October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Best way to pass CRM data from FSClient into Freeswitch?
by roland＠callrica.co.za 23 Oct '12

23 Oct '12

Further to my email, I've been stepping thru the FSClient code, and I see all calls are made using this syntax: {origination_caller_id_name='MyCallerIDName',origination_caller_id_number='M yCallerIDNumber'}sofia/gateway/1/5000 Can I add my own variables e.g. {origination_caller_id_name='MyCallerIDName',origination_caller_id_number='M yCallerIDNumber', crm_accountid='1234', crm_contactname='johnsmith'}sofia/gateway/1/5000 And use the FS server dialplan to extract them into variables I can use in CDR and Voice recording? -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of roland(a)callrica.co.za Sent: 18 October 2012 02:22 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Best way to pass CRM data from FSClient into Freeswitch? Hi Freeswitch Team, I'm looking for advice on how to pass CRM related information from a FSClient to a Freeswitch server and use that data to customise the voice recording file name and CDR The use case is like this: 1. User starts softphone and dials a contact from his contact list e.g. John Smith 2. Softphone passes CRM Data e.g. "John Smith" with call into Freeswitch 3. Freeswitch switches the call and records the conversation 4. Freeswitch labels the recording filename as "John Smith_SomeDateTime.wav" 5. Freeswitch tags the CDR with "John Smith" and call duration, start time etc... I would like all the voice recordings and CDRs to be stored on the FS Server. Is it possible for the FSClient to embed the CRM data in the SIP message, and the FS server extract it into variables for use in the dial plan? Very new to all this, so any pointers would be much appreciated. Rgds _________________________________________________________________________ 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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redundant configuration questions
by Sean Devoy 23 Oct '12

23 Oct '12

Hi All, I haven't been able to search and find an answer or suggestion to this specific redundant server question. Sorry if it has been answered and I did not see it. My VOIP Provider allows for multiple endpoints by priority (aka primary and backup FS Servers). This works fine for getting inbound calls routed to my backup FS Server when Synapse Global is searching for the cause of my outage for 3 hours!!! However, none of the users phones are registered to this server. For my most demanding customer, I can route to his answering service on the PSTN. But, how do I get the phones to switch to the backup server and get on with life? Here are the ideas I have had so far. At least on my Linksys and Cisco phones and ATAs, I can specify DNS servers. Setting the PHONES DNS server to MY DNS server, I can eliminate the delay in changing a DNS Name/ Address mapping required for timeout and propagation. Changing DNS entries is thus immediate for the phones. Then we just have the clients ALL reboot their phones. Of course I realize this make my DNS server a single point of failure! Is there a cleaner approach? I have read up on SRV and it is clear as MUD. Do I just have to learn more about it? Thanks, Sean

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debian repositories and gpg key no longer valid.
by Mauro 23 Oct '12

23 Oct '12

Hello. I want to install freeswitch on my debian squeeze. I've put deb http://files.freeswitch.org/repo/deb/debian/ squeeze main in my soiurces.list, like said in http://wiki.freeswitch.org/wiki/Installation_Guide. Then I did update apt-get. curl http://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add - apt-get update. I've a gpg error it said that the key is not valid. Where can I put a valid key? Thank you.

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[CRIT] sofia_presence.c
by Luis Daniel Lucio Quiroz 23 Oct '12

23 Oct '12

2012-10-22 16:11:05.680467 [CRIT] sofia_presence.c:2424 WTF [snom821/8.4.20][full] what does this means? LD

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How to handle Transfer in Bridged call
by Nageshwara Rao Moova 23 Oct '12

23 Oct '12

Need help! I'm following the below link http://wiki.freeswitch.org/wiki/Force_transfer_context_example Requirement is to design a XML dialplan which can bridge calls and also handle/receive a transfer/REFER on the bridged calls. Once we have this working we will be also testing *T (*8 PSTN) transfer. Status: Using, Internal Context (for bridge) and Transfer Handler Context (for REFER), both context get loaded as single dialplan but when my IVR application initiates REFER it is stuck at processing REFER. FreeSWITCH Accepted REFER but fails to send NOTIFY to the IVR machine. Issue: 2012-10-23 02:27:49.516554 [DEBUG] sofia.c:7311 Process REFER to [(null)(a)10.130.XXX.XXX] Dialplan: <section name="dialplan" description="Dial Plan For FreeSwitch"> <context name="INTERNAL"> <extension name="bridgeit" continue="true"> - <condition field="destination_number" expression="2001"> <action application="set" data="transfer_after_bridge=ReleaseResource" /> <action application="set" data="call_timeout=30" /> <action application="set" data="exec_after_bridge_app=transfer" /> <action application="set" data="exec_after_bridge_arg=ReleaseResource" /> <action application="set" data="hangup_after_bridge=false" /> <action application="set" data="continue_on_fail=true" /> <action application="set" data="force_transfer_context=transfer_handler" /> <action application="bridge" data="sofia/VOIPBRD1/2001(a)10.21.XXX.XXX" /> <action application="unset" data="force_transfer_context" /> <action application="transfer" data="ReleaseResource" /> </condition> </extension> </context> - <context name="transfer_handler"> - <extension name="siprefer" continue="true"> - <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/VOIPBRD1/$1" /> </condition> </extension> </context> </section> -- regards & thanks -- m nageshwara rao

