- FreeSWITCH-users - FreeSWITCH Mailing Lists

Equipment Location Survey
by Sean P Devoy 12 Dec '13

12 Dec '13

Hi All, I have had a moment of clarity and I think I need to rethink my business strategy with regard to FS for SMALL Businesses (less than 5 DIDs and 5 phones/extensions). To date I have attempted to HOST the customers phone services on a multitenant FS box "in a bunker". This has led to many setbacks due to NAT issues on the CLIENT end with different router and phone hardware. Perhaps I have a self-imposed problem that could be easily rectified. My strategy has been to be sure the switch is always reachable and route to alternate phones or voicemail if the client site is unreachable. I had thought this would provide for the highest availability of inbound call handling. I now think installing a FS PC onsite maybe be far simpler provided that the VOIP provider can handle routing to a failover switch. So I am considering changing from: <FS> -- public IP -- <internet> -- <NAT Router> -- Nat'ed client phones to: <internet> -- <NAT Router> -- Nat'ed FS -- local lan - non-nat phones I am looking for folks to talk about this configuration with FS NATed/Port Forwarded or FS dual home IP with a public IP and a nat ip. Many small businesses can't provide a public IP. In fact most will occasionally have their public IP change without notice. Clearly the VOIP provider will have to be updated with the new address. Is there a slick way to handle that problem? How well has this configuration worked for other people? Thanks, Sean --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

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Set custom SIP header in ESL Outbound
by Alex Balashov 12 Dec '13

12 Dec '13

Another newbie question, if I may: (I realise this may be more of a modesl/node-esl question than an ESL question per se.) I'm using node-esl/modesl for ESL outbound actions, and can't figure out how to get it to append a custom SIP header on the B leg. Were it the case that the 'set' formula were the same as with most conventional variables, set via a 'params' tag attribute, it'd be easy: conn.execute('set', 'foo=bar'); However, in the case of the ${sip_h...} container, this is specified as the body value of the <action> tag, not a 'params' attribute. Example per http://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Custom_Headers: <action application="set"><![CDATA[sip_h_P-Charge-Info=<sip:${caller_id_number}@${domain_name}>;npi=0;noa=3]]></action> I can't quite figure out how to get the API to mimic this effect. The second argument to the execute() method appears to get turned into tag attributes only. From https://github.com/englercj/node-esl/blob/master/lib/esl/connection.js: Connection.prototype.execute = function(app, arg, uuid, cb) { ... opts['execute-app-name'] = app; opts['execute-app-arg'] = arg; ... Am I just SOL on that one? I suppose one option is to run the call through a loopback endpoint that takes care of the header-setting in dialplan XML, but, as the aim is to build a thoroughly API-controlled environment, I'd rather avoid that kludge if I can help it. Any insights appreciated! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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make mod_h323 not working
by Richard Mace 11 Dec '13

11 Dec '13

Hi I am currently trying to build H323 on Ubutu 10.04 and am getting: /bin/bash: /usr/bin/ptlib-config: no such file or directory. Any ideas? PS I have build from source using the following page: http://wiki.freeswitch.org/wiki/Mod_h323 Richard

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Packet Fragmentation
by Charles Bujold 11 Dec '13

11 Dec '13

Using Freeswitch 1.2 on UBUNTU 12.04 and working well except there is one telephone number which I cannot call. When I reported it to our VOIP supplier he found that for that telephone number we have fragmented packets. Below is his comments. How can I fix this in Freeswitch. I do not know where to make the change. Thanks Hi, >From what I can see, there is fragmentation when trying to call the number. The packet size is 1514 bytes where the data is 1480 bytes. Also, your system sends this is the INVITE: X-Serialnumber: 0004134129BA P-Key-Flags: resolution="31x13", keys="4" which could be removed to lower the packet size.

