August 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch dialplan issue
by Nick Zaitsev 20 Aug '14

20 Aug '14

Good day to you. I have strange issue. I would like to set the ext in dialplan in that way: if there is local registered ext in freeswitch fs should route the call there,else route the call outside, i wrote this in dialplan: <extension name="fs3_outbound"> <condition field="destination_number" expression="^([123][012345]\d\d\d)$"> <action application="set" data="ext=$1"/> <action application="set"data="domain_name=${domain}"/> </condition> <condition field="${sofia_contact(*/${ext}@${domain_name})}" expression="error/user_not_registered"> <action application="bridge" data="sofia/internal/${ext}(a)10.160.1.21"/> <anti-action application="bridge" data="user/${ext}@${domain_name}"/> </condition> </extension> But it seems,that anti-action is not match at all( Thank you for your time,best regards. -- Nick Zaitsev

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Re: [Freeswitch-users] Freeswitch srtp rejecting SRTP with 488
by dodul＠live.com 20 Aug '14

20 Aug '14

Hello, Thanks for your reply. I'm sorry I guess didn't get you. Are you saying the log I provided in my earlier email is incomplete? Or I have to check some other log? If that is so can you please clarify which log? (Eg log name) Or it is there in the log I provided, but I have ignored the part I needed to go through? If that's the case can you please indicate which part? Thanks in advanced :) Sent from my “contract free” BlackBerry® smartphone on the WIND network. -----Original Message----- From: freeswitch-users-request(a)lists.freeswitch.org Sender: freeswitch-users-bounces(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 10:06:25 To: <freeswitch-users(a)lists.freeswitch.org> Reply-To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 98, Issue 128 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..."

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Authenticating a user through ESL script
by sami sakly 20 Aug '14

20 Aug '14

Hi, I'm looking to issue calls from the CLI (using originate command ) the same way it's done through any Dialer like SIPml5. I think I need to do something about the dialplan because once I'm calling using SIPml5 I'm passing "through" default dialplan and when I'm only using originate from external script I'm passing "through" Public dialplan. May be something about authenticated user ? 2014-08-18 18:20:37.712649 [INFO] mod_dialplan_xml.c:558 Processing <0000000000>->1000 in context public for the originate command I'm using esl.php : $ec1 = $sock->bgapi("originate","{origination_uuid=$uuid1}sofia/internal/ 1000(a)12.0.0.104 &park()"); Thanks for your help. Regards,

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mod_spidermonkey / freeswitch-v8.x86_64
by Mujahid Ali 20 Aug '14

20 Aug '14

when i install freeswitch-v8.x86_64 by yum like yum install freeswitch-v8.x86_64 it install but there is no file mod_v8.so in /usr/lib64/freeswitch/mod/ even else every thing is OK is it completely install or not ?

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Registering with digest authentication, but empty password. How?
by Bob Hartwig 20 Aug '14

20 Aug '14

Hi all, This is going to sound like a weird problem. I'm trying to test a SIP trunk from a vendor that requires registering with digest authentication, but an empty password, and I don't know how to specify the empty password in my profile. If I configure my sofia profile like this "<param name="password" value="boguspassword"/>", the vendor gives me a 407, so I know that they need that empty password. If I configure it like this: "<param name="password" value=""/>", Freeswitch complains that "[ERR] sofia.c:2703 ERROR: password param is REQUIRED!". So is there any way to configure the profile with an empty password? Thanks, Bob Hartwig

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Freeswitch srtp rejecting SRTP with 488
by Kamrul Khan 19 Aug '14

