate to set the sample rate of the decoder ? Regards, Dragos Oancea

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Announcement: Kamailio becomes systemd-rtc-server
by Alex Balashov 01 Apr '15

01 Apr '15

For immediate release: ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software vendor specialising in Kamailio-based service delivery solutions for the VoIP ITSP market, is pleased to announce that it, in collaboration with Red Hat Software and Ringfree Communications, has finalised the absorption of the Kamailio SIP Server into the 'systemd' system management platform for Linux. The new component shall be called 'systemd-rtc-server', or 'Systemd Real-Time Communication Server'. Alex Balashov, principal of Evariste and leader of the tri-vendor collaboration effort, will officially announce the handover of the reigns of the Kamailio project to the personal leadership of Lennart Poettering at the upcoming Systemd Real Time Communications World conference, to be held in Berlin on 27-29 May of this year. John Knight, Director of GNOME 3 Integration and part-time usability consultant at Ringfree Communications, based in Hendersonville, North Carolina, summarised the triumphs of the long-standing integration effort. Remarked Knight: "The industry has recognised for years that a SIP proxy is a basic building block in the 'init' subsystem of any Linux host. In this age of multimedia communication with voice and video, it was a travesty that systemd handled time synchronisation, network configuration, login management, logging, and console, but not SIP message routing." Sean McCord, a veteran partner at Atlanta-based integrator CyCORE & Docker, was quick to concur: "SIP calls are much easier to troubleshoot with binary logs. Combined with packet captures of TLS-encrypted WebRTC calls, systemd-journald is the ultimate call setup troubleshooting methodology of the responsive, kinetic enterprise." To support the integration of Kamailio into the ecosystem of every major Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio' modules for the project. Balashov stated, "We fully expect to use the D-Bus API to achieve gnome-session integration with systemd-rtc-server-usrloc, but we aren't going to leave Windows users behind; KamailioSvcHost.exe will support Domain Controller policies for G.722 in Active Directory forests." Despite an aggressive delivery timeline by the tri-vendor consortium behind systemd-rtc-server, industry commentators have widely lambasted the fact that it took so long for Kamailio to become integrated into systemd. Fred Posner, solutions architect at The Palner Group in Fort Lauderdale, Florida, recently wrote in a widely-publicised blog post: "sr-dev have been keeping their heads in the sand for too long. For years now, it has been completely obvious and self-evident to anyone with half a brain that all kinds of VoIP software should be included in systemd. It's a basic building block of the whole OS, having absorbed functionality previously provided by all kinds of packages like util-linux and wireless-tools." John Knight of Ringfree accepted the criticism readily, but advocated a forward-thinking orientation focused on breaking with the uncertainty of the past: "In the absence of a SIP component for routing calls to the PSTN, some people thought, 'systemd has no clear direction apart from the whims of its developers, and is a perpetually moving goal post.' Well, a SIP server should put an end to that whole discussion; that's exactly what was missing, and now that we have systemd-rtc-server, we've eliminated all doubts about the coherence, conceptual integrity and finality of systemd." -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Error Creating SIP UA for ......
by T Fred Farmington 31 Mar '15

31 Mar '15

Newest challenge.... I built a new Windows Server box having the same single network configuration (IP address, etc.) as a previously working FreeSWITCH server which was running on a Virtual server (Win Server 2008). Just in case it should matter, this new box is running Win Server 2012. I next Installed FreeSWITCH onto the new physical server. Then I copied all of the files from the working FreeSWITCH installation running on a Windows Virtual workstation to a new physical box and over-wrote all of the files under the FreeSWITCH 'root' directory. After copying these files I changed the network address of the Virtual Server so as to prevent network conflicts. And then I shut down the Virtual Server. On the new physical server I attempted to Start Up FreeSWITCH. The FS Service started fine, but looking at FS_CLI, I began getting error messages. I have configured the physical server's network card to Not use IP_V6 at all - the same as it was in the Virtual server. And in the FS conf/sip_profiles/internal_ipv6 & conf/sip_profiles/external_ipv6 directories I have no xml files And no where was there a definition of local_ip_v6 in either the working Virtual server nor in the new physical server. The error messages are: Error Creating SIP UA for profile: external-ipv6....... Error Creating SIP UA for profile: internal-ipv6....... The likely causes for this are.... 1. Another application is already listening on the specific address 2. The IP the profile is attempting to bind to is not local to this system Questions: * This is a brand new server box only running FS, so I can't imagine there being something else listening to the address. If so how would I know? * Since no IP for ipv6 was configured on the network card, nor within the FS xml files on the working Virtual server and the same exists on the new server box - what is going on? Thanks, TF

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Question re: local_ip_v4
by T Fred Farmington 31 Mar '15

