- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Realm Value
by Ali Al-Rubaie 17 Feb '09

17 Feb '09

Thanks Brian, Actually we're using freeswitch ver 1.0.2. Regards, Message: 5 Date: Thu, 12 Feb 2009 15:48:00 -0600 From: Brian West <brian(a)freeswitch.org> Subject: Re: [Freeswitch-users] Realm value To: freeswitch-users(a)lists.freeswitch.org Message-ID: <F7EFCA6F-FA3F-4CC1-8CEC-4AA56EDFF75D(a)freeswitch.org> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes What SVN rev? /b On Feb 12, 2009, at 3:41 PM, Ali Al-Rubaie wrote: > Hi, > > How can the default value of "realm" be changed? I had changed the > command: > > <param name="challenge-realm" value="150"/> > > in the file internal.xml but FS still uses the server IP address as > the challenge realm. > > Thanks in advance! >

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SIPX/FS Auto attendant
by Chris Jones 17 Feb '09

17 Feb '09

I'm having a problem that calls to the auto-attendant won't transfer. I know this has been a problem in the past but thought that it was fixed. Whenever I enter an extension (or press a key to transfer me to one), the call just hangs up. I ran Freeswitch on the console, but all I see happening is this: 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Ring-Ready sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116! 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Pre-Answer sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116! 2009-02-16 20:22:51 [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116] has been answered 2009-02-16 20:23:07 [NOTICE] sofia.c:3179 sofia_handle_sip_i_state() Hangup sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 1 ( sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116) Ended 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/papayaecs.bluewavecorp.local/8323160383(a)64.2.142.116 [CS_HANGUP] Does anyone have an idea as to how to fix this? The call is coming from a SIP Trunk on vitelity.

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Voicemail prompts playback too quickly
by Maxim Karp 17 Feb '09

17 Feb '09

Hello, Our voicemail prompts playback much too quickly on FS v1.0.1. Any suggestions to slow them down? Thanks, Maxim.

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Playing a G729 file as ringback
by Cesar Cepeda 17 Feb '09

17 Feb '09

Hi, I'm using FS with g279 on passthrough mode and I'm trying to play a g729 file as ringback to the A-leg while bridging a call. As far as I understand it should go something like this: . originate {channel_vars}dialstring . set some combination of values on 'ringback', 'transfer_ringback', 'instant_ringback' . bridge <bridge_destination> The bridge works correctly, but no matter what combination of values I try for the 'xxxringback' vars I never hear the ringback on the A-leg. Can you tell me what I'm missing? Thanks. Cesar Cepeda.

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FS SIP audio quality?
by Paul D. 17 Feb '09

17 Feb '09

Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip call, or call to VM prompt, or call via gateway to PSTN - FS audio volume level (should I say gain?) seems noticeably lower than on *, this may be a reason that FS audio seems to be subpar, more noise less clear. Test calls made using PCMU codec from X-Lite and Linksys 2002. Is there anything can be tweaked in FS to correct that? Same issue was with 1.0.2.

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Re: [Freeswitch-users] FS SIP audio quality?
by Anthony Minessale 17 Feb '09

17 Feb '09

The typing it takes to start a pcap of each call and email them is less than you have typed thusfar. Please just take the captures and send them to us to examine. That's all. If you have a real issue we would like to address it. On Feb 15, 2009 8:06 PM, "Paul D." <pauld(a)versafon.com> wrote: Well, I tried several call scenarios: 1. Call from X-Lite or Linksys to VM. 2. Call from X-Lite or Linksys to a conference. 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs. I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise grade Intel server. So just comparing audio in the call scenarios above * somehow does noticeably better job, sounds clearer and volume is at the right level. I am not changing any phone settings of course when switching between * and FS. I am not biased towards FS or * at the moment, though FS seems to have a better designed configuration options and community. Just wanted to share my experience, and hear some opinions. Unfortunately I cannot spend whole amount of time investigating this case now, capturing packets etc., but I will try to do that once I have time. Meanwhile I will have to stick to * for prod. Anthony Minessale wrote: > it's digital audio. The only thing doing sampling and reconstruction ... > <mailto:pauld@versafon.com>> wrote: > > Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Si... > <mailto:Freeswitch-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.f... > <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale(a)hotmail.com> > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> > <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale(a)gmail.com> > > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org<sip%3A888(a)conference.freeswitch.org> > <mailto:sip%3A888@conference.freeswitch.org<sip%253A888(a)conference.freeswitch.org> > > iax:guest@conference.freeswitch.org/888 > <http://iax:guest@conference.freeswitch.org/888> > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888(a)conference.freeswitch.org> > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-use...