19 Aug '14

Hi, Below is my dialplan: <extension name="cext"> <condition field="destination_number" expression="^(2000)$"> <action application="set" data="rtp_secure_media=true"/> <action application="bridge" data="user/bob"/> </condition> <condition field="destination_number" expression="^(3000)$"> <action application="set" data="rtp_secure_media=true"/> <action application="bridge" data="user/bob"/> </condition> </extension></context> You can see I have rtp_secure_media=true for both extension. However, when I try to call I get rejected with 488 Not acceptable here. In my freeswitch console log I find sofia/internal/alice(a)192.168.146.133 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION. What should I do to fix this? Below is my complete console log: 2014-07-22 19:51:32.572499 [DEBUG] sofia.c:5847 IP 192.168.146.133 Rejected by acl "domains". Falling back to Digest auth.2014-07-22 19:51:32.707332 [DEBUG] sofia.c:5847 IP 192.168.146.133 Rejected by acl "domains". Falling back to Digest auth.2014-07-22 19:51:32.724263 [NOTICE] switch_channel.c:669 New Channel sofia/internal/alice(a)192.168.146.133 [4101d950-1214-11e4-933c-61443ed3c9c7]2014-07-22 19:51:32.725481 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_NEW2014-07-22 19:51:32.725481 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/alice(a)192.168.146.133) State NEW2014-07-22 19:51:32.762327 [DEBUG] sofia.c:4153 Channel sofia/internal/alice(a)192.168.146.133 entering state [received][100]2014-07-22 19:51:32.762327 [DEBUG] sofia.c:4164 Remote SDP:v=0o=Mozilla-SIPUA-31.0 12934 1 IN IP4 0.0.0.0s=Doubango Telecom - firefoxt=0 0a=ice-ufrag:5e3a0151a=ice-pwd:9df6e8f0963f033e8573a23e4ce62298a=fingerprint:sha-256 90:BD:D7:9C:DE:66:5D:86:D4:8A:61:89:75:07:EF:F9:9A:06:AE:29:C8:1D:BD:C3:BD:C0:41:12:01:B5:45:FDm=audio 56772 UDP/TLS/RTP/SAVPF 109 0 8 101c=IN IP4 184.69.59.132a=rtpmap:109 opus/48000/2a=ptime:20a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-levela=setup:actpassa=candidate:0 1 UDP 2128609535 172.16.1.188 56772 typ hosta=candidate:2 1 UDP 1692467199 184.69.59.132 56772 typ srflx raddr 172.16.1.188 rport 56772a=candidate:5 1 UDP 2128543999 192.168.56.1 56773 typ hosta=candidate:10 1 UDP 2128478463 192.168.232.1 56774 typ hosta=candidate:15 1 UDP 2128412927 192.168.146.1 56775 typ hosta=candidate:0 2 UDP 2128609534 172.16.1.188 56776 typ hosta=candidate:1 2 UDP 1692467198 184.69.59.132 56776 typ srflx raddr 172.16.1.188 rport 56776a=candidate:5 2 UDP 2128543998 192.168.56.1 56777 typ hosta=candidate:10 2 UDP 2128478462 192.168.232.1 56778 typ hosta=candidate:15 2 UDP 2128412926 192.168.146.1 56779 typ hosta=rtcp-mux 2014-07-22 19:51:32.762327 [NOTICE] sofia.c:4354 Hangup sofia/internal/alice(a)192.168.146.133 [CS_NEW] [INCOMPATIBLE_DESTINATION]2014-07-22 19:51:32.762327 [DEBUG] switch_channel.c:2102 Send signal sofia/internal/alice(a)192.168.146.133 [KILL]2014-07-22 19:51:32.762327 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK]2014-07-22 19:51:32.762327 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_HANGUP2014-07-22 19:51:32.762327 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/alice(a)192.168.146.133) State HANGUP2014-07-22 19:51:32.762327 [DEBUG] mod_sofia.c:414 Channel sofia/internal/alice(a)192.168.146.133 hanging up, cause: INCOMPATIBLE_DESTINATION2014-07-22 19:51:32.780344 [DEBUG] mod_sofia.c:476 Responding to INVITE with: 4882014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:46 sofia/internal/alice(a)192.168.146.133 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/alice(a)192.168.146.133) State HANGUP going to sleep2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/alice(a)192.168.146.133) State Change CS_HANGUP -> CS_REPORTING2014-07-22 19:51:32.780344 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK]2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_REPORTING2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/alice(a)192.168.146.133) State REPORTING2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:53 sofia/internal/alice(a)192.168.146.133 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/alice(a)192.168.146.133) State REPORTING going to sleep2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/alice(a)192.168.146.133) State Change CS_REPORTING -> CS_DESTROY2014-07-22 19:51:32.780344 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK]2014-07-22 19:51:32.780344 [DEBUG] switch_core_session.c:1164 Session 138 (sofia/internal/alice(a)192.168.146.133) Locked, Waiting on external entities2014-07-22 19:51:32.780344 [NOTICE] switch_core_session.c:1182 Session 138 (sofia/internal/alice(a)192.168.146.133) Ended2014-07-22 19:51:32.780344 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/alice(a)192.168.146.133 [CS_DESTROY]2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_DESTROY2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/alice(a)192.168.146.133) State DESTROY2014-07-22 19:51:32.780344 [DEBUG] mod_sofia.c:341 sofia/internal/alice(a)192.168.146.133 SOFIA DESTROY2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:60 sofia/internal/alice(a)192.168.146.133 Standard DESTROY2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/alice(a)192.168.146.133) State DESTROY going to sleep

