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weird behavior when originating call with 3cx
by Sayyed Mohammad Emami Razavi 23 Jun '13

23 Jun '13

When i send call to one user that register with 3cx, 3cx response unalocate number. I can listen to IVR with 3cx when call IVR extension but when i want to originate one call with user at 3cx i can not. All tcp logs attached and FS logs are here: http://pastebin.freeswitch.org/21095

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opensips route one registration to two FS
by ZhouJianhua 22 Jun '13

22 Jun '13

Hi all, OpenSIPs is used for load balancing in from of two FS. And in some cases, there are more than one item inFS's "sip_registrations" table for one sip client. Then, the execution of "originate sofia/internal/SIP_ACCOUNT%FS_IP", will fail if FS_IP is wrong. I use the "expires" field in table of "sip_registrations" to figure out which is the newest FS_IP for the sip client, but sometimes it fails. I use the config file from http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSW… Any suggestions will be appreciated!

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really weird 2 extension number being processed in different context.
by Vincent Xia 22 Jun '13

22 Jun '13

i have extension 1003 and 1006 registering to FS, both with the default 1003.xml and 1006.xml, but when dial any number (e.g. 3000 for a conference call), the 1003 is processed by FS in context public while 1006 in context default, 2013-06-21 10:59:01.149804 [INFO] mod_dialplan_xml.c:557 Processing 1003 <1003>->3000 in context public 2013-06-21 10:58:31.169161 [INFO] mod_dialplan_xml.c:557 Processing Ext.1006 <1006>->3000 in context default im pretty sure the following param is the same for 1003 and 1006, they are from the FS tarball, no one has ever modified it: <variable name="user_context" value="default"/> really weird...

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Collecting DTMF digits with mod_httapi
by Rafal Gwizdala 22 Jun '13

22 Jun '13

Hi, could you give me some example of what's necessary to collect DTMF digits during audio file playback and how to return collected data to my web application? I'm sorry but even after reading FS 1.2 book I don't know how to do that. Here's my application's httapi script: <document type="text/freeswitch-httapi"> <work> <pause milliseconds="2000" /> <playback file="http://wypas2/simver/audio/test2.wav" name="digits" digit-timeout="5" input-timeout="15" loops="3" terminators="#"> <bind strip="#">~\\d{3}</bind > </playback> </work> </document> but i don't see the collected digits anywhere in request parameters coming from FS. Thanks RG

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Playback sound file
by Ivan C Myrvold 22 Jun '13

22 Jun '13

I am using mod_xml_curl to control the dialplan of FreeSWITCH. When a user is dialing a number he is not allowed to call (111), I want FreeSWITCH to play back the phone_not_auth.wav sound file. I am returning the following XML. <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan"> <context name="default"> <extension name=no_name> <condition> <action application="playback" data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/phone_not_auth.wav"/> </condition> </extension> </context> </section> </document> Here is the log from FreeSWITCH. I don't hear the sound that is played back, just the busy signal after a couple seconds of silence. What is wrong here? Ivan 2013-06-21 11:51:54.690026 [NOTICE] switch_channel.c:978 New Channel sofia/internal/160(a)178.79.177.164 [f803e92a-da68-11e2-993c-1d15db576d4a] 2013-06-21 11:51:55.369954 [INFO] mod_dialplan_xml.c:558 Processing Linksys 160 <160>->111 in context default 2013-06-21 11:51:56.129953 [INFO] switch_core_session.c:2578 Sending early media 2013-06-21 11:51:56.129953 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/160(a)178.79.177.164! 2013-06-21 11:51:58.849958 [NOTICE] switch_core_state_machine.c:262 sofia/internal/160(a)178.79.177.164 has executed the last dialplan instruction, hanging up. 2013-06-21 11:51:58.849958 [NOTICE] switch_core_state_machine.c:264 Hangup sofia/internal/160(a)178.79.177.164 [CS_EXECUTE] [NORMAL_CLEARING] 2013-06-21 11:51:58.849958 [NOTICE] switch_core_session.c:1566 Session 207 (sofia/internal/160(a)178.79.177.164) Ended 2013-06-21 11:51:58.849958 [NOTICE] switch_core_session.c:1570 Close Channel sofia/internal/160(a)178.79.177.164 [CS_DESTROY]

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is it possible to have FS bind to certain type of softphone?
by Vincent Xia 22 Jun '13

22 Jun '13

normally a SIP REGISTER request will contain information about the terminal like: User-Agent: X-LITE 3.0 so is it possible to have FS reject all softphone registering requests except, for example, X-LITE, without changing FS source code? BR

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how to pass the original caller number through FS gateway?
by Vincent Xia 22 Jun '13

22 Jun '13

i have 2 FS server connected, FS A reg to FS B, and FS B is a gateway for FS A, the external conf of FS A is <gateway name="gwB"> <param name="username" value="1000"/> <param name="password" value="1234"/> <param name="realm" value="192.168.0.100"/> <param name="proxy" value="192.168.0.100"/> </gateway> and i can make call from a phone(e.g. 1001) regestring in FS A to one(e.g. 1005) in FS B, but the calling number shown on the called party is "1000", how can i pass the original caller number("1001") to the called party in FS B?

