- FreeSWITCH-users - FreeSWITCH Mailing Lists

Inbound call through Registered gateway
by Luis F Urrea 16 Apr '14

16 Apr '14

Hi all, I am trying to receive a call through a gateway that is suing registration on external profile and I don't clearly understand how to configure the extension to receive the call. I appears from the example.xml gateway configuration that the an extension name with the username used for gateway registration can be used to get the dialplan to accept the call. I assume this needs to be configured on the public context, however I see the INVITE coming in, but the dialplan never kicks in. Do I have to receive the INVITE at port 5080? If that is the case, registration doesn't really make sense. public.xml <extension name="tone_detect"> <condition field="destination_number" expression="^tone_detect$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> Gateway config <include> <gateway name="tone_detect"> <param name="username" value="tone_detect"/> <...snip...> </include> Thanks in advance for your help.

2 7

Freeswitch Week in Review(Master Branch) April 6th-12th
by Kathleen King 15 Apr '14

15 Apr '14

Hello, again. This week in the FreeSWITCH master branch we had 51 commits. This week was mostly filled with miscellaneous work toward clean up and stability fixes. If you are interested in seeing more details about what was committed, please see the attached text document. In other news, ClueCon is still looking for presentation proposals to be submitted to marketing(a)cluecon.com <mailto:marketing@cluecon.com>. If you have a topic that you think people would be interested in hearing about, please send in a proposal! The following bugs were squashed: f1d7721 FS-6448 --resolve revert string func patch because some things free it Jira: http://jira.freeswitch.org/browse/FS-6448 72c5205 FS-6436 disable PLI and instead only use FIR in switch_core_media Jira: http://jira.freeswitch.org/browse/FS-6436 7428746 fixed allowance of duplicate keys in switch_hashtable Jira: http://jira.freeswitch.org/browse/FS-6453 de7d5a9 Resolve key length issue with AES GCM 128/256 a31491d pulling in a patch to fix a bug with AES_GCM and key size in libsrtp 80e2034 fixed read of memory out of 'seq' variable bounds and fixed AEAD IV calculation for big-endian machines Improvements in cross platform build supports: 7bc92a1 ESL-89: fix install path and single_command require 23fe7f5 Add libyaml pkg-config LIBS to LIBADD not LDFLAGS c915139 Make sure to copy pcre.h to library path, when building Win32/Debug in VS2012. d6fa81b ESL-89: --resolve add rubymod-install and add ability to use a ruby not in your path by setting RUBY env var 4bdb712 FS-353 Ungetlib libmemcached c86b939 FS-353 Show message in configure when YAML isn't there In terms of stability these were the use cases that were fixed: 2ace7bc Fix memory leak in phrase macro playback in switch_ivr_play_say 264a12a fix memory corruption in dial, input, output in mod_rayo 3f8cf4a Fix memory leak in subscription handling in mod_sofia 886a730 Avoid leaking memory on mod_hash unload a866239 fixed input crash introduced in 2014-04-07 commit in mod_rayo 711e49c fixed leaking hash tables when reloading mod_fifo 65fd534 fixed a segfault in console autocomplete when calling delete with no argument 2dba7dd Avoid leaking memory on mod_cdr_sqlite unload aae1954 FS-6442 fixed a mem leak on abandoned pickup calls in mod_dptools Jira: http://jira.freeswitch.org/browse/FS-6442 237da22 Avoid leaking memory on unload of mod_dptools f754057 ESL-90 fixed crash caused by receiving an event without a content-type header in esl 39f960a Avoid leaking memory on mod_voicemail unload 1a71cf8 Fix memory leak on memory error in switch_ivr_play_say 4178688 add switch_hashtable_insert_destructor so you can insert a pointer into a hash with a custom destructor and use it in spandsp to fix a leak on reloadxml with the tone_descriptor tables and fix a bunch of random tiny leaks etc 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

