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Switch Gateway Routing
by Pieter Eduard 21 Sep '07

21 Sep '07

Hi, I just installed Freeswitch and my intention to use fs to route calls between gateways but am still stuck with the proper configuration. The simple diagram is like this : /subscriber A------> SIP Server A ------>Freeswitch----->SIP Server B-----> subscriber B/ voip prefix ip a.b.c.d ip 1.2.3.4 ip w.x.y.z voip prefix 777888x 999111x I tried to call subscriber B from subscriber A but getting this error : 2007-09-20 10:40:24 [NOTICE] sofia.c:1171 sofia_handle_sip_i_state() Hangup sofia//777888888@/a.b.c.d/:5060 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2007-09-20 10:40:24 [DEBUG] switch_channel.c:1076 switch_channel_perform_hangup() Kill sofia//777888888@/a.b.c.d:/5060 [KILL] 2007-09-20 10:40:24 [DEBUG] switch_core_session.c:638 switch_core_session_signal_state_change() Kill sofia//777888888@/a.b.c.d/:5060 [BREAK] 2007-09-20 10:40:24 [DEBUG] sofia.c:71 sofia_event_callback() event [nua_i_state] status [488][Not Acceptable Here] session: sofia//777888888@/a.b.c.d/:5060 2007-09-20 10:40:24 [DEBUG] sofia.c:1032 sofia_handle_sip_i_state() Channel sofia//777888888@/a.b.c.d:/5060 entering state [terminated] 2007-09-20 10:40:24 [DEBUG] switch_core_state_machine.c:347 switch_core_session_run() (sofia//777888888@/a.b.c.d/:5060) State HANGUP 2007-09-20 10:40:24 [DEBUG] mod_sofia.c:217 sofia_on_hangup() Channel sofia//777888888@/a.b.c.d/:5060 hanging up, cause: INCOMPATIBLE_DESTINATION 2007-09-20 10:40:24 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia//777888888@/a.b.c.d/:5060, cause: INCOMPATIBLE_DESTINATION 2007-09-20 10:40:24 [DEBUG] switch_core_session.c:697 switch_core_session_thread() Session 1 (sofia//777888888@/a.b.c.d/:5060) Locked, Waiting on external entities 2007-09-20 10:40:24 [INFO] switch_core_session.c:703 switch_core_session_thread() Session 1 (sofia//777888888@/a.b.c.d/:5060) Ended 2007-09-20 10:40:24 [NOTICE] switch_core_session.c:705 switch_core_session_thread() Close Channel sofia//777888888@/a.b.c.d/:5060 [CS_HANGUP] Which configuration file should i edit so i could pass the traffic from A to B through fs? Here's my config@fs : /_*default_context.xml* _/<context name="default">  <extension name="test1"> <condition field="destination_number" expression="^(9991111[0-3]{3})$"> <action application="set" data="call_timeout=30"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/test1/$1(a)/w.x.y.z/"/> </condition> </extension> <extension name="test2"> <condition field="destination_number" expression="^(777888[0-9]{3})$"> <action application="bridge" data="sofia/$${sip_profile}/$1(a)/a.b.c.d/"/> </condition> </extension> </context> *_/freeswitch.xml/_* <?xml version="1.0"?> <document type="freeswitch/xml">   #set "sip_profile=/1.2.3.4/"  #set "global_codec_prefs=PCMU@20i,G729@20"           <section name="configuration" description="Various Configuration">   /_ _/                                                     </section> <section name="dialplan" description="Regex/XML Dialplan">  </section>   <section name="directory" description="User Directory">  </section>  <section name="phrases" description="Speech Phrase Management"> <macros> <language name="en" sound_path="/snds" tts_engine="cepstral" tts_voice="david">  </language> <language name="fr" sound_path="/var/sounds/lang/fr/jean" tts_engine="cepstral" tts_voice="jean-pierre">  </language> </macros> </section> </document> */_Sofia.conf.xml_/* <configuration name="sofia.conf" description="sofia Endpoint"> <profiles> <profile name="test1">  <aliases> <alias name="test1"/> </aliases>  <gateways> <gateway name="test1">  <param name="username" value="/myusername B/"/>  <param name="realm" value="/1.2.3.4/"/>    <param name="password" value="xxxx"/>    <param name="extension" value="/myusername B/"/>  <param name="proxy" value="/1.2.3.4/"/>  <param name="expire-seconds" value="60"/>  <param name="register" value="true"/>  <param name="retry_seconds" value="30"/>  <param name="disable-transcoding" value="true"/> <param name="caller-id-in-from" value="false"/> </gateway> </gateways> <domains>   </domains> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="sip-ip" value="$${bind_server_ip}"/>  <param name="inbound-codec-negotiation" value="generous"/>             <param name="accept-blind-reg" value="true"/>  <param name="nonce-ttl" value="60"/>           /_ _/ </settings> </profile> <profiles> </profiles> <profile name="test2">  <aliases> <alias name="test2"/> </aliases> <gateways> <gateway name="test2">  <param name="username" value="/username A/"/>  <param name="realm" value="/a.b.c.d/"/>    <param name="password" value="/password/"/>  <param name="caller-id-in-from" value="false"/>    <param name="proxy" value="/a.b.c.d/"/>  <param name="expire-seconds" value="60"/>  <param name="register" value="true"/>    -<param name="caller-id-in-from" value="false"/> <param name="disable-transcoding" value="true"/> </gateway> </gateways> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5061"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="sip-ip" value="$${bind_server_ip}"/>  <param name="inbound-codec-negotiation" value="generous"/>             <param name="accept-blind-reg" value="true"/>  <param name="nonce-ttl" value="60"/>                  </settings> </profile> </profiles> </configuration> appreciate it if anybody could give me clue Thx, ~pieter~ /_ _ /

