September 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Is FIFO in memeroy or in DB?
by absdou24＠gmail.com 17 Oct '13

17 Oct '13

I hava two freeswitches ,these freeswitches share the same mysql, I make extension1 which stands Member join the fifo named 111 of FS1 to waiting for a Caller. Then a comming caller from FS2 hits the diaplan that matches fifo 111, I hope extension1 will answer the comming caller,However not. why?Is the data about FIFO in Memeroy,So FS2 can not get the extension1 status from the memory fron FS2. Please help me,Thanks a lot Keep in mind Quality First

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Sending SAVPF INVITE to Opensips
by James Mortensen 12 Oct '13

12 Oct '13

Hello, I have a bandwidth.com number pointed to opensips, and a WebRTC peer registered with Opensips. I'm trying to dial the 10 digit number from a cell phone and connect the call through FreeSWITCH to the Chrome WebRTC client. I defined opensips as a gateway, in the external profile: <include> <gateway name="ws-Opensips">  <param name="from-domain" value="54.X.X.75"/> <param name="proxy" value="54.X.X.75"/> <param name="expire-seconds" value="600"/> <param name="register" value="false"/> <param name="retry_seconds" value="30"/> <param name="extension" value="18257773456"/> <param name="context" value="public"/> <param name="avpf" value="yes"/> <param name="username" value="11234"/> <param name="password" value="password"/> </gateway> </include> In the public dialplan context, I added in a condition to catch the INVITE coming in from opensips and pass it to a context I've called "default-inbound". See the second condition: <extension name="from_opensips"> <condition field="network_addr" expression="^54\.X\.X\.75$" break="never">  <action application="transfer" data="${destination_number} XML default"/> </condition> <condition field="network_addr" expression="^54\.X\.X\.111$">  <action application="transfer" data="${destination_number} XML default-inbound"/> </condition> </extension> Then, in the default-inbound context, I match the dialed number, answer the call leg from the PSTN, and then try to transfer back through opensips to oversip and to Chrome. The problem is that I either end up sending back AVP INVITES, or Opensips refuses to authenticate the user. <extension name="bandwidth.com inbound bridge"> <condition field="destination_number" expression="^\+1(5035551212)$"> <action application="answer" /> <action application="set" data="variable_sip_auth_username=11234"/> <action application="set" data="variable_sip_auth_password=password"/> <action application="bridge" data="sofia/external/ws-Opensips/11234(a)54.X.X.75"/> </condition> </extension> As you can see, I've hard-coded my peer, 11234, in the configuration. This is a registered user on Opensips. How can I get FreeSWITCH to send the SAVPF INVITE through to Opensips for the WebRTC portion of the call leg? I apologize if this is covered somewhere, but I've been wracking my brain on this for days and am not getting anywhere. The Opensips configuration I have works with existing Asterisk 11 servers, and I'm hoping I can just simply plug in FreeSWITCH servers seamlessly into the mix. Thank you! James

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originate not creating xml_cdr record
by Matt Broad 11 Oct '13

11 Oct '13

Hi, I have setup xml_cdr and it is working great, both aleg and bleg calls are logged. I am using the api originate method, but this does not create an xml_cdr record (though a record is logged in /log/cdr-csv/master/csv). Is this correct? thanks Matt

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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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Checking call is connected while bridged in lua
by David Villasmil 09 Oct '13

09 Oct '13

Hello Guys, I've had this "problem" since forever :) (I've been a freeswitch user for a long time) I'm using lua and I want to bridge a call and to check immediately whether the call is connected or not WHILE connected. lua execution continues only after "bridge" has returned something.. connected, busy, whatever... But i would like check the status of the call immediately after executing the bridge... Is this possible WITHOUT esl? Thanks! David

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Optimizing profile
by Guillermo Ruiz Camauer 08 Oct '13

08 Oct '13

In the Wiki, under "Performance testing and configurations", one of the suggestions given under "Recommended SIP Settings" is: libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles If I only have one Voip provider, can I still have more than one SIP profile? My provider authenticates via IP. I currently run 240 concurrent calls through this Sip trunk, I I see CPU close to 12.5% on a 8 core machine. This means that one core is maxing out. How can I get more threads up to distribute the load? -- Guillermo Ruiz Camauer

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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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Sending and Receiving faxes - freeswitch
by Ravi 01 Oct '13

01 Oct '13

Hello Everyone, I am trying to send and receive faxes. I have assigned the number 4302050 for receiving faxes, and the use configuration is given below: <include> <user id="1050"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1050"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1050"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1050"/> <variable name="effective_caller_id_number" value="1050"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> The dial plan that I have in place to receive faxes is given below: <extension name="fax_receive"> <condition field="destination_number" expression="^(050$)"> <action application="set" data="ignore_early_media=true"/> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="$${base_dir}/faxes/${caller_id_number}.tif"/> <action application="hangup"/> </condition> </extension> When this fax number is called from another telephone, I hear the fax dialtone. But when the fax number is called from any fax machine and try to send a fax, it gives a number not available message. I do not recieve this call in the freeswitch server as well. Would really appreciate if anyone could help me in this, to let me know if I am missing something. Thanks in advance. Ravi

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