December 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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DNS SRV in bridge, problems
by Miha 11 Dec '13

11 Dec '13

Hi, I have in a bridge dns srv domain which works right. When server with lower priorty is up, FS sends invite's to it which is ok. But when primary/lower priority server is down FS still send's invite's to it. All UAC re-register to server with higher priory which is only avalible. Are there any other configuration to be made on fs, so that FS will sends invites to right server? tnx miha

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Recording G729
by Kees Varekamp 11 Dec '13

11 Dec '13

Hi all, I'm suffering from poor sound quality on my setup, using GSM on the a-leg and PCMU on the b-leg. I'm thinking it's mostly bandwidth/network related, so I'm considering switching to G729 on both legs, as that is the only compressed codec that the b-leg supports (it's a sip gateway). But a complicating factor is that the calls need to be recorded in mp3. So I have some questions that I hope someone here can answer: - Does transcoding (for example GSM<>PCMU) negatively affect the sound quality, even when the CPU seems to be ok with it? - If yes, does recording 2 legs using the same codec (say G729) to a different sound format (say mp3) have the same (negative) effect? - When you use G729 on both legs (using mod_com_g729), it is even possible to record to mp3? Thank you, Kees

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Luasocket
by Neil Patel 11 Dec '13

11 Dec '13

I have built and installed Luasocket 2.0.2 as per the instructions on the wiki <http://wiki.freeswitch.org/wiki/Installing_LuaSocket>. I am running FS 1.2 stable. But as has been <http://jira.freeswitch.org/browse/FS-4712> noted<http://jira.freeswitch.org/browse/FS-2893>in Jira, Luasocket causes FS to crash from time to time. In particular socket.http.request() and post(). Has anyone been able to use Luasocket in a stable environment without FS crashing? What luasocket version? Any special build instructions? I would be happy to update the wiki with working instructions. Thanks, Neil

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