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Re: [Freeswitch-users] Should I choose FreeSwitch?
by Anthony Minessale 20 Sep '07

20 Sep '07

Everyone, What I want to see is someone to start a wiki page that is a TODO list for missing documentation and then for *everyone* who uses FreeSWITCH to think back to their first impression and what things they wish were spelled out better and add it to that page. Or at the very least submit your account to this mailing list thread. Or maybe add unanswered questions to the FAQ page for people to fill in? Something....? All I have heard are blanket statements like: "There is little or no documentation..." This is a very relative statement. What exactly do you want to see? For instance the API is well-documented with doxygen but if you don't plan to code anything then that is not any help for you. There are getting started guides and dialplan guides but maybe you don't understand the paradigm behind the software enough. There are nearly 500 wiki pages but maybe none of them tell you what you want to know? I obviously do not need any documentation since I invented most of this stuff off the top of my head and it all makes perfect sense to me. I am offering to push for this for everyone else's sake. This is the whole point of a community project so what we need is feedback or I will consider it safe to assume we are already as well-documented as we can be. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Jonas Gauffin <jonas.gauffin(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, September 20, 2007 9:17:54 AM Subject: Re: [Freeswitch-users] Should I choose FreeSwitch? > Ahh you kept saying people who 'cant' which led me to believe you were > talking about people who cant as opposed to wont. If they wont add, why > would they add to something else? So I am still confused why there needs > to be a "second class" documenter. Those that request others do things for > themselves that they wont. > How can they add/improve documentation if they can't understand how things work? What they can do is to tell us what they don't understand or if they are missing something. But how should they do that? And do they know that they can give us feedback? _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/

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Re: [Freeswitch-users] Should I choose FreeSwitch?
by Anthony Minessale 20 Sep '07

