April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_conference lag
by Clint Popetz 02 Apr '10

02 Apr '10

Hi, I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with mod_skypopen, and when I call the echo test with skype it is _beautiful_ and has no lag, and the same is true for my coworker, but when we both dial a mod_conference bridge with skype, we get a 10-12 second lag. CPU usage on the machine is nil. Any ideas? Thanks, -Clint -- Clint Popetz http://42lines.net Scalable Web Application Development

4 5

difference
by Jean-Yves F. Barbier 02 Apr '10

02 Apr '10

Hi, What is exactly the difference(s) between extensions and endpoints? -- Why do we have two eyes? To watch 3-D movies with.

3 4

Grouping channels in span
by Nagalenoj H. 02 Apr '10

02 Apr '10

Dear friends, I'm using Sangoma A102 and I've configured it with wanpipe. I've question with regard to grouping the spans. There are 2 spans in the card and I would want to group the spans separately, so when I want to make call I could specify as either openzap/1/ or openzap/2/... But, Now I'm unable to make calls like this. It is asking me to give the dial string as openzap/1/a/123@g1, openzap/1/a/123@g2.. I don't want this way., I would want to group the spans differently. Kindly help me. -- Regards, Nagalenoj H.

3 3

default_gateway
by Jean-Yves F. Barbier 02 Apr '10

02 Apr '10

Hi list, I'm making tests for a residential conf that use an ATA to bridge with PSTN, can I define 'default_gateway' to a SIP extension (something like: 2222.xml), or am I oblige to use transfer (or something else)? -- It's the RINSE CYCLE!! They've ALL IGNORED the RINSE CYCLE!!

2 2

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

PSTN Integration and real deployments
by guru singh 02 Apr '10

02 Apr '10

Hi, After long nights and lots of coffee =) , I think I've largely understood FreeSwitch. I've been playing with it and have managed most fancy things it can do. But I've done this on my LAN using SIP softphones. Here's my problem now, I know nothing about PSTN integration and real deployments. Here are my questions, mostly based on what I read on wikipedia. PSTN integration: I have an ADSL internet connection, with a split-box? installed by my ISP which splits the incoming line to two, one for the phone provided and one for the adsl modem. I want to handle incoming PSTN calls via FreeSwitch and also be able to outbound calls to PSTN/VoIP phones via an SIP client registered with my FS server through an external gateway or the PSTN line. 0) I should get an ATA to do this? Is an ATA just a dumb adaptor that seamlessly converts SIP-PSTN traffic both ways or does it require configuration? What are the ATA's that work best with FS? 1) I should register with a VoIP/SIP/DID? provider for making outbound calls? Will I be provided with an incoming number reachable by normal PSTN numbers? If yes, where will the number reside, as in will PSTN numbers calling me be charged extra? Real Deployments: Supposing I'm to do a real deployment for a client. What are the options that I have for hardware? 0) Get IP phones that talk SIP? Is this the most expensive option? 1) Suppose the client has a traditional plain intercom installment(think hotels etc). with phones connecting via RJ11 connectors. Is it possible to have something like an ATA with lots of ports working as a hub/switch, So that all phones can be plugged into ATA and managed via FS? Thanks PS: If the above hardly makes sense, pardon me, you can understand my confusion =). FS has really got me hooked and I'm itching to do more with it.

3 3

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

FS to answer a call with ZRTP and do text-to-speech of SaS
by Fabio Pietrosanti (naif) 01 Apr '10

01 Apr '10

Hi all, i would like to call from the product my company is going to release (www.privatewave.com, mobile voice encryption with ZRTP for Nokia, iPhone, Blackberry, Android) a FS extension and, without the ZRTP enrollment, having the extension to be answered negotiating ZRTP and playing with Text to Speech the SaS . Is this possible with current FS/ZRTP integration? Fabio

2 3