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How to keep leg A from hanging up from a bridge when Leg B hangs up (and vice versa)?
by Simon Tang 16 Feb '09

16 Feb '09

Hello, I'm using event socket outbound and have a framework that does stuff based on the events that come back (this includes my own IVR). What I have now is an IVR system that allows 2 users to bridge to one another at will, and to 'unbridge' at will by catching DTMF events. I have 2 requirements: 1. When one leg hangs up during a bridge, the other leg is presented with the IVR 2. After a bridge, when one leg sends a DTMF tone, both legs will be presented with the IVR and no longer be bridged (they can bridge with other sessions again after this point if they desire) I have done multiple experiments by using netcat and 2 sessions. Here is what I have found: * The hangup_after_bridge variable does nothing for me. I've set it on both legs, but whenever one leg hangs up after a uuid_bridge, the other leg will automatically hang up * I've tried setting "park_after_bridge=true" on both legs, and this works to a certain extent. If one leg hangs up, the other leg will be parked, and I can present that user with my IVR. This meets requirement #1. However, requirement #2 won't be met because: o If I set "park_after_bridge=true" and one leg sends a DTMF tone to signal an unbridge, I will "unbridge" the legs by "parking" both legs and I am able to present them both with an IVR. If they decide to bridge with each other again (by selecting an option in the IVR), I will attempt to do a uuid_bridge and this will FAIL! (both parties do not hear each other.) In the simplest terms, I can't do "uuid_bridge uuidA uuidB", "park", "uuid_bridge uuidA uuidB". * With "park_after_bridge=false" (default), I can do "uuid_bridge uuidA uuidB", "park", "uuid_bridge uuidA uuidB" with no issues, meeting requirement #2. However, this will not meet requirement #1, because when one leg hangs up, it will trigger a hangup on the other. Please help. How can I meet both of my requirements? Thanks. Simon Tang Lead, Server Team Suite 706, 1155 Robson Street Vancouver, B.C. Canada V6E 1B5 T: +1.604.408.2228 Ext. 116 F: +1.866.874.8136 E: simon(a)airg.com <mailto:simon@airg.com> W: www.airg.com <http://www.airg.com> The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material communicated under NDA. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer.

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Build and Sofia issues with recent svn trunk revisions
by Jason White 16 Feb '09

16 Feb '09

Following the resampling discussion, I tried upgrading to revision 12018. which compiled cleanly, but then failed to load my internal SIP profile: [ERR] sofia.c:739 sofia_profile_thread_run() Error Creating SIP UA for profile: internal The same configuration works under revision 11488, to which I've temporarily downgraded. Either something has changed in FreeSWITCH that requires modifications to my SIP configuration, or this is a regression. I decided to try rev. 12027, which, on the same machine (Debian Sid) fails to build with the following error: x86_64-linux-gnu-gcc -c -ggdb -I. ./dftables.c In file included from ./dftables.c:50: ./pcre_internal.h:239:2: error: #error LINK_SIZE must be either 2, 3, or 4 make[2]: *** [dftables.o] Error 1 I suspect that recent changes to the build system are responsible for the latter. I'm sure these are minor matters that will be sorted out soon. Thanks once again to the developers for a great project!

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Location of config files - Debian packaging issue?
by Jason White 16 Feb '09

16 Feb '09

I've found the cause of my problem: As of the 12018 build, FreeSWITCH is searching for its configuration files in /etc/freeswitch rather than /opt/freeswitch/conf. I am using Debian packages built from a copy of the repository. If this is a deliberate change, it's fine, but if it isn't deliberate then something is amiss with the packaging.

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[newbie] Clean start with a simple configuration
by xs 16 Feb '09

16 Feb '09

Hi, I installed Freeswitch with this http://wiki.freeswitch.org/wiki/Quick_Startmanual and it works! But it has al sorts of options enabled: 5000,9995, 9996, 9999, 80+[group], 30+[conf], voicemail, something about pizza (00_pizza_demo.xml) and a lot of other xml files and directories (default.xml and default/) etc. etc. That is not what i want. When i install Asterisk (the only PBX that i have experience with) and i clean out sip.conf and extensions.conf and replace them with this: ---------------------------------------- sip.conf [general] [1001] username=1001 secret=password1001 type=friend context=phones host=dynamic qualify=yes disallow=all allow=g726 [1002] username=1002 secret=password1002 type=friend context=phones host=dynamic qualify=yes disallow=all allow=ilbc ---------------------------------------- extensions.conf ---------------------------------------- [phones] exten => _[1001-1002],1,Dial(SIP/${EXTEN},60) exten => _[1001-1002],n,Hangup() ---------------------------------------- then i can simply call from one (local) sip phone to another and force transcoding between them. Nothing else. I want that also with Freeswitch. It is a good starting point but i am fiddeling with it for a couple of days now, read the docs but i can't get Freeswitch to do just this. So just calling between a few local sip phones with transcoding and _everything_ else disabled. Can anyone help me a little? Thanks in advance!

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