- FreeSWITCH-users - FreeSWITCH Mailing Lists

Problematic Behaviour of FS regarding ptime negotiation
by Jan Riedinger 11 Oct '10

11 Oct '10

3 10

subscribe
by Nicolas Brenner 11 Oct '10

11 Oct '10

subscribe

1 0

Where does "Expires:" in a sip trace come from?
by Mario 10 Oct '10

10 Oct '10

This newbie has 60+ hours into this problem and I really needs some expert explanation for 2 items in a "sofia profile external siptrace on": 1. The section that starts with "REGISTER", the line that reads "Expires:". Does the value always reflect what FS has set or is it reflecting the ITSP? I thought it was the expiry-seconds from the gateway definition. 2. Same question for the "SIP/2.0 200 Ok" section, line "Contact:", at the end it reads "expires:". Thanks. Mario

2 2

Get on IRC day!!
by Michael Collins 09 Oct '10

09 Oct '10

Hey all, we need bodies in IRC! It's Friday, let's have a good group to answer questions and help all the newbies. -MC

9 11

FS Components
by Abid Saleem 09 Oct '10

09 Oct '10

Hi, Could someone through some light on what are the functions FS can provide out of the below. SIP Server/Proxy ServerLocation/Presence ServerRedirect Server Thanks. Regards-----------Abid SaleemSr. Product Manager Terminus Technologies

2 1

event_socket - sendevent PRESENCE_IN
by Christian Löschenkohl 09 Oct '10

09 Oct '10

hello has this event changed lately? if i send this event with actual fs versions (latest git) the following notify sip body is send to our snom phones <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="2" state="partial" entity="sip:xxx@domain.com"> </dialog-info> this function is used to switch lights on an off (phone has an active subscription for xxx(a)domain.com) ----------------- in the versions used bevor something like this was send <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="" state="full" entity="sip:xxx@domain.com"> <dialog id="" direction="recipient"> <state>confirmed</state> </dialog> </dialog-info> this works as phones get the state of the led correctly ----------------- the normal notify (a calls b and c is subscribed with state of a or b) works as expected (sip body of notify packet is complete). br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

1 0

MIKEY-Support again
by Michael Baye 09 Oct '10

09 Oct '10

Hi, I was looking for MIKEY support in freeswitch and found an old thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-January/029829.… Here, Brian wrote: "Ed, Awesome. Looks like a Mikey lib will have to be created since the only libmikey out there is GPL. Unless someone wants to talk them into releasing it LGPL? I have talked to projects that are libraries in the past and asked if they would release LGPL and most have done so... It never hurts to ask." Well, looking at http://www.minisip.org/ it is said: "The source code is available as a number of libraries under the GNU Lesser General Public License (LGPL) and applications under the GNU General Public Licence (GPL)." Looking at svn://svn.minisip.org/minisip/trunk/libmikey/libmikey.spec only GPL is mentioned. Has anyone asked them since then if they would release it as LGPL? Why wouldn´t it possible to add an own branch with an GPL only libmikey extension within freeswitch? Sincerely yours, Michael

1 0

german prompts - first release
by Christian Löschenkohl 08 Oct '10

08 Oct '10

hello freeswitch community i can now announce the first release of the german prompt set. you can download it under http://freeswitch.xpirio.com/. this package is not useable at this time, because there are some xml files that need to be created or changed first. also the prompts are not 100% perfect and complete. i'll work on these topics and keep you updated. if somebody wants to help, please take a look at the prompts an send me your corrections (directly to my e-mail address, not to the list please). also look at the file de2.xls where you can find the prompts the need to be recorded or corrected. please take a look at the xml files under conf/lang, they also need to be created and/or modified too. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

4 5

FreeTDM Partial Spans?
by Jeroen C. van Gelderen 08 Oct '10

08 Oct '10

Hi Guys, This might be a dumb question but I can't seem to find an answer by Googling. I have oodles of FXO ports configured on Xorcom Astribanks. On the Astribank each set of 8 FXO ports forms a zaptel/openzap/freetdm span. Now if a trunk consists of an integral multiple of 8 FXO ports it is easy to just dial out on the first available channel: <action application="bridge" data="freetdm/1/a/$1|freetdm/2/a/$1"/> This grabs the first available channel out of 16 FXO ports. But what is the "right" way to do it when I need to use 1.5 spans (i.e. 12 FXO ports)? Is there such a thing as a virtual span that can be built out of individual FXO ports? I have a feeling I'm missing something rather obvious. Thanks, -Slim -- Jeroen C. "Slim" van Gelderen Cheers, -Slim -- Jeroen C. "Slim" van Gelderen Olympic Sports Data Services Email: slim(a)thegreek.com Phone: +1 876 953 6182 x128

2 3

Loss of first second of media from termination provider
by Fraser Redmond 08 Oct '10

08 Oct '10

I've just set up with a new termination provider, and am finding that the first second of media is being lost. I get early media (ringing), then when the call is answered the first half a second to a second is lost, then the rest of the call is fine. Their engineers have run a sip trace and a full packet trace, and it looks like everything is being delivered from the pstn to my server. They said that none of their other customers have reported this problem before. I've tested connecting to pstn's in several different countries, with the same results. I've used 2-3 other termination providers (1 still in current use) which have all been fine. So it seems like it's a combination of Freeswitch with this provider. This is running on Ubuntu on a Amazon AWS server, the version of freeswitch is about 2 weeks old, I'll be trying on a new server with the latest version tomorrow, in case that helps. Any ideas? Cheers, Fraser

1 0