June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

implementation of freeswitch in javascript / node
by michal 27 Jun '12

27 Jun '12

Hi there! Can someone recommend some resources on freeswitch and javascript/node? Ideally I would like a tutorial or guide on how it all works together (how to talk to FS) Thanks in advance!

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FreeSwitch Sending NOTIFY to incorrect port based on SUBSCRIBE request over UDP
by Parfitt, Stephen 27 Jun '12

27 Jun '12

Hi All, I've been doing some testing against FreeSwitch 1.2.0 and noticed what looks like a bug. root@Freeswitch> version FreeSWITCH Version 1.2.0-rc2+git~20120616T193227Z~068586f28f+unclean~20120618T093505Z 192.168.8.238 = Freeswitch 192.168.8.168 = proxyhost.test.net A Client sends the following SUBSCRIBE to freeswitch SUBSCRIBE sip:2001@proxyhost.test.net SIP/2.0 Call-ID: 2039516157894306129(a)192.168.8.168 Via: SIP/2.0/UDP 192.168.8.168:5070;branch=z9hG4bKC0A808A82E8652DB03 Via: SIP/2.0/UDP 192.168.8.168:5070;branch=z9hG4bKC0A808A82E8652DB11 To: sip:2001@proxyhost.test.net CSeq: 1 SUBSCRIBE Contact: <sip:proxyhost.test.net:5060> From: sip:2001@proxyhost.test.net ;tag=1297284766*476741369860*990463165~476741369860~fmc.sar~ Content-Length: 0 Expires: 3600 Event: message-summary Accept: application/simple-message-summary Route: <sip:192.168.8.238;lr> Max-Forwards: 69 FreeSwitch responds with: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.8.168:5070;branch=z9hG4bKC0A808A82E8652DB03 Via: SIP/2.0/UDP 192.168.8.168:5070;branch=z9hG4bKC0A808A82E8652DB11 From: <sip:2001@proxyhost.test.net >;tag=1297284766*476741369860*990463165~476741369860~fmc.sar~ To: <sip:2001@proxyhost.test.net>;tag=9t8fu6DedmnQ Call-ID: 2039516157894306129(a)192.168.8.168 CSeq: 1 SUBSCRIBE Contact: <sip:2001@192.168.8.238:5060> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120616T193227Z~068586f28f+unclean~20120618T093505Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Length: 0 So I would expect subsequent NOTIFY messages to be send to the address & port in the contact: sip: proxyhost.test.net:5060 (contact header from SUBSCRIBE) However I do not see the NOTIFY requests arrive at sip:proxyhost.test.net on port 5060, a wireshark trace on FreeSwitch says is sending them to a different port, port 33406 which is the port Freeswitch received the request on. This is how the packet looks in WireShark: Ethernet II, Src: 42:61:a8:ef:c3:de (42:61:a8:ef:c3:de), Dst: 96:cd:bb:a4:a4:2e (96:cd:bb:a4:a4:2e) Internet Protocol Version 4, Src: 192.168.8.238 (192.168.8.238), Dst: 192.168.8.168 (192.168.8.168) User Datagram Protocol, Src Port: sip (5060), Dst Port: 34406 (34406) **Would expect this to be 5060** Request-Line: NOTIFY sip:proxyhost.test.net:5060 SIP/2.0 I thought this defect might be in the ball park: http://jira.freeswitch.org/browse/FS-4014?page=com.atlassian.jira.plugin.sy… Anyone seen this? To resolve I used TCP instead of UDP but wanted to flag it up to see if anyone else had seen it Regards Steve P -------------------- Note: The information contained in this message may be privileged and confidential and protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by replying to the message and deleting it from your computer. Thank you. Thrupoint, Inc.