31 Mar '15

We have 2 network cards on our FS server. * One IP goes to the 'outside' world (IP = 107.1.nn.nn) * And the other IP goes to the 'inside' world (IP = 192.168.nn.nn) In spite of making suggested changes to vars.xml when FS launches it sets itself up to 'listen' to the 'outside' world (IP = 107 etc.) But the result of that is that none of the in-house SIP phones can register to it. The changes I made (following suggestions found on the web) to vars.xml are as follows: New entry: <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.nn.nn"/>  This new line was entered immediately before <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> Additionally another web reference suggested that the following 2 lines be added near the bottom of vars.xml <X-PRE-PROCESS cmd="set" data="internal_sip_ip=192.168.nn.nn"/> <X-PRE-PROCESS cmd="set" data="internal_rtp_ip=192.168.nn.nn"/> PROBLEM: None of that seems to work. Even after not only ReStarting the FS Service, but also after ReStarting the FS Server itself and then Starting the FS Service When through FS_CLI I run: sofia status and/or sofia status profile internal both of them show the 107 IP instead of the 192 IP Those changes have not caused FS to 'see' and 'listen to' the 192 port. What am I doing wrong? Your advice/suggestions would be greatly appreciated. Thanks TF

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User registering on wrong sofia profile ??
by Tim Smith 31 Mar '15

31 Mar '15

Hi, Any ideas why user 2110 is coming up on external rather than internal ? 2005|default|v.example.com|default|sofia/internal/sip:2005@10.10.10.1:5060 2110|default|v.example.com|default|sofia/external/sip:2110@10.10.10.2:61039 The directory XML for user 2110 is no different to that for 2005, so I've no idea why Freeswitch is behaving like this ? I've taken a look around the sofia xml files but can't find anywhere in there that might be distinguishing between user extension numbers. My vars.xml is correctly set so that v.example.com (not my real domain) is the registration domain. Any ideas ???

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XML CDR unpredictable ?
by Sharath Kumar 31 Mar '15

31 Mar '15

All, I use mod_cdr_xml with xml_curl and receive messages on the backend PHP for call control logic. I am a little mystified by the XML CDR requests. I see in the logs CS_HANGUP to CS_REPORTING but I don’t always see the XML requests but I do see for dialplan so the call does succeed. I don’t understand why the CDRs XML requests are intermittent? Also, I changed the config to log the b-leg cdr as well, it didn’t seem to make any impact. So for each leg do I expect a CDR ? Also,if I do a redirect and then a bridge I only see the redirect app’s cdr and not the bridge apps. I need the bridge app’s cdr since It contains QOS metrics for that call. A little help would be appreciated. Thank you Sharath

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Changing BLF lamp persistently
by Paul Atreides 31 Mar '15

31 Mar '15

Hi, does someone know how to change the BLF lamp persistently? I found the channel variable presence id but this one will only last as long as the channel is active. Is there a way to change it permerently? Thanks for helping.

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FreeSWITCH Week in Review (Master Branch) March 21st-27th
by Ken Rice 30 Mar '15

30 Mar '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/19qOGWv FreeSWITCH Week in Review (Master Branch) March 21st-27th Hello, again. This passed week in the FreeSWITCH master branch we had 18 commits. The features for this week include: support for mute in mod_verto and support for filtering on file:func to mod_logfile mapping. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7401 Add support for mute in mod_verto FS-7402 Add support for filtering on file:func to mod_logfile mapping Improvements in build system, cross platform support, and packaging: FS-7383 Fixed a memory leak causing building Freeswitch with clang AddressSanitizer enabled to fail FS-7149 The Windows portion of the update to use flite-2.0.0-release The following bugs were squashed: FS-7300 Fixed mod_rayo to properly handle errors in mod_unimrcp FS-7385 Fixed a segfault caused by an invalid contact URI and instead fail the call with 502 bad gateway if the outbound leg returns a redirect with an invalid URI. FS-6954 Detect when we have T.38 in nomedia or proxy media mode and apply same fixups as in media mode. FS-7391 Corrected currency for mod_say_de it’s now “Ein Euro und Ein Cent” rather than “Eins Euro und Eins Cent” FS-7396 Fix for WebRTC call setup hang when Chrome 42 includes IPv6 ICE candidate FS-7386 Allows auto-adjust to fix media endpoints after receiving a 183 then a 200 from a different media endpoint, like in the case of proxy forking scenarios

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FreeSWITCH 1.4.18 Released!
by Ken Rice 30 Mar '15

