February 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP REFER - URI Parameters
by Phil Dunks 28 Feb '14

28 Feb '14

Hi Guys I’m trying to use FS (1.5.8b from last week) as an SBC and load-balancer (using mod_distributor). It is load balancing between several FS conference nodes. It’s working a treat, apart from one scenario : Say conference xyz is already running on conf-node-1. Next incoming call is sent to conf-node-2, pin is collected, and conference xyz is identified as the required conference. Conf-node-2 checks presence, and sees the conference is already running on conf-node-1, and does a deflect. We add some query params to the refer-to URI to tell conf-node-1 not to collect the pin, but put the caller straight into the conference. e.g <action application="deflect" data=“sip:1234@10.1.1.2;conf=xyz" /> In the REFER I can see the params : Refer-To: <sip:1234@10.1.1.2;conf=xyz> But they are not present on the request URI or To header in the INVITE to conf-node-1. So conf-node-1 requests the PIN again. Is there any way I can pass this info to the other conference node by using the REFER? Thanks for your help. Phil

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Week in Review Feb 16-23
by Kathleen King 27 Feb '14

27 Feb '14

Hello, again. This week in the FreeSWITCH master branch we had heck of a lot of activity with 89 commits and most of those commits were touching BSD and libsndfile. And we also saw new additions to language support with the form of adding Swedish to mod_say and improvements to Spanish phrase macros. *The following bugs were squashed:* c945584 FS-6237 --resolve --fixed a regression from December in inherit_codec --http://jira.freeswitch.org/browse/FS-6237 81f9303 FS-6219 --resolve --bug: update mod_cidlookup to properly handle unknown and unavailable --http://jira.freeswitch.org/browse/FS-6219 *New features that were added:* 0521493 Add man page for freeswitch --Added a man page for FreeSWITCH 9d30007 FS-6231 add support for sending call waiting tone when a simultaneous call is received --resolve --feature: added support for sending call waiting tone when a simultaneous call is received in mod_skinny --http://jira.freeswitch.org/browse/FS-6231 1c3001f FS-2999 --resolve --misc: in spanish phrase macros fixed some translations and added others --http://jira.freeswitch.org/browse/FS-2999 97e2340 FS-6204 refactor some dtls code to generate a 1024 bit cert suitable for dtls in all version of chrome --feature: updated FreeSWITCH_core to support new Chrome DTLS requirements --http://jira.freeswitch.org/browse/FS-6204 bd72cd9 Allow setting late margin on register expirations --feature: in mod_sofia added parameter to allow for configuration of register expiration late margin --Thanks, Emmanuel Schmidbauer. 19dbd07 Allow access to headers from INVITE --feature: copies all INVITE headers to channel variables on the a leg for inbound calls --http://jira.freeswitch.org/browse/FS-6075 --Thanks, Peter Olsson. Libsndfile was updated from a version from March 3rd 2009 to a version from April 5th 2013. *Improvements in cross platform build supports:* e209b1a left line in that breaks mod_dingaling build update and run sh config.status from build root or do a clean build to correct --build: fixed build issue that broke mod_dingaling requires a clean build to correct f206700 FS-6189 --resolve --build: fixed compilation issue in Ubuntu 14.04 --http://jira.freeswitch.org/browse/FS-6189 There were many commits for FreeBSD and OpenBSD build support. *In terms of stability these were the use cases that were fixed:* f826b4b FS-5845 --resolve --fixed a memory leak in mod_lcr related to switch_channel_expand_variables --http://jira.freeswitch.org/browse/FS-5845 Feedback welcome and the referenced commits are in the attached text file with 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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Help with gateway audio
by Normando Hall 27 Feb '14

27 Feb '14

Hello all. I am triying to use FS (git) with an external gateway. This is the schema: FS (has public IP over virtual interface eth0:1 - xxx.xxx.xxx.xxx) -> INTERNET -> Router TP-LINK (with ports forwarded to GW) -> HT503 (192.168.0.160). HT503 has two sip accounts. One for an internal extension (FXS) and one for PSTN (FXO). I can make calls from an internet softphone to FXS and the phone attached rings, but when I pecked up I can't hear audio in both directions. The same happens when I call FXO to make a PSNT call. The PSTN calls is ok but no audio in both directions. BUT, if I picked up the internal phone attached to FXS and make a call to an internal extension o calls to 5000 I hear the IVR and audio flows in both directions. Because a rule in HT503 on FXO port, when receive a call from PSTN this forwarding to 1015 extension. This works and I hear the audio in both directions. FS in running on a Centos 6 VPS. Are there a way to debug this? Some lights? Thank you Normando

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NUA issue.... Freeswitch down
by Michal Borkowski - MORCOM 27 Feb '14