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FreeSWITCH Conf Call Today
by Michael Collins 27 Jun '12

27 Jun '12

Hello gang, This week's conference call will be starting in less than an hour. Here's the agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_27 Talk to you soon! -Michael

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Successful faxing with spandsp / T38 and stats
by Mitch Capper 27 Jun '12

27 Jun '12

Hello everyone, I have been working on faxing for awhile now and figure I would post back with some results and method. Much of my method I have documented on the wiki about faxing but some more details on my method are also below. Faxing overall is a horrific thing and when I first looked into VOIP faxing I did not think I would be able to achieve nearly the success rate I have been able to with FreeSWITCH. These stats were collected over 1000 individual faxes and over 4000 pages sent to over 500 unique fax numbers. I would guess a lot of these numbers are geographically clustered around the northwest of the US but I doubt that should matter a whole lot. Up until last week I had but 1 fax that didn't complete (it was an 80+ page fax that failed twice mid-fax before we just emailed it so it probably would have completed with some extra time). In the last week I have had two I have yet to get through (not sure why or if I have finally found fax machines I just can't send to or what have you). I was blown away with the current success rate. I did not collect as detailed stats the entire time I was logging data (probably only 3/4's+ of the total faxes have full detailed stats). With all that said the below numbers should be decently reflective of real world use. First on the method: I try a progressive retry system, I try one method if it fails I wait 60 seconds then try the next. I use two providers, flowroute and t38faxing.com. Flowroute I highly recommend they are a great provider and have worked very well for faxing. T38Faxing.com / EZCALL are quite bad and I highly recommend avoiding (see bottom of email for why). My 8 methods are: T38,ECM / ECM / T38_DISABLED,ECM / T38,SLOW / T38_DISABLED,SLOW / BACKUP,ECM / BACKUP,SLOW / BACKUP,SLOW,T38_DISABLED Slashes delimit the method and commas delimit the flags for that method. Slow=fax_disable_v17, T38=fax_enable_t38_request, T38_DISABLED=fax_enable_t38=false BACKUP=T38Faxing.com (T38faxing.com fails if t38_request is enabled so thats why T38 is not used with them). If the line is busy I don't move to the next method nor if there is a channel error. The methods are tried in order they are not tried randomly. Randomly would make the below stats better but I care about fax through a fast as possible so I try in order of what I thought had the best chance of success. average secs per page: 28.055 average tries until success: 1.33 (not including busy). Method Usage Rates: 75% T38,ECM success rate: 82% avg secs/page: 27.4 13% ECM success rate: 46% avg secs/page: 29.5 7% ECM,T38_DISABLED success rate: 60% avg secs/page: 28.3 2% T38,SLOW success rate: 56% avg secs/page: 45.7 1% SLOW,T38_DISABLED success rate: 57% avg secs/page: 40.6 1% ECM,BACKUP success rate: 57% avg secs/page: 35.8 Failure Reasons: 40% The call dropped prematurely 25% No response after sending a page 11% Invalid response after sending a page 6% Invalid ECM response received from receiver 5% Received no response to DCS or TCF 2% Unexpected message received 2% Timed out waiting for initial communication 2% The HDLC carrier did not stop in a timely manner 1% Received bad response to DCS or training 1% Disconnected after permitted retries 0% Timed out waiting for the first message Hope you enjoy the method and stats, honestly with this rate of success and either a few better backup providers or even a single hard line for faxing over you could run a very respectable faxing service using FS. ~Mitch Providers: As I mention on the wiki certainly one very important thing other than trying various methods is quality providers. Flowroute handled 99% of the faxes above obviously and did a great job. T38Faxing however I cannot recommend avoiding more. I used them as my backup provider after surveying #freeswitch a bit and while I normally try to avoid a brand bash it has just be a horrible experience. They have a very poor backend which would be livable if their support was not equally poor. They frequently don't answer the phone and their support email system has been down for awhile before (emails bounce) along with their control panel for their website. Their entire faxing service wasn't working for over a week when contacted they said they were adjusting their north american routing and it took over a day before it was back up. So overall I would suggest finding a different provider from t38faxing.

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user variables on outbound calls
by mbo 27 Jun '12

27 Jun '12

Hello, I've configured the following sip extension : <user id="1005"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1005"/> </params> <variables> <variable name="effective_caller_id_name" value="0221123456789"/> <variable name="effective_caller_id_number" value="02212345679" /> <variable name="outbound_caller_id_name" value="02212345679"/> <variable name="outbound_caller_id_number" value="02212345679"/> <variable name="user_context" value="sipoutbound"/> </variables> </user> When I'm dialing out from my sip client, the variables effective_caller_id_number and outbound_caller_id_number are not set. <context name="sipoutbound"> <extension name="socket"> <condition field="destination_number" expression="^(\d{5,20})$"> <action application="set" data="calltype=OUTBOUND" /> <action application="log" data="CRIT effective_caller_id_number ${effective_caller_id_number}"/> <action application="log" data="CRIT outbound_caller_id_number ${outbound_caller_id_number}"/> <action application="socket" data="192.168.1.185:8025 async full" /> <action application="log" data="DEBUG Back from Socket"/> </condition> </extension> </context> should't the variables be see with that configuration? Thanks Markus

