- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Auto test FS install - now necessary?
by Jerre Cope 04 Feb '11

04 Feb '11

I second Mr. Ayre. Thanks everyone. Especially the Rocket Scientists. On 02/04/2011 11:49 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: Auto test FS install - now necessary? (Steven Ayre) > 2. Re: Auto test FS install - now necessary? (Steven Ayre) > 3. Re: Auto test FS install - now necessary? (Phone) > 4. Re: Auto test FS install - now necessary? (mazilo) > 5. curl: parse XML response? (james.royer(a)gmail.com) > 6. mod_directory multi-tenant (Erick Baum) > > > _______________________________________________ > 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

1 0

context
by Sam 04 Feb '11

04 Feb '11

I have defined a user as : <user id="2099"> <params> <param name="password" value="1234"/> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> <param name="mwi-account" value="2099@voicemail"/> <param name="vm-password" value="1234"/> <param name="vm-email-all-messages" value="true"/>   </params> <variables> <variable name="user_context" value="inter"/> <variable name="force_transfer_context" value="inter"/> <variable name="accountcode" value="2099"/> <variable name="internal_caller_id_name" value="2099"/> <variable name="internal_caller_id_number" value="2099"/> <variable name="callgroup" value="1"/> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> </variables> and in the dialplan i have <context name="inter"> <extension name="2099"> <condition field="destination_number" expression="^(2099)"> <action application="set" data="call_timeout=30"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@ ${domain_name}"/> </condition> </extension> </context> Now extension 2075 is also made on above lines in the same context for user and dialplan, here the call when initiated from 2075 to 2099 searches for context public instead of context inter mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public any reason why ? Regards Sam

3 10

Wrong codec used on outbound leg
by Marcin Wojtowicz 04 Feb '11

04 Feb '11

I have the following setup: voip.ms < (external profile) > freeswitch < (internal profile)> my devices. On my internal profile, the only inbound and outbound codec I allow is iLBC (I tried 20ms and 30ms), and on my external profile the only inbound and outbound codec I allow is PCMU. All my internal settings are compatible with the devices that I'm using, and everything works well when I call from outside in, and freeswitch now correctly transcodes from PCMU to iLBC. However, when I call from my internal devices to outside, the iLBC codec is offered on outbound leg B, not PCMU like I have on my external profile. Looking at wireshark voip call flow, my machine is sending iLBC to voip.msand voip.ms is sending me PCMU RTP.

3 3

Re: [Freeswitch-users] Auto test FS install - now necessary?
by Peter Olsson 04 Feb '11

04 Feb '11

I agree, I always follow the commit log, and usually I get the big idea about what's going on. /Peter ----- Reply message ----- Från: "Steven Ayre" <steveayre(a)gmail.com> Datum: fre, feb 4, 2011 23:10 Rubrik: [Freeswitch-users] Auto test FS install - now necessary? Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Best source of that information is the commit log http://fisheye.freeswitch.org/changelog/freeswitch.git There's also a changelog file that's periodically brought up-to-date. -Steve On 4 February 2011 15:58, Tim St. Pierre <fs-list(a)communicatefreely.net<mailto:fs-list@communicatefreely.net>> wrote: > > Testing yourself is always a good idea before rolling out to a > production system. That'll show any obvious ones up that can be fixed > quickly. Not that this will always catch everything, but is there a "release notes" or some other simple document that outlines what parts of Freeswitch have recently been changed? We ended up building a sort of beta-test network, with a second server and some customers that are very benevolent about this sort of thing, and if I had a simple checklist of things that had been worked on, I can make sure that I test those areas more closely. I don't have any experience working with software development beyond my own little LUA and PHP scripts, so maybe this is already available and I just don't know where to look. Is there a tutorial somewhere on how to be a good user, from a developers perspective? -Tim _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4d4c248132761067889338!

1 0

iLBC to PCMU transcoding broken on windows?
by Marcin Wojtowicz 04 Feb '11

04 Feb '11

I built freeswitch from the latest git on my windows machine, and it looks like transcoding from iLBC to PCMU is broken, and I hear only squeal on the line. I have only PCMU on my external profile and iLBC on my internal profiles, so no codec mismatch is possible, and all my codec settings are compatible with the devices that I'm using. I inspected the traffic flow with wireshark, and there is nothing incorrect there, SIP messages and SDP is ok, and traffic looks like my device <-> iLBC <-> freeswitch <-> PCMU <-> voip.ms. I've searched around and it looks like my issue is identical to the discussion here: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/063891.…. If I run PCMU on both ends, there are no problems and if I call into freeswitch voicemail on iLBC, there are no problems either.

5 11

FS 1.0.7 crash on originate
by Anirudha Shimpi 04 Feb '11

04 Feb '11

Just downloaded (from git) and installed FS 1.0.7, however, every time I do originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.

2 1

H.323 implementation on FS
by Vedran Zeljeznak 04 Feb '11

04 Feb '11

hi everyone, do any of you have any suggestions which H.323 implementation (mod_opal or mod_h323) should be used on a production Freeswitch platform (HEAD version)? Is it better to forward calls from H.323 trunk through Yate (configured as H.323 to SIP proxy) before terminating them on Freeswitch? --- Vedran Zeljeznak

1 0

How to use sofia_gateway_data ?
by David Ponzone 04 Feb '11

04 Feb '11

All, I am still trying to reroute a call in case a gateway is down, but I decided to do that another way. I'd like to detect my gateway is down, and I was trying to use sofia_gateway_data function for that purpose. But I can't figure out the expected args. I try: sofia_gateway_data <gatewayname> var Status or ivar/ovar instead of var I tried with both a gateway I register to, and an IP-auth-based gateway I ping regularly. So I am stuck now, and I can't find any info anywhere on how to detect a gateway is flagged as DOWN. TIA David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

2 7

deflect
by Sam 04 Feb '11

04 Feb '11

Hi, When call comes on 1 server and plays an application and after execution of the application the call is bridge to the other server ,but here after bridging the call should refer/deflect to other server, how this can be done ? Here just using the deflect variable is not recommended as there is proxy in between, so once the call is bridge the next step would be deflect the leg totally to another server via proxy. Regards Sam

3 12

Strange behaviour with leg_timeout
by David Ponzone 04 Feb '11

04 Feb '11

It's my time to mess with XXX_timeout variables in FreeSWITCH. What I need to do is to route a call to a gateway, and if I don't have an answer (like not even the 100) within X seconds, I try to route the call to a backup number. I tried it this way: <extension name="gw_then_backup"> <condition field="destination_number" expression="^(XXXXXXXXXX)$"> <action application="bridge" data="{leg_timeout=5}sofia/external/$1(a)a.b.c.d|sofia/gateway/provider.out/YYYYYYYYYY"/> </condition> </extension> My understanding was that a.b.c.d would be tried for 5 seconds, and if no 100 is received, gateway provider.out would then be used. It works...but after the first INVITE to provider.out (which sends back 100/180/183), I can see some more INVITEs sent to a.b.c.d. I am wondering if I am doing something wrong, or if there is an issue. I use latest git from 5 minutes ago. Thanks David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

2 2