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Help ！ How to adjust the speex Codec 's parameter for my sip phone
by sky tan 20 Dec '09

20 Dec '09

hi, all I used my soft phone with speex phone in speex, 8000hz , my softphone ' codec is base on opal lib. but when I register it in freeswitch and call in a conference . I fine the voice quality is not good enough。 I did print the parameter in switch_speex_decode() function , I find that my softphone sent encoded_data_len=21. if I send frame as encoded_data_len=20, then the voice quality become nice. So ,who can tell me how to change the freeswitch's speex parameter, then I can operate the encoded_data_len=21 ? best regards . your freeswitch' fan. _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so…

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iksemel
by Kristoff Bonne 20 Dec '09

20 Dec '09

Hi all, Can somebody please explain how exactly the "iksemel" library works. I've been trying to hunt down the bug that causes freeswitch to crash when making a jingle/gtalk call from ubuntu Empathy instead of the google talk web/flash application. I have narrowed the issue down to the difference in jingle-messages call-setup messages send over the jabber-channel by empathy, compaired to what the google webapplet sends. (see http://jira.freeswitch.org/browse/LBDING-18 for more info on the exact differences). However, I'm trying to understand how exactly freeswitch processes these messages. So, if I get it correct, it goes like that: - jabber xml-messages are treated by the iksemel library. - this library is based on "filters", which are configured beforehands. - when a jabber xml-message comes it, these filters are called. (va the function "iks_parse"). - these filters call, in their turn, the function then call a number of times the function "iks_find_attrib". In this bug, the problem seams to be when a call-setup packet comes in from empathy, on of these filters call the function "find_with_attrib", however pointing to a datarecord which is not a valid iks-record, resulting in a memory violation and hence a core-dump. But can somebody explain how exactly the "iksemel" library works. There seams to be a system of "hooks" used to attach pointers dynamically created filters-functions. But how exactly it all fits together is not very clear to me). If I understand it well, the problem needs to be solved somewhere in the code that creates these filters and define on what datarecord they should be called. However, I got lost in the code. Can somebody explain this a little bit. Cheerio! Kr. Bonne. -- jabber/gtalk: kristoff(a)krbonne.net

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multiple actions in an extension with different condition executing each after possible???
by rentmycoder rentmycoder 19 Dec '09

19 Dec '09

Hi pals, I would like to convert the following dialplan int fs xml dialplan, is this possible??? ;----- check for ivr vars ----- exten => s,16,GotoIf($["${IvrID}" = ""]?17:18) exten => s,17,Set(IvrID=0) exten => s,18,GotoIf($["${IvrActionID}" = ""]?19:20) exten => s,19,Set(IvrActionID=0) exten => s,20,GotoIf($["${QueeID}" = ""]?21:22) exten => s,21,Set(QueeID=0) exten => s,22,GotoIf($["${ChannelID}" = ""]?23:24) exten => s,23,Set(ChannelID=${CHANNEL_INNER}) exten => s,24,Set(CallStateID=${CALLSTATE_DIALING}) It would be easy if we could use nested conditions... <extension name="test"> <condition field="destination_number" expression="^9999$"> <condition field="${IvrID}" expression="^$"> <action application="set" data="IvrID=1"/> </condition> <condition field="${QueeID}" expression="^$"> <action application="set" data="QueeID=1"/> </condition> </condition> </extension> But we can't:( <extension name="test"> <condition field="destination_number" expression="^9999$" break="never"/> <condition field="${IvrID}" expression="^$"> <action application="set" data="IvrID=1"/> </condition> <condition field="${QueeID}" expression="^$"> <action application="set" data="QueeID=1"/> </condition> </extension> I'we tried this trick also, but second and third condition get's never executed... As a workaround I could define different extensions and use transfer to jump to the next but I have a lot of thing to check resulting a lot of extensions... There should be a more elegant way... isn't it?