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is there any solutions for registering only one sip phone at the same time?
by Samira Mh 27 Jun '12

27 Jun '12

is there any solutions for registering only one sip client at the same time, on the other hand if i issue following command : freeswitch@internal> sofia status profile internal reg there is one registered sip client, but at the same time if i registered again with the other sip phone (like 3cxphone) on the same computer o other computer with different or same ip ,so i registred again , but i don't want do that.. is there any solutions? Call-ID: NWYyNGU0N2RkOWJmYWU4NDlmZjZkZmUxZDdmZGQ5N2U. User: 100(a)192.168.10.1 Contact: "100" <sip:100@192.168.10.1:32838;rinstance=738b4fe258f6941e> Agent: eyeBeam release 1102q stamp 51814 Status: Registered(UDP)(unknown) EXP(2012-06-27 15:03:28) EXPSECS(3656) Host: pbx IP: 192.168.10.1 Port: 32838 Auth-User: 100 Auth-Realm: 85.15.7.66 MWI-Account: 100(a)192.168.10.1 Total items returned: 1 =================================================================================================

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FS upgrade succeeded but failed to recover voicemails
by dingdong 27 Jun '12

27 Jun '12

Hello all, i have a box that has been running for like 8 months now in production,its setup has been using a odbc in core. so i upgraded to a newer FS version,and honestly didnt know that it resets the config file to the defaults too,(without the odbc configurations). tried to reconnect the whole odbc thing. and it erased my old voicemail settings including ALL my users greetings and passwords. how can i prevent this from happening the next time i upgrade to newer versions? thanks

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xml_curl - group - URL Request
by Lloyd Aloysius 27 Jun '12

27 Jun '12

Hi All, How xml_curl works with the group. For Example in the default configuration <group name="billing"> <users> <user id="1005" type="pointer"/> <user id="1006" type="pointer"/> <user id="1007" type="pointer"/> <user id="1008" type="pointer"/> <user id="1009" type="pointer"/> </users> </group> ======= User Information FreeSWITCH URL Request => * section=directory&user=mike&domain=abc.mydomain.com* For Group what is the format of the request url? I could not find any reference in wiki Thanks Lloyd

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Strange behaviour when sending to voicemail
by ocset 26 Jun '12

26 Jun '12

Hi I have a FreeSWITCH install that uses a Faktortel SIP gateway for incoming/outgoing calls. The gateway codec is g.722 and everything works fine except for redirection to voicemail. (I have hard coded the codec for the gateway since it would not work otherwise.) Here is the conf/dialplan/public/01_line1.xml file with the dialplan for incoming calls <extension name="line1"> <condition field="destination_number" expression="^(5555555555)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="group/sales@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain} 1000"/> </condition> </extension> When the call is redirected to voicemail, half of the playback message is lost and the first thing the user hears is "to listen to the message press 1..." If I change the dialplan to transfer to the "1000" extension instead, allowing FreeSWITCH to redirect to voicemail automatically when ext 1001 does not answer, then voicemail works correctly. Any suggestions? Thanks O.

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softphones and background noise
by Juan Pablo L. 26 Jun '12

26 Jun '12

Good morning, i m having lots of issues with softphones, when a call is initiated/terminated to a softphone the background noise is too high that you can hardly hear the actual voice. i have played around with VAD and CNG turning them on and off in the profile and/or the dialplan but i have not had any luck. it does not matter if the call is internal or goes through a gateway, it is however, worse when it goes through a gateway, those are the cases you can not even hear the voice for so much noise and when the call is to a next extension the background noise is high but not as high. But, if you initiate/terminate the call to a hardphone (like polycom phones) the call is perfectly clear in all cases (except when the other end is an internal softphone). can someone please give me a hint. thanks a lot!

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