1 0

OpenSIPS configuration for 2 or more FreeSWITCH installs
by Marcin M. 15 Apr '14

15 Apr '14

Hi , iam using this configuration http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSW… for load balancing FreeSWITCH servers, everything works fine if i use port 5060 ( user -> opensips-> freeswitch ). If i want to use other port for incoming calls from providers , let say 5080 as is in external profile in Freeswitch ( provider -> opensips:5080-> freeswitch:5080 ), how i can do that ?? maybe someone have same situation ?? or maybe i can configure gateways on internal profile on freeswitch , if that will work ?? Thanks for any info Marcin

2 2

Mod_rtmp
by Amit K Nepal 15 Apr '14

15 Apr '14

Hello all, I am new to freeswitch, trying to get the mod_rtmp working. I am able make call from web client (example) to a sip client, but I am not able to call a web user from another web user , it says user not available. Can you please provide any pointers so I can troubleshoot in the right direction ? I haven't made any changes to the dialplan etc. Thank you Amit

2 2

Debian Nightly packages
by Ben Langfeld 15 Apr '14

15 Apr '14

There was at one point talk of building nightly Debian packages of FreeSWITCH. Can anyone confirm what happened to this idea?

2 1

Mod_Managed 4.5.1 crashes
by andretodd＠verizon.net 15 Apr '14

15 Apr '14

1 0

Re: [Freeswitch-users] Snom 370 and Day/Night led
by Mr.Gab 15 Apr '14

15 Apr '14

I'm terribly sorry, i used nabble.com to post the first time but i didn't see the message so i thought there was an error and i sent a direct mail to the users list. I would never dare to demand an immediate response. I read mod_snom before posting but i didn't understand how to do it. I'll continue to try. Best regards crazygabry ---------- Forwarded message ---------- From: Brian West <brian(a)freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Date: Mon, 14 Apr 2014 12:01:38 -0500 Subject: Re: [Freeswitch-users] Snom 370 and Day/Night led Since you sent this yesterday, Sunday @4:11pm for most of us, and you’ve asked the same question again this morning at 9:49am, http://wiki.freeswitch.org/wiki/Mod_snom Please allow people time to actually read, respond to your questions before sending another of the exact same question. -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 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)

