May 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Sip Register Problem
by Nameer Kazzaz 07 May '09

07 May '09

Hi all, Please if anybody can help me with a problem I'm having. I have 2 gateways Quintum AX and OneAccess 100D both are setup to register with Freeswitch 1.0.4pre7, when I first switch on the gateways all is well and then after about 5 min sip register starts falling with sip 401. I can send what ever log's traces you need. Thank you.

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rtpDir Freeswitch
by Karl Vesterling 07 May '09

07 May '09

Roger on the previous post asking the question some time ago. I don't have time to get into this sort of thing, I've got too much on my plate right now as it is. Hopefully someone will take up the challenge. I've cross-posted this to the freeswitch forum. 73 de n2vqm Best Regards, Karl J. Vesterling kjv(a)ken-ton.com 202-461-3231 x0 On May 6, 2009, at 10:28 PM, ham44865 wrote: > > > Karl, I just remembered > about a few months back, someone logged in here and > asked the same thing and I sent him all > the channel drivers I had, except chan_dstar.c(NEW at the time). > Anyway, now it is all OPEN SOURCE. > > I dont know if he did something with them, but I found his posts > and my posts and pasted them now here. > Maybe he did something with them, I dont know. > > In eithe case, chan_rtpdir.c and chan_dstar.c are posted here. > > Question for you: > > chan_rtpdir.c and chan_dstar.c talk to a radio application > within asterisk so that the audio can go over the air. > > Is there any application inside Freeswitch that we can use > to interface chan_dstar.c and chan_rtpdir.c with ? > > Anyway the posts are 711 and 712 and here they are: > > ========================================== > Great, the more options the better. > > I will send you the Echolink channel driver and the > IRLP channel driver for Asterisk. > For IRLP you could also download Speak-Freely but it is > messy code. > Any questions you may have, let me know. > > 73, > de KI4LKF > > --- In rtpDir(a)yahoogroups.com, "Gerry Hull" <gerry@...> wrote: > > > > Hi All, > > > > rptDir looks pretty cool. I am doing a lot of work witrh > FreeSwitch, an > > open-source softswitch (http://www.freeswitch.org) It is much more > > scaleable and flexible than > > Asterisk (though I use Asterisk also). > > > > I'd like to write IRLP/Echolink conferencing for FreeSwitch. Are > their any > > good documents descibing the protocols, or are my only > alternatives to > > reverse-engineer the existing code? > > > > Of course, my resulting code would be open source. > > > > 73, Gerry W1VE > > sip: gerry(a)w1ve.com > > ================================================ > > --- In rtpDir(a)yahoogroups.com, Karl Vesterling <kjv@...> wrote: > > > > Roger on the channel drivers. > > > > www.freeswitch.org > > > > > > Best Regards, > > Karl J. Vesterling > > kjv@... > > 202-461-3231 x0 > > > > On May 6, 2009, at 9:00 PM, ham44865 wrote: > > > > > > > > > > > Is there anything in Freeswitch that resembles > > > "channel drivers". > > > > > > I am sure we can take the channel drivers I have for > > > Asterisk and maybe with a few modifications > > > make them run in Freeswitch. > > > > > > For example: I have > > > chan_rtpdir.c(for IRLP, Echolink), > > > and chan_dstar.c (for D-STAR) > > > > > > They call these things channel drivers. > > > I guess in Freeswitch, you might call them "patches", > > > not sure, but whatever they call them in Freeswitch, > > > we can port them to run there also. > > > > > > I am not familiar with Freeswitch, but it seems > > > a interesting project. > > > > > > I am not endorsing asterisk over Freeswitch. > > > I'd say the more competition the better. > > > I do not use asterisk at all, neither Freeswitch. > > > I just write and develop code for others to run. > > > > > > My main platform is rtpDir(D-STAR, Echolink, IRLP). > > > Let me know if I can help with Freeswitch > > > > > > 73 > > > Scott > > > > > > --- In rtpDir(a)yahoogroups.com, Karl Vesterling <kjv@> wrote: > > > > > > > > Folks; > > > > > > > > I notice there is a way to patch the rtp-dir into Asterisk, > but what > > > > about Freeswitch? > > > > Has there been any development in that area? > > > > > > > > The reason I ask is because I was running Asterisk for years, > but as > > > > of late last summer switched to Freeswitch to gain stability and > > > > capability over Asterisk. > > > > > > > > Hence, setting up rtpdir and integrating it into the VoIP > network > > > > (we're mostly hams) would imply some sort of Asterisk > installation > > > at > > > > each of the locations. And that's not an option any of us are > likely > > > > to entertain. > > > > > > > > The main problem with asterisk problem with asterisk with > asterisk > > > was > > > > the lengthy echoes that we encountered that we encountered > > > encountered > > > > that we encountered 'click'. > > > > > > > > We like our VoIP solid, secure, and free of problems. > > > > > > > > > > > > Best Regards, > > > > Karl J. Vesterling > > > > kjv@ > > > > 202-461-3231 x0 > > > > > > > > On May 6, 2009, at 6:15 PM, ham44865 wrote: > > > > > > > > > > > > > > > > > > > There will be no need any more for the > > > > > dextra_srv D-STAR gateway owner to create or maintain > > > > > the Berkeley DB database file dextra.db used by dextra_srv D- > STAR > > > > > gateway, > > > > > because direct access to ICOM's Postgres database will > > > > > be used by dextra_srv. > > > > > > > > > > As you know, dextra_srv is an OPEN SOURCE replacement > > > > > to the "closed source" dplus. > > > > > > > > > > 73 > > > > > Scott > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > __._,_.___ > Messages in this topic (5)Reply (via web post) | Start a new topic > Messages | Members > > Change settings via the Web (Yahoo! ID required) > Change settings via email: Switch delivery to Daily Digest | Switch > format to Traditional > Visit Your Group | Yahoo! Groups Terms of Use | Unsubscribe > RECENT ACTIVITY > 12 > New Members > 14 > New Files > Visit Your Group > Give Back > Yahoo! for Good > Get inspired > by a good cause. > Y! Toolbar > Get it Free! > easy 1-click access > to your groups. > Yahoo! Groups > Start a group > in 3 easy steps. > Connect with others. > . > > __,_._,___