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How to add a display name to the To header
by Yannick LE COENT 23 Oct '12

23 Oct '12

Hello, Is it possible to add a display name to the To header? Thanks, Yannick

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FreeSWITCH Weekly News and Notes
by Michael Collins 23 Oct '12

23 Oct '12

Hello again! Another busy week in the land of FreeSWITCH. As you know we have been doing some work on the FreeSWITCH project's infrastructure. We are happy to report that this work is essentially complete. Thank you for your patience in this process. We've experienced intermittent outages but thanks to the efforts of Brian West and Raymond Chandler these have been kept to a minimum. We also thank Ken Rice for his hard work during this time. His knowledge and experience have been invaluable. Last week we had a very interesting discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2012_10_17>from Chad Engler about his nodesl <https://github.com/englercj/node-esl> library - a connector between node.js and ESL <http://wiki.freeswitch.org/wiki/Esl>. Chad describes himself as a Web developer learning the telephony side of things. However, we were all impressed with how much knowledge he had with using ESL given the relatively short amount of time he's been working with it. This week<http://wiki.freeswitch.org/wiki/FS_weekly_2012_10_24>we will be looking in on some Windows-centric ways of handling real-time data and related PBX items. The following week, on October 31, we will be hearing from our friends over at Redshift Networks<http://www.redshiftnetworks.com/>. They will be giving us close-up look at some of the worldwide SIP attacks that are being perpetrated against FreeSWITCH and other IP telephony servers. On a personal note I would like to thank everyone who volunteered to be a technical reviewer of the new edition of the FreeSWITCH book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-tel…>. I was overwhelmed with just how many people who are willing to volunteer their time. Packt Publishing will be contacting those who've been selected for this important job. This is a great community and we are all glad to be a part of it. Take care and have a great week. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Re: [Freeswitch-users] Problems with GV/mod_dingaling on latest stable version?
by Phil Quesinberry 23 Oct '12