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Periodic SIP messaging on bridged call
by Scott Ross 11 Dec '13

11 Dec '13

I'm using freeswitch for call control and just ran into an unexpected requirement for SIP keep-alive messaging. I've taken a look at the various keep-alive options on the wiki, including "all-reg-options-ping" and none seem to fit my requirements - I have an outbound call to an unregistered endpoint which expects a SIP MESSAGE every 60 seconds. Can I register an outbound call destination before-hand so all-reg-options-ping takes effect? Unfortunately, I don't have any control over these requirements. Is there a way to do this? Thanks, Scott

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Olly olly oxen FreeSWITCH!
by Brian West 11 Dec '13

11 Dec '13

FreeSWITCHers, Many commercial companies are very secretive that they use FreeSWITCH in their infrastructure, What can we do as a community to welcome larger commercial involvement in FreeSWITCH as a community? I would love to broaden our community out reach if possible. What does everyone else think? Also don’t forget to follow us on @freeswitch on twitter now! ;) PRETTY PLEASE!!!!! -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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Sofia bridge with RURI != destination
by Alex Balashov 11 Dec '13

11 Dec '13

Hi, Newbie here, and thus apologise if asking a banal newbie question: Is there a Sofia parameter that can be passed to the 'bridge' app (with a Sofia target, naturally) that specifies a network and transport-layer destination (IP:port) that overrides the request URI, which would normally get consumed for this purpose? That is, can I have FreeSWITCH send this request line: INVITE sip:abalashov@1.2.3.4:5060 SIP/2.0 But have it actually send the INVITE to 5.6.7.8:5060, not 1.2.3.4:5060? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Recordings out of sync again
by Kees Varekamp 11 Dec '13

11 Dec '13

2 years ago, I had this problem that recordings would get badly out of sync. I've never been able to pinpoint the problem but found that the voip client's "preserve bandwidth during silence" setting made the problem go away. But after moving to a new server, the problem has come back with a vengeance. And "preserve bandwidth during silence" does not seem to make any difference at all The new server runs on vmware - which got me thinking that running on a vm perhaps could cause such a problem? Has anyone had similar experiences? Thanks, Kees -- Kees Varekamp *MROffice Director* 44 Castleford street, 0604, Auckland, New Zealand Phone: +6492821656 Skype: kees.mroffice Web: mro <http://mro-global.com>ffice.org *MROffice - Meeting Market Research Needs*

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ClueCon Weekly! Today Codec Negotiation! (11-DEC-2013)
by Ken Rice 11 Dec '13

11 Dec '13

Hey Guys, Don¹t forget to Join us today 1PM EST, 10AM PST for ClueCon Weekly! Today, Codec Negotiation, how it works and how to control what Codecs FreeSWITCH is using. Join us on the Bridge! sip:888@conference.freeswitch.org or for other access methods see: http://fs0.us/call888 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Invite and Users Directory
by FERNANDO VILLARROEL 11 Dec '13

11 Dec '13

Dear All. I am confused with my user directory config I do not understand the follow: I am trying to config FS on Public IP addresse like 190.xxx.xx.xxx and fs and log show me: 2013-12-11 00:41:39.095757 [DEBUG] sofia.c:8066 IP 200.104.24.129 Rejected by acl "domains". Falling back to Digest auth. 2013-12-11 00:41:39.095757 [WARNING] sofia_reg.c:1634 SIP auth challenge (INVITE) on sofia profile 'internal' for [09999(a)190.xxx.xx.xxx] from ip 200.xxx.xx.xxx and i see that the calls are processed by public context 2013-12-11 00:41:39.155760 [INFO] mod_dialplan_xml.c:558 Processing 5644 <5644>->09999 in context public Why the calls are not processed by default context like say user config directory? My vars.xml <X-PRE-PROCESS cmd="set" data="domain=190.xxx.xx.xxx"/> <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/> My ~/conf/directory/default.xml <domain name="$${domain}"> <params> And when i try to set some variables like ${accountcode} this var did not set How i can solve my problem, i appreciated your advice Regards

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