March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

domain-wise context
by Jayesh Nambiar 03 Apr '10

03 Apr '10

Hi All, I am quite very new to freeswitch and I am kind of playing with it to understand it better. I am primarily using Opensips as registrar and SIP Proxy and intend to use FS as media server handling voicemails, IVR, Announcements, MeetMe etc. My Opensips is a multi-domain setup and I wish to have all the configuration of media-capabilties segregated domain-wise in the FS too. For eg: When a call for user(a)domain1.com needs to go to voicemail, I redirect that call to FS IP address keeping the URI intact. I add the mailbox number as a header as X-Mailbox and have FS extract it and go to appropriate mailbox. Similarly when a call for user(a)domain2.com needs to go to voicemail I do the same thing. The requirement is I want to maintain the dialplans for each domains separately. Thus if call from Opensips comes to FS with domain as domain1, the call should go to dialplan context domain1 and similarly if call from Opensips comes to FS with domain2 the call handling should be mentioned in the domain2 context. The problem is; I am not able to send the calls to respective contexts according to their domains when they come from Opensips. I've read the examples on multi-domain setup and have tried taking some help from that example, but whenever the call comes from Opensips to FS, it tries to go into the context that is defined in the SIP Profile. If i don't mention anything in the SIP Profile, it tries to search for default context. I have tried the following: 1) Created file domain1.xml and domain2.xml in the directory folder. 2) mentioned parameters in domain1.xml as follows: <include> <domain name = "domain1.com" <params> <param name="user_context" value "domain1.com"/> </params> </domain> </include> 3) Similarly done for file domain2.xml. But I am just not able to get the calls to the required context according to the domain value in the r-uri. In asterisk something like this in sip.conf worked fine for me: domain=domain1.com, domain1.com domain=domain2.com, domain2.com Can someone please help me understanding where I am going wrong or have I mis-understood something? Thanks in advance !! --- Jayesh

3 4

Re: [Freeswitch-users] voicemail options [temporarly solved]
by Madovsky 02 Apr '10

02 Apr '10

Also no need to uninstall sendmail, just install ssmtp and that's it ! ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, March 18, 2010 1:59 PM Subject: Re: [Freeswitch-users] voicemail options [temporarly solved] Hi Anthony, after 5 days of tests to resolve the hell of sendmail segmentation fault, my conclusion is there is definitely a problem with the combination of cat, pipe and sendmail or other mail app. I made a php script that fits the actual switch_utils.c exec, but it's the same, it works one time on 10. So the robustest solution is to replace sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...) configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the switch.conf.xml like this <param name="mailer-app" value="/usr/sbin/ssmtp"/> <param name="mailer-app-args" value="-t"/> and apply the patch attached (no choice since it needs filename as argument and not STDIN). hope it will help people who have sendmail segmentation fault problem. Regards Franck ----- Original Message ----- From: Anthony Minessale To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, March 17, 2010 4:37 PM Subject: Re: [Freeswitch-users] voicemail options [temporarly solved] FYI if you made the change in your working copy you could generate a patch with svn diff > patch.txt After looking at your change, the problem is that many people depend on this working the way it currently is. Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works? On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos(a)madovsky.org> wrote: Sorry Anthony I never made an official update in C language with diff patch so the change is at the line 620, I commented out the original line ----- Original Message ----- From: Anthony Minessale To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, March 17, 2010 3:55 PM Subject: Re: [Freeswitch-users] voicemail options [temporarly solved] could you get a svn diff of the switch_utils.c so we can see what the change was? On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos(a)madovsky.org> wrote: Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also) it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist. I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification Regards Franck ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 4:35 PM Subject: Re: [Freeswitch-users] voicemail options ok, I tried 50 times to leave a message, it worked only one time. It seems that sendmail needs to be free to send the email.... ----- Original Message ----- From: devel(a)thom.fr.eu.org To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 3:59 PM Subject: Re: [Freeswitch-users] voicemail options Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.…) and cannot fix it. I don’t know if it’s a priviledge problem, as I got the same results running FS with or without –u. François De : freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Anthony Minessale Envoyé : mardi 16 mars 2010 20:30 À : freeswitch-users(a)lists.freeswitch.org Objet : Re: [Freeswitch-users] voicemail options are you running FS as root or a regular user? maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary? On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos(a)madovsky.org> wrote: I noticed that sendmail makes a segfault when FS is trying to send email and sendmail has some emails locked in folder queue.... Regards Franck _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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:+19193869900 ------------------------------------------------------------------ _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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:+19193869900 ------------------------------------------------------------------------ _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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:+19193869900 ---------------------------------------------------------------------------- _______________________________________________ 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