2 1

No audio stun for a long time!
by rockxsj 15 Apr '14

15 Apr '14

I can see that sometimes my server product some logs like: deb9157a-b9b1-11e3-8ee1-7b004e0e7c32 2014-04-02 00:03:44.083032 [ERR] switch_rtp.c:749 No audio stun for a long time! 8c9f55cc-b9b8-11e3-9cca-7b004e0e7c32 2014-04-02 00:21:02.441130 [ERR] switch_rtp.c:749 No audio stun for a long time! a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:32:04.021509 [ERR] switch_rtp.c:749 No audio stun for a long time! a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:32:35.021902 [ERR] switch_rtp.c:749 No audio stun for a long time! a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:33:05.061286 [ERR] switch_rtp.c:749 No audio stun for a long time! a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:33:36.061678 [ERR] switch_rtp.c:749 No audio stun for a long time! a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:34:06.101062 [ERR] switch_rtp.c:749 No audio stun for a long time! a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:34:37.101453 [ERR] switch_rtp.c:749 No audio stun for a long time! a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:35:07.121681 [ERR] switch_rtp.c:749 No audio stun for a long time! a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:35:37.161065 [ERR] switch_rtp.c:749 No audio stun for a long time! a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:36:08.161457 [ERR] switch_rtp.c:749 No audio stun for a long time! 74c1f71e-b9fb-11e3-bbbd-7b004e0e7c32 2014-04-02 08:14:35.502605 [ERR] switch_rtp.c:749 No audio stun for a long time! c323e07a-b9fb-11e3-bc46-7b004e0e7c32 2014-04-02 08:15:02.921673 [ERR] switch_rtp.c:749 No audio stun for a long time! fa17e09a-b9fb-11e3-bd18-7b004e0e7c32 2014-04-02 08:16:35.121138 [ERR] switch_rtp.c:749 No audio stun for a long time! fa17e09a-b9fb-11e3-bd18-7b004e0e7c32 2014-04-02 08:17:06.121531 [ERR] switch_rtp.c:749 No audio stun for a long time! 43945544-ba03-11e3-8df9-7b004e0e7c32 2014-04-02 09:09:08.621875 [ERR] switch_rtp.c:749 No audio stun for a long time! 05e67bae-ba09-11e3-a24c-7b004e0e7c32 2014-04-02 09:50:28.321208 [ERR] switch_rtp.c:749 No audio stun for a long time! b151ab62-ba09-11e3-a4da-7b004e0e7c32 2014-04-02 09:58:45.491465 [ERR] switch_rtp.c:749 No audio stun for a long time! 552384de-ba0c-11e3-afc5-7b004e0e7c32 2014-04-02 10:14:03.691235 [ERR] switch_rtp.c:749 No audio stun for a long time! 04bc9112-ba0b-11e3-aabc-7b004e0e7c32 2014-04-02 10:16:39.392036 [ERR] switch_rtp.c:749 No audio stun for a long time! 04bc9112-ba0b-11e3-aabc-7b004e0e7c32 2014-04-02 10:17:37.411787 [ERR] switch_rtp.c:749 No audio stun for a long time! 04bc9112-ba0b-11e3-aabc-7b004e0e7c32 2014-04-02 10:18:07.431169 [ERR] switch_rtp.c:749 No audio stun for a long time! 2e316a40-ba10-11e3-8076-7b004e0e7c32 2014-04-02 10:41:28.251996 [ERR] switch_rtp.c:749 No audio stun for a long time! 7396150e-ba10-11e3-8186-7b004e0e7c32 2014-04-02 10:43:24.291472 [ERR] switch_rtp.c:749 No audio stun for a long time! 388ba6ac-ba12-11e3-8801-7b004e0e7c32 2014-04-02 10:56:48.411568 [ERR] switch_rtp.c:749 No audio stun for a long time! 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:10:06.051664 [ERR] switch_rtp.c:749 No audio stun for a long time! 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:10:36.091046 [ERR] switch_rtp.c:749 No audio stun for a long time! 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:11:06.111419 [ERR] switch_rtp.c:749 No audio stun for a long time! 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:11:37.111820 [ERR] switch_rtp.c:749 No audio stun for a long time! 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:12:07.151202 [ERR] switch_rtp.c:749 No audio stun for a long time! 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:12:38.151594 [ERR] switch_rtp.c:749 No audio stun for a long time! 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:13:08.190977 [ERR] switch_rtp.c:749 No audio stun for a long time! 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:13:39.191370 [ERR] switch_rtp.c:749 No audio stun for a long time! 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:14:10.191752 [ERR] switch_rtp.c:749 No audio stun for a long time! abb38326-ba16-11e3-9cb0-7b004e0e7c32 2014-04-02 11:27:40.391902 [ERR] switch_rtp.c:749 No audio stun for a long time! 75cee1e2-ba1b-11e3-b14f-7b004e0e7c32 2014-04-02 12:01:57.731915 [ERR] switch_rtp.c:749 No audio stun for a long time! My server is not behind firewall or router, I want know why the freeswitch still product these logs? Is something wong with my config?