27 Feb '14

Hi, I'm not sure why - but is is happening that my Freeswitch is going down with the error: 2014-02-27 11:01:50.289493 [DEBUG] sofia.c:1495 nua_i_outbound: unknown event 8: 101 NAT detected What can cause the issue? Maybe somebody had this issue? I was trying to find - but can not... Thanks in advance for help Michal

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Say Application ESL Error Reporting
by Callum Guy 27 Feb '14

27 Feb '14

Hi All, Simple question - i'm using "say" application over ESL to read back an iterated list of numbers however there is a chance of pollution in the variable. I can of course cleanse this myself in the application logic however the current application result reporting seems incorrect. Here is my result if I'm asking it to read 1234: [variable_current_application_data] => en NUMBER iterated 4448 [variable_current_application] => say [variable_sound_prefix] => /usr/share/sounds/en/us/callie [Application] => say [Application-Data] => en NUMBER iterated 4448 [Application-Response] => _none_ As expected this reads "4, 4, 4, 8" Here is my result with pollution (i do not have a prompt for "star") [variable_current_application_data] => en NUMBER iterated 4**5 [variable_current_application] => say [variable_sound_prefix] => /usr/share/sounds/en/us/callie [Application] => say [Application-Data] => en NUMBER iterated 4**5 [Application-Response] => _none_ This makes the application fail immediately and does not even play the first 4 - so its failing in the preparation. In the CLI i can see the following line for the failed say session: 2014-02-24 11:10:06.079062 [ERR] mod_say_en.c:123 Parse Error! There does not seem to be any other variable to indicate the failure of this application. Perhaps "Application-Response" is not the correct place to look? If so it'd be useful to know what this variable *IS* used for! If anyone is able to provide some details on how best to capture an error like this remotely please let me know. Thanks, Callum _____________________________ Callum Guy Senior Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy(a)x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email.

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park_after_bridge
by Alex Lake 27 Feb '14

27 Feb '14

I have a query relating to the "park_after_bridge" channel variable: Firstly, I find the definition in the wiki a little vague - could do with an example (which I will provide once I'm sure I understand what it's supposed to do) If on an a-leg, I initiate a b-leg by executing bridge with park_after_bridge=true, the idea is that the a-leg could terminate and the b-leg stays connected. How about if I initiate a b-leg without park_after_bridge, but then while the calls are connected, set park_after_bridge=true (a b-leg channel variable) then when a-leg terminates, will b-leg still be parked?

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Calls Authentication
by Lloyd Aloysius 27 Feb '14