5 12

need help on call to gateway
by false 02 Apr '10

02 Apr '10

Hi all i use freeswitch to register to sip server the domain is not real: sip.yeah.com the outbound proxy : 118.69.145.5 i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder </include> <gateway name="sip.yeah.com"> <param name="username" value="071234"/> <param name="from-user" value="071234"/> <param name="proxy" value="sip.yeah.com"/> <param name="username" value="071234"/> <param name="password" value="1234"/> <param name="register-proxy" value="118.69.145.5"/> <param name="register" value="true"/> <param name="retry-seconds" value="10"/> </gateway> </include> so i use xlite register to freeswitch with user 1000 i start freeswitch , freeswitch start ok, freeswitch register to sip server successfull with username 071234 xlite register to freeswitch successfull so i edit the public.xml for incomming call from sip server, i will forward to extension 1000 <extension name="sip.yeah.com"> <condition field="destination_number" expression="071234"> <action application="transfer" data="1000 XML default"/> </condition> </extension> but when i use xlite make call out from freeswitch to sip server and get no routing nta: received INVITE sip:071234@192.168.1.90;user=phone SIP/2.0 (CSeq 3621) nta: INVITE (3621) going to a default leg nua(0xb6e20f90): adding session usage nta: sent 100 Trying for INVITE (3621) nua(0xb6e20f90): event i_invite 100 Trying nua(0xb6e20f90): call state changed: init -> received, received offer nua(0xb6e20f90): event i_state 100 Trying 432146 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000(a)192.168.1.90 [ef1af4b1-4d07-4b8e-8520-d99363347006] 446099 [INFO] mod_dialplan_xml.c:418 Processing 1000->071234 in context default 455113 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000(a)192.168.1.90 to enum[071234@default] 566021 [INFO] switch_core_state_machine.c:142 No Route, Aborting 566021 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000(a)192.168.1.90 [CS_ROUTING] [NO_ROUTE_DESTINATION] nua(0xb6e20f90): recv signal r_respond 404 Not Found i am not clear about routing call in freeswitch, i did read the dialplan in wiki but still not clear, is there any info to easy understand routing in freeswitch please help, Thank you Hãy cùng chat với bạn bè trên Yahoo! Pingbox mới toanh! Tạo một không gian riêng để chat trên blog là chuyện nhỏ. http://vn.messenger.yahoo.com/pingbox/

5 14

svn up then make current - Errors
by Aloysius Lloyd 02 Apr '10

02 Apr '10

Hello, Trying to update the svn source. svn up make current I got the following Errors. cd libs/openzap && autoconf configure.ac:9: error: possibly undefined macro: AM_INIT_AUTOMAKE If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. configure.ac:14: error: possibly undefined macro: AM_PROG_CC_C_O configure.ac:41: error: possibly undefined macro: AC_PROG_LIBTOOL configure.ac:119: error: possibly undefined macro: AM_CONDITIONAL make: *** [libs/openzap/Makefile] Error 1 How to fix the make error. Thanks Lloyd