30 Mar '15

New Post on freeswitch.org from krice387 check it out at http://ift.tt/1Izjq3R FreeSWITCH 1.4.18 Released! FreeSWITCH 1.4.18 has been released! This is routine maintenance release. Source Tarball available at http://ift.tt/1Izjssi Debian and Yum Repos have been updated as well. See the release notes below for a list of notable changes. For additional information about the FreeSWITCH release process, please see http://ift.tt/1E59SyO . FreeSWITCH 1.4.18 Release Notes FreeSWITCH 1.4.18 is a routine maintenance release. New features that were added: FS-7201 Set ANI plan and ANI type for ftmod_libpri FS-7209 If ANI TON is not interpreted correctly by libpri, fallback to calling TON/type. FS-7265 Add mongo_find_n API FS-7269 Add error logs in mod_java FS-7284 A simplification of command line arguments to allow for using -base instead of specifying each directory when using alternate configs. FS-7285 Allow eavesdrop to bridge only one leg FS-7164 Added support for URL attribute in the grammar tag for mod_rayo. This is useful for MRCP engines to look up their grammars directly. FS-7299 Implement cookie-file option for mod_xml_cdr FS-7302 Added params to fs_encode.c: -c for path to conf_dir -k for path to log_dir -m for path to mod_dir FS-7309 Allow removal of User-Agent header from the sip message FS-7304 Multiple and reversed ranges for XML dialplan date and time conditions FS-7312 Update mod_verto to proxy additional variables FS-7323 Add ability to force URL refresh in mod_http_cache using {refresh=true} parameter that can be prefixed to a URL to force refresh when using http:// https:// file formats or the http_get API. And added http_remove_cache API call to manually expire a cached URL. FS-7354 Filter feature ported from mod_event_socket to mod_erlang_event Improvements to the documentation: FS-7362 Add minimal configuration for configuring FreeSWITCH from scratch Improvements in build system, cross platform support, and packaging: FS-7149 Update Windows build to use flite-2.0.0-release FS-7346 Update mod_mongo driver to 1.1.0 FS-7122 Fixed issues building on CentOS 5 and other distributions with older autotools FS-6520 Fix for libv8 build issue using MSVC 2013 FS-7245 Don’t rebuild core on mod_foo-clean targets FS-7270 Set the makefile to look for libtool-bin first and update libjpeg-dev to libjpeg8-dev in Debian makefile FS-7318 Debian rules update to handle a pre-bootstrapped orig file FS-7149 Update freeswitch.spec for flite-2.0.0 FS-7236 Fix code before declaration in mod_conference FS-7264 Fix signed/unsigned warnings on Windows building ws.c FS-7294 Enable -Werror when building with clang compiler FS-7296 Fix build error on newest gcc FS-7314 Fix for configure error caused by a broken openssl 1.0.2 includes FS-7322 Fix for issues building on CentOS 5 and other distributions with older autotools FS-7340 Remove json-c dependency in favor of our own json code FS-7350 Add –enable-address-sanitizer configure flag to enable clang address sanitizer FS-7355 Fix rpl_realloc symbol missing link error that can occur when using clang The following bugs were fixed: FS-7193 Fix for sofia contact being encoded which makes it impossible to call a registered user FS-7191 Edit pgsql example connection string to remove unnecessary option that may cause a failure on some systems FS-7205 Do not url encode unless an ‘@’ is in the uri FS-7211 Fix for sofia_contact returns unable to locate registered user FS-7208 _undef_ as the header and/or ident will make it be an empty string which is the same you were doing on your local builds in mod_spandsp FS-7214 Fix segfault caused by bad command argument bounds checking for flush and delete in mod_memcache FS-7217 Use upper case when you query FS-7197 If the span has been already fully stopped and ftdm is not running, return success from the span stop function. FS-7235 Fix for call recording deleting recorded files in append mode if appended data is shorter than RECORD_MIN_SEC FS-7236 Added lock to prevent a race condition and segfault in mod_conference FS-7236 Fix mutex use before init error caused by 27c8622 0dc48df Fix for a bug from original implementation, cannot send call state about state destroy, this is an internal state and the session is already destroyed. FS-7256 Fix for being unable to load mod_java FS-7252 Fix for 6-year-old regression from commit 525f1ac back in 2008 FS-7260 Fix for L16@16000h with Asterisk negotiation issue FS-7236 Re-factor to fix audio problem from commit 7c63670 FS-7250 Removed the FreeSWITCH core handler for SIG_CHLD because it isn’t necessary anymore and it causes dependent libraries that tried to start a child process to hang waiting on a signal that FreeSWITCH core intercepted. FS-7066 Fixed a bug causing higher cpu load averages on older kernels with related bugs FS-7253 and FS-7231 FS-7298 Fix race condition when callcenter member cancels the call FS-7301 Fix for issue faxing to numbers with a pass through tone FS-7192 Exclude Expires header in INVITEs responding to an auth challenge in mod_sofia FS-7308 Only log SLA SQL query SQL when debugging is enabled in mod_sofia FS-7306 Fix for fs_encode in mod_spandsp sleeping too much FS-7230 Fixed a memory leak in mod_conference FS-7307 Fixed buffering issue when recording calls in native format FS-7126 Fixed coredump when calling the translate application FS-7313 Fix for coredump when passing invalid params to the vm_fsdb_msg_email api in mod_voicemail FS-7339 Move the creation of view sql statements for basic_calls and detailed_calls to happen after the creation of the tables so the creation works and won’t have to be run a second time. FS-7342 Fixed a crash regression in mod_conference caused by FS-7230 FS-7305 Fix for making embedded versions of FS startup and shutdown faster, like in the case of tone2wav. FS-5570 Patch to add “multi” parameter to group api command. When the “multi” parameter is present, the group command will return a list of group members delimited by :_: which allows for multiply-registered endpoints to participate in a group. FS-7300 Handle all MRCP completion causes in SPEECH-COMPLETE event and validate load input grammar URLs

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