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Busy tone and text message configuration
by chenexyee 07 May '09

07 May '09

Hi all I'm not familiar with the configuration for freeswitch. Anyone who could help me sovle the following two problem: 1. user A is in conversation with user B, and at this time, a incoming call from user C comes to A, in this case, I want freeswitch to play busytone to C, how to configure? 2. I'd like freeswitch to relay text message(use sip),as below scenario: entity1--------------------------freeswitch---------------------------entity2 | message(text/plain) ------> | message(text/plain) ------> | | <-------------200 OK | <-------------200 OK | | | | To implement above case, does need any special configure for freeswitch? Appreciate your help in advance. _________________________________________________________________ 看看谁是Live小博士，Windows Live知识竞赛, 精美礼品等你拿。 http://events.livetome.cn/2009/knowlive

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Interesting Blog About HD Telephony
by Michael Collins 06 May '09

06 May '09

FYI, Here's a nice story <http://www.freeswitch.org/node/182> for you all to check out. Please check it out and pass it on. -Michael

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Sphinx 4 Integration...
by Moiz Chinoy 06 May '09

06 May '09

Hi All, Has anyone tried Zanzibar and Cairo. They have implemented MRCP 2.0 and integrated Sphinx 4. Since MRCP 2.0 supports SIP for communication, it can easily be integrated with FS! Although it is implemented in Java, VoiceXML is also supported! -- Regards, Moiz Chinoy.

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ANNOUNCEMENT: FreeSWITCH 1.0.4pre7 Now Available
by Michael Collins 06 May '09

06 May '09

FYI, Please update your installations as soon as possible. More information on this update is available here <http://www.freeswitch.org/node/181>. Thanks for all of your feedback - please keep it coming and join us on IRC if you have any questions about the newest version. -Michael S Collins http://www.cluecon.com

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Profile reloading
by Jonas Gauffin 06 May '09

06 May '09

Hello, I have to static IP:s on my server. FS has been bound to one of them. Yesterday evening I got these log messages: 2009-05-05 19:17:37 [INFO] mod_sofia.c:2938 general_event_handler() IP change detected [85.89.XX.XX9]->[85.89.XX.XX8] []->[] 2009-05-05 19:17:37 [NOTICE] sofia_glue.c:3303 sofia_glue_restart_all_profiles() Reload XML [Success] And since FS changed IP to the other one, none of the phones could register to FS. The problem is that both IP addresses are the same ones that the server always have had. And XX9 is the one that I've bound freeswitch too. Regards, Jonas

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"conf-is-unlocked.wav" missing
by Peter P GMX 06 May '09

06 May '09

Hello, I tried conferencing for FS und tried to lock/unlock conferences. While "conf-is-locked.wav" was played, "conf-is-unlocked.wav" was missing in the file system. Any idea where I can download this? Best regards Peter

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ClueCon 2009 Blog, Moises Silva Speaking
by Michael Collins 06 May '09

06 May '09

FYI, I just wanted to let the community know that we maintain a blog<http://cluecon.com/blog/1>on the ClueCon website <http://www.cluecon.com>. Please check it out. The latest entry mentions Moises Silva's recent blog entry about his speaking at ClueCon this year. Please check the ClueCon blog periodically as we will be adding new information about speakers, sponsors, and other good stuff. -Michael S Collins http://www.cluecon.com

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Invite with TLS when originate
by Peter P GMX 05 May '09

05 May '09

I want to invite another party into a conference with TLS and SRTP enabled. Internal phones are invited by the following dialstring: {originate_timeout=30,sip_secure_media=true,context=default}sofia/default/723321(a)sip2.mydomain.de 72332200 Conference'. This enables SRTP but no TLS. Is there any variable I can set in order to enable TLS? set internal_auth_calls=true is meant only for configuration, hein? Also context=default doesn't succeed in this case. The call is passed to the public context. Best regards Peter

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