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Oubound ESL: unbridge
by Grant Bagdasarian 24 Apr '14

24 Apr '14

Hello, I'm working on a C# application which takes control over an inbound call in Freeswitch using ESL outbound and it works like a charm! I can answer the call, play something, get digits, etc. Now I'm at the point of implementing bridge functionality in the application. Which one is the best approach and why? 1) From the Outbound ESL Application use the originate command to create a new outbound call and then use uuid_bridge to connect both incoming and outgoing? 2) Use the bridge command from the esl application? I would assume the first option would give me control over the b leg created as well, and the second option would not? This also brings me to my next question: - Is there a way to "unbridge" both call legs, do some stuff on the incoming (a) leg, and then bridge them back without disconnecting any of the legs? Regards, Grant

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How transport sip package via tcp protocol?
by rockxsj 24 Apr '14

24 Apr '14

I want to use tcp to transport the sip package from pjsip or linphone.But I set use tcp on the client,I found it connot connect to the server.Is there some config file should be modify?

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Exporting Diversion Header Twice
by Sanjiv Kumar 24 Apr '14

24 Apr '14

I am trying to export sip_h_Diversion header twice but I only see the last attempt on leg-B. It appears that second export attempt overwrites the first one. How can I export it twice? <extension name="public_did"> <condition field="destination_number" expression="^9999(\d{10})$"> <action application="set" data="ocn=18005551212"/> <action application="set" data="rdn=18005552323"/> <action application="export" data=" nolocal:sip_h_Diversion =<sip:${ocn}@${network_addr}>;reason=user-busy;counter=3;privacy=off"/> <action application="export" data=" nolocal:sip_h_Diversion =<sip:${rdn}@${network_addr}>;reason=no-answer;counter=1;privacy=off"/> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="continue_on_fail=true"/> <action application="transfer" data="1001 XML default"/> </condition> </extension>

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REGISTRATION
by Miguel Oyarzo 24 Apr '14

24 Apr '14

Hi all, I'm sorry for this basic question, I'm new in FS. Is it possible to accept any username&password for REGISTER messages in FS I'm trying to do it in directory/users.xml <user id="*"> but it doesn't work (this file force me to use a username&password for REGISTER). Is there any simple way to accept any user id? Scenario: Client ----> REGISTER (any user/pass) ---> FS Client <--- 200 OK <------------------------- FS Thanks, -- ================================== Miguel Oyarzo DevOps & VoIP Engineer Linux User: # 483188 - counter.li.org http://au.linkedin.com/in/mikeaustralia Melbourne, Australia ==================================

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Distinctive Ring
by Sean Devoy 24 Apr '14

24 Apr '14

Hi All, I want to make the destination phone have a different ring for Internal Calls and External Calls. We use Cisco SPA504Gs which have 10 system and 2 user defined rings to choose from for "default ring". They also have 10 "alternate ring patterns", but I am at a loss how to cause the phone to use an alternate ringer. Everything I see in the wiki has to do with ringback. Unless I misunderstand, that is the ring the caller hears. I want the phone to notify the callee this is a different call type. The only thing I have come up with is to setup 2 different Line Keys with different ringers and direct the calls appropriately. Besides requiring an extra <user>, it complicates the dialplan significantly. Thanks in advance, Sean

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How to dial out via a local extension? (possible?) - OBI110
by Mario G 23 Apr '14

23 Apr '14

Baffled for the last 2 days… I have an OBI110 connected to FreeSwitch as a user, so that it cal dial 911 out via a PSTN. Inbound is fine, but not matter what I do, it only dials out it’s own extension number (8181) on the PTSN line. I ran traces, etc. and don’t see what I am dialing, for example 3511, going to it. I tried to make it a gateway but FS returns 501 not supported. There are various web references to people using it for in/out on Asterisk and FreePBX but so put together FreeSwitch definitions in a similar way. I would really appreciate any suggestions at this point. I have tried many things, nothing seems to work. It always dials 8181 out. Thanks, Mario G <extension name="special.8181"> <condition field="destination_number" expression="^3([23456789]11$)$">  <action application="export" data="destination_number=$1"/-->  <action application="bridge" data="sofia/internal/$1%${sip_profile}"> </condition>

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FreeSWITCH Installer
by Thomas Quintana 23 Apr '14

23 Apr '14

Hi Guys, I have written a basic FreeSWITCH installer in Python that is fairly easy to use and figured I would share in case anyone has such a need. Any ideas or contributions are welcome :) https://github.com/thomasquintana/freeswitch-installer Best, Thomas Quintana

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Source Description RTCP Packets (SDES)
by deepak sujir 23 Apr '14

23 Apr '14

Hi, I am interested in capturing the Source Description RTCP Packets for the calls. Is there a way we can do the same? If not in order to implement this feature where shall I start looking in the freeswitch code. Thank you Deepak

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Invalid and unreliable DTMF
by Manish Kumar 23 Apr '14

23 Apr '14

Hi, I am facing an issue where DTMF digits are wrongly captured or getting entirely missed from being detected. Environment: *Sangoma *A116 / A108 *Freeswitch Version 1.2.14* *mod_freetdm* perl ivr scripts using *perl ESL* PRI lines from Indian operator. However the same script and same configuration works perfectly well on Freeswitch *version 1.0.7*. Also none of the versions gave any issue when i test by simply using mod_sofia and an extension on x-lite. I viewed the entire change logs from version 1.0.7 till 1.2.14, there are no changes on DTMF since 1.0.7 Are there any possibilities that telco plays a role as well in such DTMF issues.? Thanks,

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Help creating a JIRA for hang up that does not hang up.
by Sean Devoy 23 Apr '14

23 Apr '14

Hi, I have a VERY ODD problem. I am not totally convinced it is a FS bug. I want to file a JIRA, but I would like to know what would be the best logs/traces/packet captures to include (maybe with tips how to get them). Here is the problem: >From my SPA504G, I dialed 3101. Dialplan: <extension name="wb_conferences"> <condition field="destination_number" expression="^(31\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@wideband+XXXX"/> </condition> </extension> My conf call went smashingly well. At the end, I pressed the headset button to hang up (speaker phone is the same). The headset audio stops, the headset light goes out but the LINE is key still on solid and the display still says "To: 3101". Pressing the line key causes it to hang up. I am on a fairly recent git of the stable tree. FreeSWITCH Version 1.2.22+git~20140309T212137Z~65fed130e5~64bit (git 65fed13 2014-03-09 21:21:37Z 64bit) Centos 5 Thanks, Sean Thanks, Sean

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