March 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
by Anthony Minessale 09 Sep '08

09 Sep '08

nonetheless, can you please see my bugnote in jira. ===== can you edit /usr/local/freeswitch/conf/autoload_configs/iax.conf.xml and set the debug param to 1. Then reproduce the case where it's not working. We can see if the mod_iax is even getting the digits. ===== also be sure to sync to the current release by entering "make current" from the build root. 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: Birgit Arkesteijn <birgit(a)westhawk.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 11, 2008 9:36:22 AM Subject: Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable Hi Anthony, I'm fairly sure that it's not an IAX problem. IAX2 messages are properly acked, sound (between IAX2 and SIP) is ok, etc, etc. The problem is (only) with DTMF, as far as I can tell. It's more likely to be in core or sofia_sip? Thanks, Birgit On 11/03/08 13:20, Anthony Minessale wrote: > Just to warn you we do not use mod_iax all that much. It was used as a > tool to test the core of FreeSWITCH in the early stages of development > and we just kinda left it hanging around since it does at least work. > We had to add a bunch of code to the stock iaxclient lib to make it > threadsafe etc because the lib is designed to handle 1 or 2 calls and we > are using it in a b2bua type env where hundreds of concurrent calls may > be possible. That said, since we more or less forked the code to avoid > generating support issues for the other soft-phones that use iaxclient, > you may want to check if there were any bugs in the real version of the > library that match your issue. > > http://sourceforge.net/projects/iaxclient > > > We've had little to no demand for IAX2 so we have not focused very much > on it. We do have strategies for creating a more robust and perhaps even > an "improved" implementation of the IAX2 protocol. Alas, it's pretty > much purely an academic exercise because there is little reason to > implement it. > > > > > > > > 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: Birgit Arkesteijn <birgit(a)westhawk.co.uk> > To: Freeswitch-users(a)lists.freeswitch.org > Sent: Monday, March 10, 2008 10:52:40 AM > Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable > > Hi, > > We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64). > > I'm experiencing problems with DTMF: > The IAX2 DTMF frames aren't always transferred into RTP packets. > Sometimes none of the frames are translated, sometimes only the first > one is omitted. All the IAX2 DTMF packets are ack-ed. > > Our setup: > PhoneFromHere.com <http://PhoneFromHere.com> applet (IAX2 protocol) > -> FreeSWITCH (ns2.westhawk.co.uk <http://ns2.westhawk.co.uk>) > -> SIP to SystemX (sofia/default/123(a)66.212.134.192) > > > I searched on jira.freeswitch.org <http://jira.freeswitch.org> for a bug > that could explain this, but > failed, so I submitted this as a bug, see > http://jira.freeswitch.org/browse/FSCORE-109 > > Has anyone else experienced the same or a similar problem? > > Thanks, Birgit > -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk> _______________________________________________ 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 ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Reg: SS7
by Gopal krishnan 21 Jun '08

21 Jun '08

Hi, Is freeswitch compatible with SS7 technology. How can i Implement SS7 in freeswitch -- Thank you with regards, Gopal,

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Profile configuration and aliases
by Jonas Gauffin 19 Jun '08

19 Jun '08

Hello Question 1: I got the following config: Name Type Data State ================================================================================================= outbound profile sip:mod_sofia@MY.PUBLIC.IP:5080 RUNNING (0) nat profile sip:mod_sofia@MY.PUBLIC.IP:5060 RUNNING (0) default profile sip:mod_sofia@192.168.1.124:5070 RUNNING (0) mydomain.com alias default ALIASED ================================================================================================= 3 profiles 1 alias I can't user "originate sofia/default/auser 8888", I get "Cannot locate registered user auser@default". It works if I use "originate sofia/mydomain.com/auser 8888". Shouldn't both work? -------------------------- Question 2: Isn't this valid? <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> <X-PRE-PROCESS cmd="set" data="sip_profile=mydomain.com"/> If not, please change the description: "sip_profile: Must be a domain name if you are being a registry server; otherwise can be any string." It produces the following config: Name Type Data State ================================================================================================= outbound profile sip:mod_sofia@MY.PUBLIC.IP:5080 RUNNING (0) nat profile sip:mod_sofia@MY.PUBLIC.IP:5060 RUNNING (0) default profile sip:mod_sofia@192.168.1.124:5070 RUNNING (0) 192.168.1.124 alias default ALIASED ================================================================================================= 3 profiles 1 alias User's cant register using mydomain.com as domain. I tried by adding $${sip_profile} as an alias for the default profile, but that didn't help. -------------------------------- Regards, Jonas

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Re: [Freeswitch-users] mfcr2
by Mike Murdock 11 Apr '08