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[ANN] openerp-freeswitch-connector
by Trever L. Adams 19 Aug '14

19 Aug '14

Hello everyone, I am announcing this here in hopes that some of you may find this useful. I have forked http://www.akretion.com/open-source-contributions/openerp-asterisk-voip-con… and made it work with FreeSWITCH. At the moment it works for me. The only feature currently working is click to dial under customers. I will be implementing all of the features found in the original over the coming weeks. I will also be adding some that they do not have. Anyone, of course, is free to use this under the terms of the license. Anyone wishing to help out with documentation, code, features, what not, you are also welcome. The repository can be found at https://github.com/treveradams/openerp-freeswitch-connector Thank you, Trever

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group_confirm with inband DTMF detection
by Vladimir Ralev 19 Aug '14

19 Aug '14

Hi all, I want to enable FS to accept inband DTMF with the group confirmation. I tried to do start_dtmf action on the call legs, however that action is only executed after the confirmation, so the actual confirmation requires an out of band DTMF. Here is one example that demonstrates the issue: originate {group_confirm_file=/tmp/ivr-group_confirm.wav,group_confirm_key=1,call_timeout=60}sofia/sipinterface/number@provider &start_dtmf() Any ideas? Ideally I want to enable system-wide inband DTMF detection in a safe way at some CPU cost.

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FS-HA RTP state
by Alex Balashov 18 Aug '14

18 Aug '14

Hi, Sorry for the newbie question. Regarding this: https://wiki.freeswitch.org/wiki/Freeswitch_HA How does FreeSWITCH pick up and resume RTP stream state in this scenario, vis-a-vis the RTP sequence numbers? Or does it not? I don't really know how RTP works through the FreeSWITCH core. Based on the fact that this type of failover works, I am guessing that Freeswitch has several modes of RTP relay in a channel bridge: (1) Where the RTP is regenerated/reoriginated with logically independent SSRC and sequence numbers, which would presumably be used in situations where invasive DSP work, transcoding and/or other heavy media involvement is required of the FreeSWITCH core. I assume this involves marshalling the audio payload onto some sort of generic internal audio frame bus, too. (2) "Light" mode, analogous to Asterisk's "Packet2Packet" bridging, where the RTP is relayed more or less as-is between the endpoints, without this type of mediation. This would presumably pass through the endpoints' respective RTP stream properties. Is it reasonable to assume that the call recovery functionality relies on #2, which would imply that the calls cannot be transcoded, use in-band DTMF signalling, etc? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

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transcoding error fs 1.2.11
by Jason Holden 18 Aug '14

18 Aug '14

All. I have my endpoints setup to use PCMU but FS is still stating its transcoding the call. Also have the codec list setup to greedy. Fs 1.2.11 git version aa68eac Any suggestions on how to proceed?

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