20 Sep '07

The important message from this thread is our apparent lack of documentation. Many of our more complex features are undocumented but one of the main reasons that is the case is that unless we have people volunteering to use the features and help others with them we will be instantly overwhelmed with people who want to use the feature and have little or no interest in contributing anything back to the project. It will be inevitable that we will have users of FreeSWITCH who give little back but I am not in a big hurry to invite this type of user while we are still in beta. Our lack of documentation does add up to over 500 pages mind you so I would possibly suggest that we have a lack of a proper table of contents more than a lack of documentation. So what I would like to do at this time is call on everyone out there on this mailing list to form a collaborative wiki page and to reply to this thread documenting what is missing from the documentation and what you would exactly like to see to be satisfied with the documentation. Also if you could list every feature you know FreeSWITCH can do (I only tell a handful of people about certain features so once everyone writes them down you may be shocked). Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Lucia Rotger <lrotger(a)aircomp.aero> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, September 18, 2007 2:20:28 PM Subject: Re: [Freeswitch-users] Should I choose FreeSwitch? Thanks. As of now there are a lot of options. A BRI/SIP gateway has been suggested but as I'm starting out I'd like to narrow things down and keep it simple. I think I'll go with Asterisk (gasp) and keep an eye over here so I'm aware of the developments. Unfortunately I cannot contribute code but I will set up a FS at home and if it works, let it handle some lines at the office after BRI support comes up eventually. When I get the feeling of FS I'll contribute to the wiki, and I'm hoping to make it better. What's needed is a good and accurate reference of everything it can do, examples are secondary and if that's the only thing you see about an option it lets you wondering what more there is to that particular option. But I agree wholeheartedly that writing documentation for beta code is difficult. I've set up an account at pbxes, it's quite interesting. Best regards Lucia On Tue, 18 Sep 2007 18:54:30 +0200, Jonas Gauffin <jonas.gauffin(a)gmail.com> wrote: > I've never used Asterix and FreeSwitch is my first softswitch. But > I've been programming applications for pbx:es for about 10 years. I've > done so for brands like Alcatel, Philips, Nec, Ascom and 3COM. > > I agree with you that the wiki is still very incomplete, but it's > updated quite often. If you miss anything, just ask in this list and > we'll update the wiki. > > Also, more are more providers give you the option to connect with SIP > instead of BRI. > > I really like FS and can recommend it. There are some things still > missing, but those will get implemented eventually. > > As you, I'm quite new to VOIP. > > //Jonas > > On 9/18/07, Lucia Rotger <lrotger(a)aircomp.aero> wrote: >> I have to implement a production software PBX. After some googling I >> came >> out with two candidates: Asterisk and FreeSwitch. >> >> Asterisk looks like crap; granted, it's working in many installations >> and >> seems to be readier for prime time. But just looking at their bugs puts >> me >> away. Segfaults all over the place, crashes after handling a few hundred >> calls: in short, very poor, sloppy programming I think. Maybe I'm wrong >> here? >> >> I like the way things are handled here and I like the coding style a >> lot. >> I think it's a matter of time FS becomes a great piece of software. Also >> from looking at the bug tracker the issues don't seem to be like those >> plaguing Asterisk. >> >> What are your thoughts? apart from the obvious bias you have for FS >> since >> you are here? >> >> I'd also like to read documentation if I have to consider FreeSwitch. I >> see the wiki is incomplete in many places. There are >> variables/applications I can't even begin to guess what they do. >> >> Also I'm aware that FS is a platform and Asterisk if just a PBX so its >> ambitions are greater; in fact I only want a PBX not a communications >> framework but as I've said above I feel Asterisk is going to be a lot of >> pain. >> >> The PBX will have to handle no more than 24 calls through 12 BRI >> channels. >> The local phones will be SIP and the traffic will be mainly incoming >> calls >> although outgoing calls will be made also. I've never touched anything >> VoIP so I'm a total newbie. >> >> Thanks, >> Lucia Rotger >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Check out the hottest 2008 models today at Yahoo! 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Getting started guide
by Jonas Gauffin 20 Sep '07

20 Sep '07

A bot have edited the quick start guide, can someone be kind and revert it to previous version?

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Re: [Freeswitch-users] Should I choose FreeSwitch?
by Anthony Minessale 19 Sep '07

19 Sep '07

I know you excused the obvious bias but even then I typically as a policy do not recommend my own software to anyone. I am able to invite you to help work on the project and to welcome any input you have. Like you say, you will find that Asterisk is designed with PBX in mind where FreeSWITCH has a primary focus on being a switching platform and an application server. However, the way FreeSWITCH is designed it's more than possible to create a PBX environment just by way of configuration and possibly the addition of a few missing features that are mostly missing due to a non-urgency. I myself have done extensive work on the Asterisk project which does give me the right to comment on the quality of the code. I have many problems with Asterisk that have a great influence on why I started this project but I will not get into the details of them in this email. That said I still contribute to the Asterisk community to this day when I find time. If you do decide to use Asterisk I highly recommend you stick with the 1.2 release for your best chance of success. You also may want to consider Call Weaver, GNU Bayonne or YATE just to make sure you have considered all options. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Lucia Rotger <lrotger(a)aircomp.aero> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, September 18, 2007 5:39:08 AM Subject: [Freeswitch-users] Should I choose FreeSwitch? I have to implement a production software PBX. After some googling I came out with two candidates: Asterisk and FreeSwitch. Asterisk looks like crap; granted, it's working in many installations and seems to be readier for prime time. But just looking at their bugs puts me away. Segfaults all over the place, crashes after handling a few hundred calls: in short, very poor, sloppy programming I think. Maybe I'm wrong here? I like the way things are handled here and I like the coding style a lot. I think it's a matter of time FS becomes a great piece of software. Also from looking at the bug tracker the issues don't seem to be like those plaguing Asterisk. What are your thoughts? apart from the obvious bias you have for FS since you are here? I'd also like to read documentation if I have to consider FreeSwitch. I see the wiki is incomplete in many places. There are variables/applications I can't even begin to guess what they do. Also I'm aware that FS is a platform and Asterisk if just a PBX so its ambitions are greater; in fact I only want a PBX not a communications framework but as I've said above I feel Asterisk is going to be a lot of pain. The PBX will have to handle no more than 24 calls through 12 BRI channels. The local phones will be SIP and the traffic will be mainly incoming calls although outgoing calls will be made also. I've never touched anything VoIP so I'm a total newbie. Thanks, Lucia Rotger _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/