27 Feb '14

Hello All, I have the the following set in my profile parameters. challenge-realm = false auth-all-packets = true But someone from the otuside can enter into my dial plan. Why the switch is not blocking(authenticate) before enter into the dialpaln. I do not have any default context or user 1001 in the system. please see the log below. What i am doing wrong in my setup. ========================= 2014-02-27 07:07:16.449120 [DEBUG] sofia.c:7868 sofia/sipinterface_1/1001@{freeswitch_ip} receiving invite from {originate_ip}:10036 version: 1.2.22 git aabf813 2014-02-24 01:22:36Z 64bit 2014-02-27 07:07:16.449120 [DEBUG] sofia.c:8025 IP {originate_ip} Rejected by acl "domains". Falling back to Digest auth. *2014-02-27 07:07:16.449120 [WARNING] sofia_reg.c:1532 SIP auth challenge (INVITE) on sofia profile 'sipinterface_1' for [018584290411@{freeswitch_ip}] from ip {originate_ip}* 2014-02-27 07:07:16.449120 [DEBUG] switch_core_state_machine.c:434 (sofia/sipinterface_1/1001@{freeswitch_ip}) State NEW 2014-02-27 07:07:16.449120 [DEBUG] switch_core_session.c:1016 Send signal sofia/sipinterface_1/1001@{freeswitch_ip} [BREAK] 2014-02-27 07:07:16.449120 [DEBUG] sofia.c:1792 detaching session b31c0e64-9fa7-11e3-8ba7-6d0b1a3b8605 2014-02-27 07:07:16.809118 [DEBUG] sofia.c:1884 Re-attaching to session b31c0e64-9fa7-11e3-8ba7-6d0b1a3b8605 2014-02-27 07:07:16.809118 [DEBUG] switch_core_session.c:1016 Send signal sofia/sipinterface_1/1001@{freeswitch_ip} [BREAK] 2014-02-27 07:07:16.809118 [DEBUG] switch_core_session.c:1016 Send signal sofia/sipinterface_1/1001@{freeswitch_ip} [BREAK] 2014-02-27 07:07:16.809118 [DEBUG] sofia.c:7868 sofia/sipinterface_1/1001@{freeswitch_ip} receiving invite from {originate_ip}:10036 version: 1.2.22 git aabf813 2014-02-24 01:22:36Z 64bit *2014-02-27 07:07:16.809118 [DEBUG] sofia.c:8025 IP {originate_ip} Rejected by acl "domains". Falling back to Digest auth.* 2014-02-27 07:07:16.829310 [DEBUG] sofia.c:5811 Channel sofia/sipinterface_1/1001@{freeswitch_ip} entering state [received][100] 2014-02-27 07:07:16.829310 [DEBUG] sofia.c:5824 Remote SDP: v=0 o=- 5753281 5753289 IN IP4 {originate_ip} s=eyeBeam c=IN IP4 192.168.1.100 t=0 0 m=audio 11424 RTP/AVP 100 6 0 8 3 18 97 5 101 a=rtpmap:100 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : 86FF30D0 0000006D 192.168.1.100 24006 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [speex:100:16000:20:0]/[G722:9:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [speex:100:16000:20:0]/[PCMU:0:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [speex:100:16000:20:0]/[PCMA:8:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [speex:100:16000:20:0]/[GSM:3:8000:20:13200] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [DVI4:6:16000:20:0]/[G722:9:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [DVI4:6:16000:20:0]/[PCMU:0:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [DVI4:6:16000:20:0]/[PCMA:8:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [DVI4:6:16000:20:0]/[GSM:3:8000:20:13200] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:3190 Set Codec sofia/sipinterface_1/1001@{freeswitch_ip} PCMU/8000 20 ms 160 samples 64000 bits 2014-02-27 07:07:16.829310 [DEBUG] switch_core_codec.c:111 sofia/sipinterface_1/1001@{freeswitch_ip} Original read codec set to PCMU:0 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5451 Set 2833 dtmf send/recv payload to 101 2014-02-27 07:07:16.829310 [DEBUG] sofia.c:6091 (sofia/sipinterface_1/1001@{freeswitch_ip}) State Change CS_NEW -> CS_INIT 2014-02-27 07:07:16.829310 [DEBUG] switch_core_session.c:1351 Send signal sofia/sipinterface_1/1001@{freeswitch_ip} [BREAK] 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_1/1001@{freeswitch_ip}) Running State Change CS_INIT 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:455 (sofia/sipinterface_1/1001@{freeswitch_ip}) State INIT 2014-02-27 07:07:16.829310 [DEBUG] mod_sofia.c:87 sofia/sipinterface_1/1001@{freeswitch_ip} SOFIA INIT 2014-02-27 07:07:16.829310 [DEBUG] mod_sofia.c:127 (sofia/sipinterface_1/1001@{freeswitch_ip}) State Change CS_INIT -> CS_ROUTING 2014-02-27 07:07:16.829310 [DEBUG] switch_core_session.c:1351 Send signal sofia/sipinterface_1/1001@{freeswitch_ip} [BREAK] 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:455 (sofia/sipinterface_1/1001@{freeswitch_ip}) State INIT going to sleep 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_1/1001@{freeswitch_ip}) Running State Change CS_ROUTING 2014-02-27 07:07:16.829310 [DEBUG] switch_channel.c:2163 (sofia/sipinterface_1/1001@{freeswitch_ip}) Callstate Change DOWN -> RINGING 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:471 (sofia/sipinterface_1/1001@{freeswitch_ip}) State ROUTING 2014-02-27 07:07:16.829310 [DEBUG] mod_sofia.c:150 sofia/sipinterface_1/1001@{freeswitch_ip} SOFIA ROUTING 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:117 sofia/sipinterface_1/1001@{freeswitch_ip} Standard ROUTING *2014-02-27 07:07:16.829310 [INFO] mod_dialplan_xml.c:558 Processing 1001 <1001>->018584290411 in context default* Thanks and regards, Lloud

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Jira has been upgraded.
by Ken Rice 27 Feb '14

27 Feb '14

Hey Guys, Jira.freeswitch.org has been upgraded. Let me know if you run across any issues with it. You¹ll notice quite a few changes. We¹re also upgrading and updating a few other systems. So watch out for brief outages on other systems. K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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fax_header & fax_ident length
by Tihomir Culjaga 26 Feb '14

26 Feb '14

hello every1 just a small question ... what is the max length of fax_header and fax_ident. from spandsp source i have found its 50 characters for fax_header and 22 for ident... but im having issues going over 26 characters for header. The fax simply doesn't go through... so im wondering if there is a limit on the T.30 protocol for the header. anyone ?

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Significant audio clipping when bridging webrtc to sip
by Joegen Baclor 26 Feb '14

26 Feb '14

Hi, I am using FreeSwitch 1.4-beta to bridge WebRTC->SIP. All is working great except for the fact that audio is almost always clipped at the beginning of the call. Looking at the logs, it seems to me that this is due to the delay incurred when performing the DTLS handshake on receipt of media. I am doing a "pre_answer" before bridging. My question is, is it possible to do the DTLS handshake prior to originate and right after pre_answer? Log is in pastebin: http://pastebin.freeswitch.org/22037. The dropped media happens during the gap between lines 310 and 311. Thanks for any pointers. Joegen

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