23 Oct '12

Any input at all on this problem? I realize that since many have been reluctant to file a Jira when they should have done so, the knee-jerk reaction has been to force everyone to do it if there's even a hint of a problem and I'll be happy to file one, but it would be nice to at least find out if anyone else is experiencing the same problem first before potentially wasting everyone's time doing so. Additionally, any hints/pointers from someone more knowledgeable on how to start tracking this problem down would be great, as there is no trace info for XMPP/Jabber listed on the console by default or at all and little documentation so I'm pretty clueless as to where to start, other than trying to figure it out with tcpdump/Wireshark. To me at least, this functionality could be likened to a black-box that's built in to FS. I was able to block out some time yesterday to "attend" the FS conference and it was quite informative, so a big thank you to those who made that possible. I'd really love to see more like this, especially advanced troubleshooting topics covered in detail. Thanks, - Phil _____________________________________________ From: Phil Quesinberry [mailto:philq@qsystemsengineering.com] Sent: Wednesday, October 10, 2012 12:35 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: Problems with GV/mod_dingaling on latest stable version? I seem to have lost the ability to dial out via Google Voice on the latest stable version of FS. Is anyone else having this problem or could this possibly be related to the fact that I recently enabled TLS on profile Internal? I don't use it very often but it's handy to have available. Any ideas? Here's a short snippet of info from the console when the call was placed: 2012-10-09 23:41:23.110207 [DEBUG] mod_dingaling.c:1629 Don't have my audio codec yet here's one 2012-10-09 23:41:23.110207 [DEBUG] mod_dingaling.c:1675 Don't have video codec. 2012-10-09 23:41:23.110207 [DEBUG] mod_dingaling.c:1686 Send Describe [PCMU@8000] 2012-10-09 23:41:24.310174 [DEBUG] mod_dingaling.c:4055 using Existing session for 4654131323 2012-10-09 23:41:24.310174 [DEBUG] mod_dingaling.c:1675 Don't have video codec. 2012-10-09 23:41:24.310174 [DEBUG] mod_dingaling.c:1686 Send Describe [PCMU@8000] 2012-10-09 23:41:33.130018 [DEBUG] mod_dingaling.c:1525 Send rtp Candidate mypbxdomain.com:30536 [SgJi84it8jJU4Thf] 2012-10-09 23:41:33.130018 [DEBUG] mod_dingaling.c:1525 Send rtcp Candidate mypbxdomain.com:30537 [jXCuyH5i8GuihYyO] 2012-10-09 23:41:33.130018 [DEBUG] mod_dingaling.c:1525 Send video_rtp Candidate mypbxdomain.com:30538 [UOkj8Ty7HjUsELPw] 2012-10-09 23:41:33.130018 [DEBUG] mod_dingaling.c:1525 Send video_rtcp Candidate mypbxdomain.com:30539 [fjSU4YUTk84Tymoa] 2012-10-09 23:41:33.130018 [DEBUG] mod_dingaling.c:1565 Accepted 0 of 0 rtp candidates. 2012-10-09 23:41:33.130018 [DEBUG] mod_dingaling.c:1567 Accepted 0 of 0 rtcp candidates. 2012-10-09 23:41:33.130018 [DEBUG] mod_dingaling.c:1570 Accepted 0 of 0 video_rtp candidates 2012-10-09 23:41:33.130018 [DEBUG] mod_dingaling.c:1573 Accepted 0 of 0 video_rctp candidates 2012-10-09 23:41:39.609921 [DEBUG] mod_dingaling.c:4055 using Existing session for 4654131323 2012-10-09 23:41:39.609921 [DEBUG] mod_dingaling.c:4373 hungup dingaling/gtalk/+14105552368(a)voice.google.com 2012-10-09 23:41:39.609921 [DEBUG] mod_dingaling.c:801 Terminate called from line 4374 state=CS_INIT 2012-10-09 23:41:39.609921 [DEBUG] switch_channel.c:2950 (dingaling/gtalk/+14105552368(a)voice.google.com) Callstate Change DOWN -> HANGUP 2012-10-09 23:41:39.609921 [NOTICE] mod_dingaling.c:826 Hangup dingaling/gtalk/+14105552368(a)voice.google.com [CS_INIT] [NORMAL_CLEARING] 2012-10-09 23:41:39.609921 [DEBUG] switch_channel.c:2973 Send signal dingaling/gtalk/+14105552368(a)voice.google.com [KILL] 2012-10-09 23:41:39.609921 [DEBUG] mod_dingaling.c:2024 dingaling/gtalk/+14105552368(a)voice.google.com CHANNEL KILL 2012-10-09 23:41:39.609921 [DEBUG] switch_core_session.c:1210 Send signal dingaling/gtalk/+14105552368(a)voice.google.com [BREAK] 2012-10-09 23:41:39.609921 [DEBUG] mod_dingaling.c:2024 dingaling/gtalk/+14105552368(a)voice.google.com CHANNEL KILL 2012-10-09 23:41:39.609921 [DEBUG] mod_dingaling.c:4375 End Call 2012-10-09 23:41:39.609921 [DEBUG] switch_ivr_originate.c:3508 Originate Resulted in Error Cause: 16 [NORMAL_CLEARING] 2012-10-09 23:41:39.609921 [INFO] mod_dptools.c:3027 Originate Failed. Cause: NORMAL_CLEARING Version info: FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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how to properly handle DTMF in intercepted session ?
by daemonserj TVC 23 Oct '12

23 Oct '12

Hello I put calls in my java script (mod_java) Script doing this: 1) It parks first call (legA) and store it uuid (execute("park") 2) When second call (legB) arrived it intercept first call with previously saved uuid (execute("intercept") My dialplan is <extension name="test"> <condition field="destination_number" expression="^000111$"> <action application="answer"/> <action application="sleep" data="500"/> <action application="java" data="DTMFTest"/> <action application="log" data="INFO AFTER JAVA!!! channel=${channel_name}"/> <action application="playback" data="conference/conf-goodbye.wav"/> </condition> </extension> Firs thing is : when legA hangups then legB listen "goodbye" so extension continues execution. When legB hangups legA listen nothing and session just breaks. Where the mistake is ? The global question is how to process DTMF codes during the bridged session (DTMF hook in script is not working)? And how to then break this bridge and re-use this sessions with another arrived calls?

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Sample pcaps with HEP encapsulation?
by Lon Baker 22 Oct '12

22 Oct '12

Can anyone send me a couple sample pcaps with HEP encapsulation? I'm working with the guys over at VoIPMonitor.org to add support for Freeswitch sip-capture settings in the Sofia module. -- Lon

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