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Odp: Re: Playback sound file
by Mariusz Czulada 21 Jun '13

21 Jun '13

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Odp: Re: [Freeswitch-dev] Github and FreeSWITCH
by Mariusz Czulada 21 Jun '13

21 Jun '13

On our wiki there is a section: http://wiki.freeswitch.org/wiki/Git_Tips#Github_Workflow I take this information as a sort of incentive to use this alternative way to publish patches. Perhaps this section should be removed? Regards, Mariusz Dnia Piątek, 21 Czerwca 2013 15:22 Brian West <brian(a)freeswitch.org> napisał(a) > I will investigate that... just wanted to make sure everyone was aware we use Jira and our wiki so we don't have fragmentation. > > On Jun 21, 2013, at 8:13 AM, Mathieu Parent <math.parent(a)gmail.com> wrote: > > > Hi, > > > > 2013/6/21 Brian West <brian(a)freeswitch.org>: > >> Please use http://jira.freeswitch.org to publish patches. > > > > Then, why don't you simply remove the ability to request pull/fill > > issues/update wiki? > > > > Regards > > -- > > Mathieu Parent > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.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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voice quality problem when using webrtc client
by Jayesh Nambiar 21 Jun '13

21 Jun '13

Hello All, I am using SIPML5 webrtc client with Kamailio as websocket and SIP Proxy and sending calls to Freeswitch to transcode call from opus to either 711 or 729 when a calls goes towards a pstn network. The two-way voice now works fine in the latest version of freeswitch master branch after ICE support was included in it. But the only problem is that while talking; the voice goes off for a second in between and this happens every 15 seconds or so. This behavior is observed on both legs of the call i.e. the browser as well as the PSTN endpoint !! I try calling from other SIP clients like CsipSimple and Xlite, the voice is consistent using Opus or 711; but voice from SIPML5 is always getting 1 second blanks in the middle of the conversation. I thought this might be because of VAD or Silence Suppression and added the following parameter in my SIP profile: <param name="suppress-cng" value="true"/> but still the quality of call from SIPML5 is the same. I also tried enabling only PCMU in inbound codec preference in the SIP Profile but still the voice quality is same. This is the SDP that I get from SIPML5: v=0 o=- 3138457300 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS l60QBLHC784xSyOLT5kAQYU4AG7U0ZwWyjcQ m=audio 52573 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126 c=IN IP4 192.168.43.167 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 CN/48000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=rtcp:52573 IN IP4 192.168.43.167 a=candidate:2649802508 1 udp 2113937151 192.168.43.167 52573 typ host generation 0 a=candidate:2649802508 2 udp 2113937151 192.168.43.167 52573 typ host generation 0 a=candidate:1368551212 1 udp 2113937151 203.144.192.136 57209 typ host generation 0 a=candidate:1368551212 2 udp 2113937151 203.144.192.136 57209 typ host generation 0 a=candidate:3547544572 1 tcp 1509957375 192.168.43.167 61288 typ host generation 0 a=candidate:3547544572 2 tcp 1509957375 192.168.43.167 61288 typ host generation 0 a=candidate:521245660 1 tcp 1509957375 203.144.192.136 61289 typ host generation 0 a=candidate:521245660 2 tcp 1509957375 203.144.192.136 61289 typ host generation 0 a=ice-ufrag:aD0ElKq0auO0/wUZ a=ice-pwd:4yT6f2XYrwKKi2JfcAHguumO a=ice-options:google-ice a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=mid:audio a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:W6pHDYfKQD4ai/Njpdo/vbgDlTZMQIPSUJODaj9V a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+FNGrHBOuNn52F0oGJWBMlyniDRmTHaGnd/0fLgw a=maxptime:60 a=ssrc:3044764842 cname:We0LcuLfLdzi5DCu a=ssrc:3044764842 msid:l60QBLHC784xSyOLT5kAQYU4AG7U0ZwWyjcQ l60QBLHC784xSyOLT5kAQYU4AG7U0ZwWyjcQa0 a=ssrc:3044764842 mslabel:l60QBLHC784xSyOLT5kAQYU4AG7U0ZwWyjcQ a=ssrc:3044764842 label:l60QBLHC784xSyOLT5kAQYU4AG7U0ZwWyjcQa0 Is there some parameter in the SDP above that must be causing this. What would be the ways to debug RTP problems on freeswitch? I also strongly feel that this only happens when ICE and Secured RTP on freeswitch gets invoked. Any ideas or pointers to improve this voice problem is highly appreciated. Thanks, --- Jayesh

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