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No audio in new bridge loopback/app after bridge timeout
by Tais Plougmann Hansen 27 Jun '13

27 Jun '13

Hi, I have an extension which first bridges a call through a sofia gateway and on timeout then bridges the call to loopback/app=voicemail. When ignore_early_media is false and the call is rejected (ie. busy) then call goes to voicemail and sound is heard. Call_timeout is ignored in this scenario so I changed ignore_early_media to ring_ready/true. Now when the bridge times out it is bridged to the voicemail as with the rejected call, but no sound is coming from voicemail at all. The console shows it going through the prompts as usual and there are no indications of an error. If I call the voicemail app directly without the bridge, sound works as expected. <extension name="test"> <condition field="destination_number" expression="^(\d+)$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="proxy_media=false"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="call_timeout=20"/> <action application="set" data="ignore_early_media=ring_ready"/> <action application="bridge" data="sofia/gateway/pbxs1/${dialed_extension}|sofia/gateway/pbxs2/${dialed_extension}"/> <action application="answer"/> <action application="sleep" data="500"/> <action application="bridge" data="{default_language=da}loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>  </condition> </extension> -- Tais Plougmann Hansen OSD Consulting ApS osd.dk Tel: +45 78101078 CVR: DK31332737

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Re: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode
by Eli Burke 27 Jun '13

27 Jun '13

We were never able to get Trusted MITM working, but ZRTP can work in proxy-media mode as long as your client sends the zrtp-hash attribute in its SDP. When Freeswitch sees the zrtp-hash attribute it enables logic to pass the zrtp packets through unmolested (using the same SSRC for each call leg). As Mehroz noted in his original question, Linphone does not currently support zrtp-hash so this can't work. However, Mehroz, if you look in the Feb 2013 archives of the linphone-developers mailing list you can find a patch that adds zrtp-hash support to Linphone. It was recently rejected because it causes side-effects when video is enabled, but you can test it out and see if it fixes the your problem. -Eli On Apr 9, 2013, at 12:01 PM, freeswitch-users-request(a)lists.freeswitch.org wrote: > From: Steven Ayre <steveayre(a)gmail.com> > Subject: Re: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode > Date: April 9, 2013 12:00:21 PM EDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > There is a way to do Trusted MITM I believe there is a setting for that > > <action application="set" data="zrtp_enrollment=true"/> > > > On 9 April 2013 16:11, Ken Rice <krice(a)freeswitch.org> wrote: > Also, keep in mind that FS in a mode other then proxy or bypass media, when > you have a ZRTP call is kind of pointless, that leads to a vector for a man > in the middle attack which is also the reason the SASs don't match is you > have 2 different ZRTP legs with FS in the middle decrypting and > re-encrypting the conversation... > > There is a way to do Trusted MITM I believe there is a setting for that > > > On 4/9/13 4:57 AM, "mehroz" <mehroz.ashraf85(a)gmail.com> wrote: > > > Any luck with this issue? > > > > I am having the same configurations, FS acting as normal mode i.e bypass and > > proxy on default(commented) > > > > Clients are Linphone, and SAS does not match...... > > > > Over jitsi ... SAS are identical ONLY IF > > > > IF linphone does not support "zrtp-hash" attribute in SIP/SDP, what is the > > next possible solution?

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what's the grammar of transfering call with console command
by Vincent Xia 27 Jun '13

27 Jun '13

i tried originate user/1002 @transfer(e9976c95-f43f-45a0-bfeb-047931d26dbc) XML default //e9976c95-f43f-45a0-bfeb-047931d26dbc is the uuid of the ongoing call between 1005 and 1006 but it seems not working.

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Dialplan XML -> regexp on a variable don't work
by Alexander Haugg 27 Jun '13

27 Jun '13

Hi, if i make a check on a variable, then it doen't work, ${destination} will shown as empty: -> <regex field="${destination}" expression="^(.+)$"/> but if i make a subcheck in this line, then it works korrektly: -> <regex field="${regex(${destination}|^(.*)$)}" expression="^(.+)$"/> Why? I understand something wrong? On the Wiki -> Dialplan_XML -> Conditions -> Nested Condition Caveats and Notes. Here is a example with conditions with variables. see my complete extension tag. <extension name="my_extension"> <condition field="destination_number" expression="^001234(.+)$" break="on-false"> <action application="set" data="destination=${global_getvar(C4B_OneNumber_$1)}"/> <action application="log" data="DEBUG ${destination}"/>  </condition> <condition regex="all"> <regex field="${regex(${destination}|^(.*)$)}" expression="^(.+)$"/>    <anti-action application="log" data="INFO [$1]"/>  <action application="bridge" data="sofia/internal/123${destination}@$${c4b_gateway01}"/> <anti-action application="log" data="ERR is empty? [${destination}]"/>  </condition> </extension> Thanks for your Help Alex

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Dynamic specify the outbound GW within source code
by fieldpeak 27 Jun '13

27 Jun '13

Dear friends, i have FS for PSTN outbound call using below dial plan, <extension name="CallToPSTN" > <condition field="destination_number" expression="^(0\d+)$" > <action application="bridge" data="sofia/internal/$1(a)1.2.3.4" /> </condition> </extension> While, now i need dynamically specify the outbound GW’s IP address according to the return result of the external command before routing in the source code , e.g. if the external command return FS the IP address of OB GW 6.7.8.9, then <action application="bridge" data="sofia/internal/$1(a)6.7.8.9" /> however, i don't know which function i should call within the source code to realize it, could anybody help to advise, P.S. i know there is existing module “mod_xml_curl” can realize similar function, however, I could not use it for this case… thanks a lot! Regards, Charles

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FreeSWITCH Weekly Conference Call Today
by Michael Collins 26 Jun '13

26 Jun '13

Hello all! We have the guys from the HOMER project joining us today. Here's the agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_26 Hope you can join us! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Caller-Destination-Number DNIS truncated
by Nageshwara Rao Moova 26 Jun '13

26 Jun '13

Hi All, Recently i updated my freeswitch and see that "Caller-Destination-Number" only passes last two digits of DNIS. I have B2B setup, earlier when driver passed DNIS as 2001 fresswitch mod_freetdm variable "Caller-Destination-Number" had 2001. But now with latest version of freeswitch, the Caller-Destination-Number has 01, what is causing this truncation when Caller-Source is mod_freetdm. But with Caller-Source mod_sofia (VoIP) we don't have any issue with Caller-Destination-Number. Any help appreciated. -- regards & thanks -- m nageshwara rao 99891 86280

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mod_sms install issue
by stephen＠picardogroup.com 26 Jun '13

26 Jun '13

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Freeswitch getting killed with segfault
by Manish Kumar 26 Jun '13

26 Jun '13

Hi, My freeswitch process gets killed randomly leaving no log trace under freeswitch logs, the only log i am able to found is under /var/log/messages is Jun 10 15:05:21 FreeSwitch11 kernel: freeswitch[26059]: segfault at ffffc20000140520 rip 00000031e44223f2 rsp 0000000041e078d8 error 7 Freeswitch Environment : AMD FX(tm)-8120 Eight-Core Processor, 16 GB RAM 3 Digital Sangoma Cards A108 (24 E1's) version : freeswitch-1.0.7 FreeTDM library Perl ESL Kindly help to debug this issue... Thanks, Manish

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Add custom variables to mod_xml_curl
by Ivan C Myrvold 26 Jun '13

26 Jun '13

Is it possible to have custom parameters in addition to the normal parameters sent in FreeSWITCH'es MOD_XML_CURL POST request?

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