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Should I choose FreeSwitch?
by Lucia Rotger 18 Sep '07

18 Sep '07

I have to implement a production software PBX. After some googling I came out with two candidates: Asterisk and FreeSwitch. Asterisk looks like crap; granted, it's working in many installations and seems to be readier for prime time. But just looking at their bugs puts me away. Segfaults all over the place, crashes after handling a few hundred calls: in short, very poor, sloppy programming I think. Maybe I'm wrong here? I like the way things are handled here and I like the coding style a lot. I think it's a matter of time FS becomes a great piece of software. Also from looking at the bug tracker the issues don't seem to be like those plaguing Asterisk. What are your thoughts? apart from the obvious bias you have for FS since you are here? I'd also like to read documentation if I have to consider FreeSwitch. I see the wiki is incomplete in many places. There are variables/applications I can't even begin to guess what they do. Also I'm aware that FS is a platform and Asterisk if just a PBX so its ambitions are greater; in fact I only want a PBX not a communications framework but as I've said above I feel Asterisk is going to be a lot of pain. The PBX will have to handle no more than 24 calls through 12 BRI channels. The local phones will be SIP and the traffic will be mainly incoming calls although outgoing calls will be made also. I've never touched anything VoIP so I'm a total newbie. Thanks, Lucia Rotger

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distance of RTP packets
by Sluschny, Thomas 18 Sep '07

18 Sep '07

Hi all, i use an older version of FreeSwitch for some time now (trunk 4560) and i want to ask wether this problem is solved now: i have huge jitter between the RTP packets, e. g. with G.711@20ms i see 3 or 4 packets every 75 ms. Even sometimes i see packet every 90ms. It seems to be machine dependend, i use always WinXP but on different machines: workstation, Laptop and so on. Anyone else have these problems? I loaded a newer binary version for windows for a couple of weeks but it crashed, so i don't now if this problem is solved now. thanks in advance, thomas sluschny

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Re: [Freeswitch-users] Mod_sofia loading fail
by Anthony Minessale 18 Sep '07

18 Sep '07

That message usually indicates the the module has a dll dependancy that is not being satisfied. Windows is not kind enough to explain what it's looking for in the error message so you will have to get some trace tools and figure out what the missing dll is. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Yogesh Chaturvedi <ychaturvedi(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, September 18, 2007 2:35:03 AM Subject: Re: [Freeswitch-users] Mod_sofia loading fail Hi, sorry for late Actually i was getting same error message as Alberto Takeshi Mayama define that's why i did not repeat it. Here is the message: 2007-09-18 12:57:35 [ERR] switch_loadable_module.c:714 switch_loadable_module_lo ad_file() Error Loading module e:\FreeSwitchTrunk\debug\mod\mod_sofia.dll **The specified module could not be found. ** Thanks On 9/17/07, Michael Jerris <mike(a)jerris.com> wrote: Yogesh- I still need the error messages you are receiving to have any guess what is wrong. Mike On 9/16/07, Yogesh Chaturvedi <ychaturvedi(a)gmail.com > wrote: Hi, Yes , i am using latest code from trucnk, One thing more i want to add here, when i use window-xp sp4 and VS 2005 for compile the code it get compiled and load mod_sofia lib also, hope this information will be helpful. Thanks Yogesh On 9/15/07, Alberto Takeshi Mayama < mayama(a)brastel.co.jp > wrote: I'm having the same issue as Yogesh, and I'm using the same environment: VS 2005 / Win2000 SP4. I just picked the latest revision 5707 and tried again, but mod_sofia keeps failing to load. Error message is as follows: 2007-09-15 15:29:10 [ERR] switch_loadable_module.c:714 switch_loadable_module_load_file() Error Loading module E:\Takeshi\Projects\Freeswitch\debug\mod\mod_sofia.dll **The specified procedure could not be found. ** Inspecting mod-sofia.dll with DependencyWalker, iphlpapi.dll is marked as "missing export functions". The missing function is GetAdaptersAddresses. Maybe GetAdaptersAddresses is only present in most recent versions of iphlpapi.dll (WinXP, Win2003, Vista). I could not test this, as I have no access to any of these new OSs. Have anyone successfully started the more recent builds using Win2000? Takeshi Anthony Minessale wrote: Also can you make sure you are up to date with the latest trunk when reporting problems because we are in beta now so we need to work with the newest code when we deal with issues. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Michael Jerris <mike(a)jerris.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, September 14, 2007 9:13:47 AM Subject: Re: [Freeswitch-users] Mod_sofia loading fail Could you post the error messages you get when it tries to load please? On 9/14/07, Yogesh Chaturvedi < ychaturvedi(a)gmail.com > wrote: Hi, i am using lastest code form trunk : http://svn.freeswitch.org/svn/freeswitch/trunk and compiling version 5615 , I compiled successfully code but it do not load mod_sofia: I debug the code and found that it fail at Apr_dso_load () function and do not return status as 0 : But when i compile code (that i am using from last 3 month) has version 4712 , It works fine , what may be the problem: I am using : visual studio 2005 as compiler Microsoft windows 2000 SP4 as OS Thanks Yogesh _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search. _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Got a little couch potato? 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Re: [Freeswitch-users] Mod_sofia loading fail
by Anthony Minessale 18 Sep '07

