August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

enable-100rel question
by Mike Burlingame 24 Oct '13

24 Oct '13

I was looking to turn on 100rel however based on the below info from the wiki I am not sure I want to run the risk of it crashing FS is this information still accurate? >From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files enable-100rel This enable support for 100rel (100% reliability - PRACK message as defined in RFC3262) This fixes a problem with SIP where provisional messages like "180 Ringing" are not ACK'd and therefore could be dropped over a poor connection without retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, see FSCORE-392. <param name="enable-100rel" value="true"/>

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Video switching
by mehroz 23 Oct '13

23 Oct '13

Hi guys, I have Linphone SIP client and my UI allows me to switch call from audio to video and video to audio. After call establishment, when i switch to video, it do so ! and when i switch back to audio , it works. But after that when i switch to video , it doesn't! Any guess ? what could be wrong here? Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Video-switching-tp7593439.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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Possible Polycom bridging issues in all recent FreeSWITCH versions
by Emrah 11 Oct '13

11 Oct '13

Hi, I have been experiencing the following issue since I upgraded my version of FS. In a scenario where a called is made from one Polycom VVX1500 to another, the video works both ways with only one way audio. A hold / unhold results in 2 way audio with 1 way video. I obviously suspected my configs, especially after subsequently trying the latest git, the 1.4 beta and the stable 1.2.10. So far, the only way I found to solve this issue is by using an old copy of FreeSWITCH with my current configs. FreeSWITCH Version 1.3.13b+git~20130119T155523Z~6cb3be7d8b (git 6cb3be7 2013-01-19 15:55:23Z) To clarify, the same config with any of the latest versions of FS does not work. Questions: How can I troubleshoot this further and help resolve this issue? Any one has an idea of what has happened? Thanks!

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Prevent A leg from hangup after bridge with inbound ESL socket
by Alex Massover 09 Oct '13

09 Oct '13

Hi, I have a very simple dialplan that just do park for incoming calls. All rest of leg management is done via ESL inbound socket. I'm trying to do the same behavior like in this dialplan example, but from ESL inbound socket: <action application="set" data="hangup_after_bridge=false"/> <action application="bridge" data="sofia/internal/1001@$${domain}"/> <action application="bridge" data="sofia/internal/1002@$${domain}"/> The problem is with bridge API, if B leg doesn't answer (e.g. 404, or busy), A leg disconnects. But I'm trying to prevent A leg from disconnecting in order to do bridge to other place. Looks like hangup_after_bridge=false, park_after_bridge=true, transfer_after_bridge etc don't have any effect when bridge done from inbound socket. A leg disconnects always. Is there any way to keep A leg after bridge with inbound socket? I'm aware of originate, but prefer to user bridge. -- Best Regards, Alex Massover

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execute_on_fax_success/failure
by Darcy Primrose 07 Oct '13

07 Oct '13

I am using excute_on_fax_success to call a script when a fax is received, the script converts the fax to pdf and emails it, Is there a way to pass the number of pages received, ie, a variable created by spandsp. Everything works, but I do not seem to be able to pass the variables. <condition field="destination_number" expression="^1(\d{10})$|(\d{10})$"> <action application="set" data="faxtempdirectory=/faxfiles/${caller_id_number}"/> <action application="system" data="mkdir ${faxtempdirectory}"/> <action application="set" data="faxDirection=outbound"/> <action application="set" data="emailAddress=$1$2@${fax2net_server}"/> <action application="set" data="faxFile=$1$2-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)-Success}"/> <action application="set" data="execute_on_fax_success=system /./var/mail/sentfaxes/fax2mail.sh ${faxFile} $1$2@${fax2net_server} ${fax_security_code} ${fromRc} fax2net_orig=${fax_email}= ${fax_transfer_rate}"/> <action application="answer"/> <action application="playback" data="silence_stream://2000"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="rxfax" data="${faxFile}.tif"/> <action application="hangup"/> </condition> Thanks Darcy Primrose