4 8

Homer SIP capture integration
by Peter Steinbach 15 Apr '14

15 Apr '14

Hello, I was following http://code.google.com/p/homer/wiki/HOWTO#Capture_Agent to have "Homer Sip capture" capturing an displaying a SIP Packets from freeswitch. This worked quite well on a Debian based system. But now we migrated it to Centos 6 and it does not work anymore, although we used the identical kamailio source code base. I think the underlying xml parser libs somehow work differently on Centos. What is puzzling me is the following Apr 4 10:33:12 homer2centos /usr/local/sbin/kamailio[20698]: ERROR: <core> [parser/msg_parser.c:705]: parse_msg(): ERROR: parse_msg: message=<#001#020#002#021#023a#023aaaaaaaSIP/2.0 480 Temporarily You can see that in front of the SIP message coming from freeswitch there are some additional chars "#001#020#002#021#023a#023aaaaaaa". And by wiresharking the SIP packages, Freeswitch in fact preceedes the SIP message it sends to homer with these additional bytes. Using a Debian based Kamailio this does not cause problems on the Kamailio/Homer side, buit on Centos it does. So my question is: Is this the right behaviour of Freeswitch when forking the SIP message to the capture server? In https://groups.google.com/forum/#!topic/homer-discuss/KhBF0Dy44SU I found the following >The problem was found: freeswitch duplicate all packets on SIP port including empty "ping" messages. >with mobile regards, >Alexandr Although I do not really understand the part "including empty ping messages": Is there a way to suppress these ping messages? Here is a complete error message: Apr 4 10:33:12 homer2centos /usr/local/sbin/kamailio[20698]: INFO: <core> [parser/parse_fline.c:229]: parse_first_line(): ERROR:parse_first_line: bad request first line Apr 4 10:33:12 homer2centos /usr/local/sbin/kamailio[20698]: INFO: <core> [parser/parse_fline.c:231]: parse_first_line(): ERROR: at line 0 char 39: Apr 4 10:33:12 homer2centos /usr/local/sbin/kamailio[20698]: INFO: <core> [parser/parse_fline.c:237]: parse_first_line(): ERROR: parsed so far: #001#020#002#021#023a#023aaaaaaaSIP/2.0 480 Temporarily Apr 4 10:33:12 homer2centos /usr/local/sbin/kamailio[20698]: INFO: <core> [parser/parse_fline.c:242]: parse_first_line(): ERROR:parse_first_line: bad message Apr 4 10:33:12 homer2centos /usr/local/sbin/kamailio[20698]: ERROR: <core> [parser/msg_parser.c:705]: parse_msg(): ERROR: parse_msg: message=<#001#020#002#021#023a#023aaaaaaaSIP/2.0 480 Temporarily Unavailable#015#012Via: SIP/2.0/UDP 192.168.178.204:5060;branch=z9hG4bK6569ec1b;rport=5060#015#012Max-Forwards: 70#015#012From: "06081688533" <sip:06081688533@192.168.178.204>;tag=as190e467c#015#012To: <sip:4488329@192.168.178.220:5090>;tag=1D8c28jZaF72r#015#012Call-ID: 17888b4f3812b7686fbaf0ca380be672(a)192.168.178.204#015#012CSeq: 102 INVITE#015#012User-Agent: FreeSWITCH-mod_sofia/1.5.11b+git~20140305T111532Z~783a4083ed~64bit#015#012Accept: application/sdp#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY#015#012Supported: timer, path, replaces#015#012Allow-Events: talk, hold, conference, refer#015#012Reason: Q.850;cause=16;text="NORMAL_CLEARING"#015#012Content-Length: 0#015#012Remote-Party-ID: "4488329" <sip:4488329@192.168.178.220>;party=calling;privacy=off;screen=no#015#012#015#012> Apr 4 10:33:12 homer2centos /usr/local/sbin/kamailio[20698]: ERROR: <core> [receive.c:146]: receive_msg(): ERROR: receive_msg: parse_msg failed -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

2 3

NDLB-connectile-dysfunction=true with a non directory user
by Jon Schøpzinsky 15 Apr '14

15 Apr '14

Hello list, We are having some problems with customer behind nat, where freeswitch is responding to the contact headers address, which fails as the client is using its internal IP and port information. We tried setting the NDLB-connectile-dysfunction=true in the dialplan, but it doesnt seem to work. Which leg should we set it on? And any reason why it wouldnt work? Kind Regards Jon Schøpzinsky

3 3