11 Apr '08

What is mfcr2 used to interface with? Michael Murdock VP Operations & Development Switchmaxx Product Line 125 N. Market. Suite 1520 Wichita KS. 67202 (316) 932-2693 -----Original Message----- From: Brian West [mailto:brian.west@mac.com] Sent: Tuesday, January 30, 2007 11:46 AM Eastern Standard Time To: Yeruult[ICN]; freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] mfcr2 Do you happen to know of an open source libmfcr2 that is license compatible with FreeSWITCH? /b On Jan 29, 2007, at 7:10 AM, Yeruult[ICN] wrote: > Hello all, > > > > Does Freeswitch supports mfcr2? I know mfcr2 is old protocol but > that doesn't mean ppl don't use it. As I know there is still many > guys from everywhere. As I understood asterisk v1.4 becoming > complicated for Steve Underwood's unicall library. Is there any > other open sources which supports mfcr2? > > > > -- > > Best regards, > > Yeruult[ICN] mailto:yeruult@icn-mn.net > > _______________________________________________ > 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

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New to FreeSwitch
by Ritesh Singh 09 Apr '08

09 Apr '08

Hi All, I am very new to freeswitch. It will be great if some one can tell me few things: 1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y. Then what changes i am supposed to do and at what place. I would like to use the mod_portaudio for this purpose. 2) I started the windows freeswitch and loaded port_audio by using "load mod_portaudio" , then used "pa call 1234"...the session gets initiated but the call gets hanged up with the log "portaudio/1234 [CS_RING] [NO_ROUTE_DESTINATION]". 3) Suppose i have a jabber server x.net. and i have 2 users having account at that jabber server. Then how can i use mod_dingaling so that i have the voice chat between those user of the jabber server x.net. Also, i would like to dump the all the record regarding the call made, like duration of call , person who initiated the call, call destination , etc. Any help in this regard is highly appreciable. Please do forgive me if you think its stupid mail but i am desparate for these answers. This is my second mail as no one replied my first mail. Please do reply this mail else i ll be broken .... Thanks and Regards Ritesh

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yahoo blocking.
by Michael Jerris 31 Mar '08

31 Mar '08

Notice to yahoo users. We have been having ongoing issues with the mailing lists with yahoo, even after going through their white-list procedures. We suggest moving your mailing list subscriptions to a different e-mail account as yahoo seems to be rejecting or delaying many list emails. Also, please do not report list messages as spam. If you have any questions, feel free to reply privately to me. Thanks Mike

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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 21, Issue 76
by Ritesh Singh 31 Mar '08

31 Mar '08

Thanks to Brian and Michael for there valuable help. While trying with the "echo" from default.xml, I was getting a beep like sound, so i assume that my port audio is working.( I didn't hear the word echo though..is it fine ?). Then i tried to play a mp3 file and then a wav file BY adding few changes. <extension name="echo"> <condition field="destination_number" expression="^9996$"> <action application="answer"/> <action application="send_display" data="Echo Test"/> <action application="playback" data="file location"/> <action application="echo"/> </condition> </extension> When i tried this, freeswitch was not able to play the file as it was not able to find the ext. So what i am supposed to do so that i am able to play wav or mp3 file from specified location. I have one more question that can get me kick start in decdoing the freeswitch code. Here it goes: 1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y.(Currently assumng that both the ips are on same lan), Then what changes i am supposed to do in default.xml so that i can bridge the call between x.x.x.xand y.y.y.y using the freeswitch both computer can talk . I am using mod_portaudio as my end point. I think, i need to have freeswitch running at both the computer as how can y.y.y.y know that x.x.x.x has called it. Can you please tell me the exact dialplan to achieve this. And also is it possible to this with mod portaudio or i am in totally worng direction. Please suggest. (My irc name is Ritesh. I have just joined the IRC) Regards Ritesh On Sat, Mar 29, 2008 at 12:24 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send Freeswitch-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: New to FreeSwitch (Ritesh Singh) (Ritesh Singh) > 2. Re: New to FreeSwitch (Michael Collins) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 28 Mar 2008 22:18:47 +0530 > From: "Ritesh Singh" <riteshsingh81(a)gmail.com> > Subject: Re: [Freeswitch-users] New to FreeSwitch (Ritesh Singh) > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <98a09a460803280948q56f005f6w129adcc4fb6f941a(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On Fri, Mar 28, 2008 at 9:30 PM, < > freeswitch-users-request(a)lists.freeswitch.org> wrote: > > > Send Freeswitch-users mailing list submissions to > > freeswitch-users(a)lists.freeswitch.org > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > or, via email, send a message with subject or body 'help' to > > freeswitch-users-request(a)lists.freeswitch.org > > > > You can reach the person managing the list at > > freeswitch-users-owner(a)lists.freeswitch.org > > > > When replying, please edit your Subject line so it is more specific > > than "Re: Contents of Freeswitch-users digest..." > > > > > > Today's Topics: > > > > 1. Re: New to FreeSwitch (Brian West) > > 2. Re: How to bridge 2 sessions with Javascript? (Dale Thatcher) > > 3. Re: How to bridge 2 sessions with Javascript? (Anthony Minessale) > > > > > > ---------------------------------------------------------------------- > > > > Message: 1 > > Date: Fri, 28 Mar 2008 09:57:23 -0500 > > From: Brian West <brian.west(a)mac.com> > > Subject: Re: [Freeswitch-users] New to FreeSwitch > > To: freeswitch-users(a)lists.freeswitch.org > > Message-ID: <CE2A83CB-29B3-41B9-BD01-DFD7861AC801(a)mac.com> > > Content-Type: text/plain; charset="us-ascii" > > > > > > On Mar 28, 2008, at 8:07 AM, Ritesh Singh wrote: > > > > > Hi All, > > > > > > I am very new to freeswitch. It will be great if some one can tell > > > me few things: > > > > Welcome to the community. > > > > > > > > > > > 1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y. > > > Then what changes i am supposed to do and at what place. I would > > > like to use the mod_portaudio for this purpose. > > > > > > 2) I started the windows freeswitch and loaded port_audio by using > > > "load mod_portaudio" , then used "pa call 1234"...the session gets > > > initiated but the call gets hanged up with the log "portaudio/1234 > > > [CS_RING] [NO_ROUTE_DESTINATION]". > > > > I'm sure this message is rather clear... You do not have a route in > > your dialplan for 1234 thus it fails. > > > > > 3) Suppose i have a jabber server x.net. and i have 2 users having > > > account at that jabber server. Then how can i use mod_dingaling so > > > that i have the voice chat between those user of the jabber server > > > x.net. > > > Also, i would like to dump the all the record regarding the call > > > made, like duration of call , person who initiated the call, call > > > destination , etc. > > > > Feels like your trying to do too much at once before you fully > > understand what is going on. > > > > > Any help in this regard is highly appreciable. Please do forgive > > > me if you think its stupid mail but i am desparate for these answers. > > > > > > > You're trying to fly before you can even crawl. You have many things > > to learn. > > > > Best things to do are start here http://wiki.freeswitch.org and > > #freeswitch on irc.freenode.net > > > > Remember I'll be very glad to answer questions but the requirement is > > that you MUST put the info you learn on the wiki and pay it forward to > > others. > > > > Thanks, > > /b > > > > > > > >