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Error Creating SIP UA for profile: X
by Henrik Aagaard Sørensen 04 Oct '13

04 Oct '13

I've had a server running only FreeSWITCH for about 6 months now. I have had it turned off for some while. Now, when trying to start FreeSWTICH, I get the errors: Error Creating SIP UA for profile: X where X differs between internal, external and others. When starting my CentOS 6 server I run: service iptables stop service ip6tables stop just to make sure that they aren't blocking anything. I have nothing else running on CentOS. I've tried to log everything from console, warning etc. but there's nothing indicating what the problem is. Can someone help me out?

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Freeswitch RTP endpoint linked to another RTP endpoint
by Danny Gershman 23 Sep '13

23 Sep '13

So I have an INVITE that comes in that doesn’t have SDP in it. I have the “enable-3pcc” turned on so FreeSWITCH does respond properly. So then it responds with SDP allocating an endpoint. Then I take that SDP information and I turn around and send a new INVITE with the SDP in it. The hopes is that this endpoint is then bridges to a new endpoint. Essentially Freeswitch is connected to itself. Here is the pastebin - http://pastebin.freeswitch.org/21327 Look at the CreatePlayConnectionForConference:1 in the User-Agent field. And then look at the CreatePlayConnectionForConference:2. Those two are supposed be linked together with RTP. Danny Gershman Principal Engineer Cinchcast 475 Park Avenue South, 19th Floor New York, NY 10016 646.807.0840 (o) 973.931.6239 (c)

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Cordless phones
by Jonas Gauffin 19 Sep '13

19 Sep '13

Hi, I'm looking for IP-dects where a base station can handle 5 phones (which all should be able to talk at the same time). All suggestions are welcome. Regards, Jonas

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Build error Mac OS X Mountain Lion
by Chris Cachor 17 Sep '13

17 Sep '13

Hello, I'm trying to build Freeswitch on my machine and when I run the make command, I get an error towards the end: /var/folders/zr/r7177m510r71w3z81959j9sc0000gn/T//ccUHViiz.s:58:suffix or operands invalid for `lea' /var/folders/zr/r7177m510r71w3z81959j9sc0000gn/T//ccUHViiz.s:63:suffix or operands invalid for `movq' make[7]: *** [gsm0610_rpe.lo] Error 1 make[6]: *** [all] Error 2 make[5]: *** [all-recursive] Error 1 make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 I've built Freeswitch on this same machine about 2 months ago, so not sure why I'm running into this issue. Has anyone had luck with recent builds on OS X Mountain Lion? - Chris

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Issue with JSSIP + Freeswitch
by Shahrzad A. 16 Sep '13

16 Sep '13

Hi everyone I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS with the latest version of Openssl ('OpenSSL 1.0.1e 11 Feb 2013') I'm using the default configuration and just uncommentated the ' <param name="ws-binding" value=":5066"/> ' in internal.xml in order to have the support for webrtc. As the client I'm having JSSIP, the latest version with the adjustment to have (DtlsSrtpKeyAgreement:true). Both my clients and server located in the same network without having any firewalls therefore I'm not using any STUN server. I can make call but there is no sound! and after setting debig_level 10 and dial from extension 1003 to 5000 I'm having the following Alert in console and no sound: [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) If I call from extension 1000 which is a SIP client set on my iPhone and call 1003 I'm having the following alerts but voice on both JSSIP client and SIP client: 2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/ 1000@10.0.14.16:5060 timer while HOT 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/ 1000@10.0.14.16:5060 Hot Hit 1 And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts: 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/ sip:1000@10.0.14.182:5065 Hot Hit 4 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/ sip:1000@10.0.14.182:5065 timer while HOT 2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) If I use bypass media or proxy media I will have voice on both JSSIP clients but cant ring any SIP clients and again no voice if I call 5000! Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP? Thanks in advanced! Sherry

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