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FreeSWITCH Weekly Conference Call Starting Shortly!
by Michael Collins 18 Dec '09

18 Dec '09

Hello everyone! Today's agenda is listed here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14 Also, we are going to be giving away goodies on some of the upcoming conferences, so call in and see what we've got in store. :) For the first 15 minutes we'll let everyone mingle and then we'll get into the agenda. Talk to you all soon! -Michael

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coredump (dingaling)
by Kristoff Bonne 17 Dec '09

17 Dec '09

Hi All, I don't know if this is the correct place for this, but I have a problem so perhaps somebody here can help. I've been playing around with freeswitch for about a week now using an old mac-mini (ppc G3) running debian. Yesterday, I wanted to give googletalk (dingaling) a try but got a coredump on freeswitch when I dialed by gtalk account from outside. Is there somewhere I can open a "ticket" for this? Can I upload the corefile (about 1.2 MB compressed) somewhere? This is on a PPC platform so I don't know if you people will have any use of it. Cheerio! Kr. Bonne. -- jabber/gtalk: kristoff(a)krbonne.net

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FreeSWITCH Weekly Conference Call Agenda For Dec 18 - Need Your Items
by Michael Collins 15 Dec '09

15 Dec '09

Hello friends, Just to let you know, I have posted the FS weekly conf call agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14 It's pretty clean at this point so if you've got things that you'd like to discuss with the group then please add your items to the list. If you have items that require attention, like documentation and janitorial items then by all means drop those on the list as well. One thing we do need to discuss is how we will accomplish screen casting. We are going to have presentations and we want as many people as possible to be able to view the screen while listening to the conference. Thanks, Michael

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FreeSWITCH 1.0.5pre9 is now available
by Michael Collins 14 Dec '09

14 Dec '09

FYI, The latest pre-release is now available. Usual information is available here: http://www.freeswitch.org/node/222 Please update as soon as you can. (SVN trunk users do the "make current" thing please.) We need your testing and feedback please! Many thanks for continuing to support FreeSWITCH. -Michael

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FreeSWITCH Weekly Conference Call Starting!
by Michael Collins 11 Dec '09

11 Dec '09

Come one, come all! http://bit.ly/8KzHCZ Talk to you soon! -Michael

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multicast presence, mwi and some registrations thoughts
by Matteo 11 Dec '09

11 Dec '09

Hi all, as promised, I've created a small patch to handle PRESENCE_IN and MWI events over multicast (using mod event multicast). When used along registrations sharing with multicast, you can send presence info over the mcast-bus :) and deliver the information to the phones, even if the call is not handled on the same switch the phone sent the subscription to. To make it work, the sip domain must be the all for all the machines and on the phones, otherwise the phones will ignore them. This allows distributed BLF and MWI notifications :) BUT doing that I noticed a issue that must be addressed imho, which arises when you handle NAT. "distributed blf" works behind nat right now, since the event will get sent to all switches, but delivered only to the phone were fs knows about the subscription. Which is the one where the phone registers to. "distributed mwi" results into sending N copies of same MWI notifications , where N is the number of switches. Behind NAT the phone will receive only the one sent by the switch where it's registered to. We can avoid sending multiple event if we're able to know which is the "original" server where the phone registered to. This is also needed to call the phone: think about a distributed registration cluster. if Alice (reg'ed on FS1) calls Bob on (reg'ed on FS2), right now the call is routed from FS1 to Bob, since FS1 knows where Bob is. This works ok without NAT. But with NAT FS1 cannot contact Bob, because Bob is on FS2 "really" and can receive calls from FS2 (nat pinhole). The solution is to check where Bob is (looking to registration table), get the "real" server and route the call from FS1 to FS2 then to Bob. But right now, sofia registration does not have a field holding the "original" server. If we want to leverage on distributed registration using multicast, I think knowing the original server is "a must". the patch is pretty trivial, but we must add a field onto the db. Doing that, we can also filter MWI notification, and send them only if the phone is "really" registered on the server which receives the MWI event over the mcast-bus . What do you think about it? Regards, mat P.S. this is my first patch to FS, so please be kind on my mistakes :)

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FreeSWITCH Weekly Conference Call Agenda For Dec 11
by Michael Collins 11 Dec '09

11 Dec '09

FYI, Here's the agenda for tomorrow's conference call: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_11 Please be ready to join at 11AM CST! :) Don't forget to bring your agenda items, questions, and a willingness to help out with our various janitor projects. Thanks! -MC

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