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Help Setting up Freeswitch
by saumya garg 31 Mar '08

31 Mar '08

The following is my scenario: I have some SIP account, xyzUSER(a)ekiga.net, registered on Ekiga. I have a 10 digit US number that is attached to this SIP account. I want Freeswitch to answer incoming calls on this account. I am able to use the ekiga client that comes with ubuntu and connect to the server and receive calls using this 10 digit number. The following is included in my /conf/sip_profiles/outbound folder: <include> <gateway name="ekiga.net"> <param name="username" value="xyzUSER"/> <param name="from-user" value="xyzUSER"/> <param name="password" value="xyzPASS"/> <param name="extension" value="xyzUSER"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> </gateway> </include> I've tried the above settings with and without the settings "register" and "caller-id-in-from" commented. When freeswitch starts, I know that these settings are at least partially correct, for I see the following in my log file: 2008-03-29 20:17:41 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering ekiga.net 2008-03-29 20:17:41 [DEBUG] sofia_reg.c:753 sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with 'Digest:"ekiga.net":xyzUSER:xyzPASS'. 2008-03-29 20:17:41 [DEBUG] sofia_reg.c:684 sofia_reg_handle_sip_r_register() Changing expire time to 3600 by request of proxy sip:ekiga.net 2008-03-29 20:17:42 [DEBUG] sofia_reg.c:87 sofia_reg_check_gateway() registered ekiga.net 2008-03-29 20:37:12 [DEBUG] sofia_reg.c:87 sofia_reg_check_gateway() registered ekiga.net 2008-03-29 21:15:35 [DEBUG] sofia_reg.c:87 sofia_reg_check_gateway() registered ekiga.net 2008-03-29 21:59:21 [DEBUG] sofia_reg.c:87 sofia_reg_check_gateway() registered ekiga.net 2008-03-29 22:42:26 [DEBUG] sofia_reg.c:87 sofia_reg_check_gateway() registered ekiga.net In spite of these log messages, when I try to dial my 10 digit number I only hear a busy signal. I feel that the error is in dialplan. As per the diagram located at http://wiki.freeswitch.org/wiki/Image:FreeswitchDiagram.png, which I created following a link that was provided to me on freenode via IRCt, calls coming in via my ekiga SIP should end up in the secure default context. Nevertheless, I have adding the following block of code to both my public and default contexts: <extension name="xyzUSER"> <condition field="destination_number" expression="^xyzUSER$"> <action application="answer"/> <action application="echo"/> <anti-action application="answer"/> </condition> </extension> What else could I be missing? Thanks in advance for your help! Sincerely, Saumya Garg

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Sip Gateway registration - User Agent Change
by Joseph Bajin 31 Mar '08

31 Mar '08

In order to register with my sip provider, I need to have a specific user agent. Can someone tell me how I change that in my setup? Thank You Joe

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Segmentation fault on mod_python, with trunk r7982
by Arnaldo de Moraes Pereira 30 Mar '08

30 Mar '08

Hello, I've found a segfault running python scripts through mod_python and filled a bug on jira: http://jira.freeswitch.org/browse/MODLANG-54 Take care, Arnaldo M Pereira

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