18 Sep '07

Also can you make sure you are up to date with the latest trunk when reporting problems because we are in beta now so we need to work with the newest code when we deal with issues. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Michael Jerris <mike(a)jerris.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, September 14, 2007 9:13:47 AM Subject: Re: [Freeswitch-users] Mod_sofia loading fail Could you post the error messages you get when it tries to load please? On 9/14/07, Yogesh Chaturvedi <ychaturvedi(a)gmail.com > wrote:Hi, i am using lastest code form trunk : http://svn.freeswitch.org/svn/freeswitch/trunk and compiling version 5615 , I compiled successfully code but it do not load mod_sofia: I debug the code and found that it fail at Apr_dso_load () function and do not return status as 0 : But when i compile code (that i am using from last 3 month) has version 4712 , It works fine , what may be the problem: I am using : visual studio 2005 as compiler Microsoft windows 2000 SP4 as OS Thanks Yogesh _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+kids&…

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Using FreeSwitch with sip-Proxy
by Yogesh Chaturvedi 18 Sep '07

18 Sep '07

Hi! I am using FreeSwitch as sipPbx and Opensbc as sip proxy in some prototype, I want to register my FreeSwitch with my sip provider (that is gafachi) and request should go viya SipProxy(Opensbc). My freeswitch settings are: <gateways> <gateway name="gatway ip"> /// account username *required* /// <param name="username" value="username"/> /// auth realm: *optional* same as gateway name, if blank ///  <param name="realm" value="gafachi"/> /// account password *required* /// <param name="password" value="password"/> /// extension for inbound calls: *optional* same as username, if blank /// <param name="extension" value="username"/> /// proxy host: *optional* same as realm, if blank /// <param name="proxy" value="sip proxy ip"/> /// expire in seconds: *optional* 3600, if blank /// <param name="expire-seconds" value="600"/> </gateway> </gateways> As we know realm is the Ip address that goes with username and make request for example username is guest and realm name is gafachi request will be guest@gafachi and proxy is the ip address of sip proxy. When we set Geteway setting in freeswitch configuration file "freeswitch.xml" We gave realm as gafachi ip address and proxy as opensbc machine address . According to this, register request should to gafachi viya sip-proxy, but when we see this wireshark it add proxy ipaddress in register request instead of realm in sip message. As: To:<sip:a2541shRzJKLjtQA@192.168.96.113> So I debugged the source code of FreeSwitch and changed the code in : mod_sofia.c and in config sofia() and chage as : gateway->register_to = switch_core_sprintf(gateway->pool, "sip:%s@%s", username, proxy); To gateway->register_to = switch_core_sprintf(gateway->pool, "sip:%s@%s", username, realm); and its working fine and request go through proxy and secrate code captured by OpenSbc. Buit i think it is wrong to edit source code: when I used Asterisk and give register request in sip.conf (Asterisk conf file) like register=>username@gafachi:password:username@opensbcip and everything seems working fine. Also In this case it sends REGISTER sip:opensbcip (request uri) and username@opensbcip(To uri) so in this case opensbc takes it as it is going to register with opensbc and doesn't forward the request to gafachi. I'll be thankful for any guidance on what's going wrong or should I set something else somewhere. Thanks in Advance! Regards, Yogesh

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