5 6

mod_local_stream path based on rate
by Tim St. Pierre 02 Apr '10

02 Apr '10

Hello, I'm using local_stream:// as our music on hold source. I have both 8000 KHz and 16000 KHz files encoded so that I can play them at the native rate. We have some endpoints that can only do PCMU and others that can do higher rates. I want to set music on hold for a channel and have it automatically pick the appropriate stream based on the channel's native rate, just like the voice prompts do. I tried creating a stream for each rate, like rock-8000 and rock-16000 and then set the channel's music on hold variable to local_stream://rock-${read_rate} but the read rate hasn't been negotiated when this is set in the dialplan. Any suggestions? -Tim

2 2

Trying to test mod_silk
by Jason White 01 Apr '10

01 Apr '10

I am trying to test mod_silk with FreeSWITCH rev. 17118. The problem appears to be that I haven't been able to get my FreeSWITCH to offer SILK as one of its codecs. mod_silk is loaded. I am using the PortAudio interface to make the call; the other end is running a recently built FreeSWITCH as well, thanks to Bill Acker for participating in the testing as always. I have global_codec_prefs and outbound_codec_prefs in vars.xml configured with SILK@24000h as the first item in the comma-separated list. I have also tried it with SILK (no sampling rate specified). The SIP trace doesn't show that it is being offered. Bill has configured the other end in the same way. For example: a=rtpmap:10 L16/48000 a=rtpmap:114 CELT/48000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 is what appears as the SIP message from my local FreeSWITCH with SILK@24000h,CELT@48000h,G722,BV32,PCMU,PCMA,GSM in the codec preferences. (Actually, bv32 is out of date; I know the name changed and I should fix it - I haven't been playing with these settings for a while.) Full SIP traces are available if desired. It's x86_64 Debian GNU/Linux and IPv6 for making the connection. I know Brian and others have this working, so there's obviously something amiss here. As usual, there's no urgency - it's another attempt at a community contribution through testing.

3 7

mod_unimrcp with Nuance "Timed out waiting for channel to be ready"
by Adam Meehan 01 Apr '10

01 Apr '10

I'm using mod_unimrcp module with the Nuance Speech Server 5.1 and I'm getting the "Timed out waiting for channel to be ready" when trying to use TTS. This is using MRCP v2 and FS trunk rev 17126. I've not used MRCP and I am stuck. Any suggestions? Thanks Adam

2 4

Call Recording & bind_meta_app
by Bradley Brashier 31 Mar '10

31 Mar '10

I need to add the ability to record a call to an existing infrastructure, and I'm intending to use FreeSwitch to do this. The recording needs to happen upon a DTMF command, which I understand is normally done via bind_meta_app. From what I can tell, bind_meta_app requires that you use the * key in addition to some other key. Unfortunately, the system is already doing something else with *, so that's not a key I can use to get this to work. My question, then: what is the best way to record a call using DTMFs that do not include *? - Should I have Freeswitch forward an appropriate DTMF to a controlling application that starts recording? - Should I try to modify the bind_meta_app code to use another digit instead (say, #)? If so, where is the right place to look for this? As a separate, but related question, I also need to play a disclaimer when the recording starts ("This call is now being recorded."). Is it OK to use bind_meta_app to bind 2 apps to the same DTMF combo? BB

2 1

ClueCon MMX - It Will Be Here Before You Know It!
by Michael Collins 31 Mar '10

31 Mar '10

Greetings! We wanted to remind everyone to get a head start on their ClueCon arrangements. We have great things planned for this year and we don't want anyone to miss out. Please call 1-877-742-CLUE (2583) to register or if you have any questions at all about ClueCon. See you in August! -ClueCon Team http://www.cluecon.com 877.742.CLUE

1 0

Phone Registration Events
by Dave Stevenson 31 Mar '10

31 Mar '10

Hi, Is there a log option to turn on logging of SIP registration messages ? I have a mixture of phones and am having a problem with some of them periodically losing registration with FreeSWITCH. All the phones start off working after provisioning and have registration expiry times configured of 3600 seconds. I have tried changing the expiry time to no effect and I wanted to see if the phones were actually trying to renew the registration when they should. Regards Dave

2 1