July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Distortion on approx first 200ms of G722 prompts on DECT based CPE
by Keith Laaks 28 Jul '09

28 Jul '09

Hi All, I am testing a range of G722 capable DECT based CPE. With one range, I have noticed that the first 200ms or so of each separate prompt file being played back is played out distorted from the DECT handset. When having a normal conversation, the quality is excellent, but when accessing your vmail, all the individual audio files making up the menu choices exhibit the distortion, which is pretty annoying. The same unit using G729, alaw or ulaw works 100%. I wonder if anybody else has uncounted this issue? My guess at this point - There may be a short break in the RTP between the separate files being played out by FS that makes up any menu. During this time the DECT handset's AGC probably goes to MAX amplification (as its not receiving any input during the short break in RTP). Then, when the RTP returns at the start of the next file, the AGC boosts the audio into clipping zone and takes 200ms to dampen down back to normal good levels. Looks like in these devices the G722 encode/decode is actually done in the DECT handset and not the voip-base unit. Is there any parameter that can be set in FS to ensure that the RTP keeps flowing, sending 'silence' between prompts ? Would be interesting to validate the above 'guess'. Best Regards Keith

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AbsoluteTimeout: The FreeSWITCH way
by Kristian Kielhofner 28 Jul '09

28 Jul '09

Hello everyone, I need to set a maximum call duration. What is the current recommended way to implement this in FreeSWITCH? I'm looking for something similar to AbsoluteTimeout() in Asterisk. Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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DTMF confusion
by Jesse Peterson 28 Jul '09

28 Jul '09

Hello, If I wanted a bridged call to a gateway to use inband DTMF for incoming recognition and outgoing generation I'm unclear on what to do because the wiki clearly states[1] not to use the "start_dtmf" and "start_dtmf_generate" together for cause of loops. Wouldn't it be technically possible to generate DTMF only on the outbound leg and recognize DTMF only on the inbound leg without interference? I assume I'm not understanding something correctly here - could somebody elaborate? The end goal for me is to detect the absense of "telephone-event" rtpmap and enable inband DTMF from a gateway. Thanks! - Jesse [1] http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf

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SIP instant messaging presence signaling doesn't work.
by Gregory Charles 28 Jul '09

28 Jul '09

Hi everybody, I intend to use Freeswitch with two Ekiga Softphones. SIP Instant messaging works between the two softphones but SIP presence signaling is not managed by the softphones. I try to use other softphones (QuteCom and SIPCommunicator) and it is the same. I have the following error in my FreeSwitch console: [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete subscriptions for failed notify Is there any special configuration for SIP instant messaging presence? Thanks. G.C. Hi everybody, I intend to use Freeswitch with two Ekiga Softphones. SIP Instant messaging works between the two softphones but SIP presence signaling is not managed by the softphones. I try to use other softphones (QuteCom and SIPCommunicator) and it is the same. I have the following error in my FreeSwitch console: [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete subscriptions for failed notify Is there any special configuration for SIP instant messaging presence? Thanks. G.C.

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Connecting FreeSWITCH with Asterisk
by velusamy velu 28 Jul '09

28 Jul '09

Dear All, I have tried to connect the FreeSWITCH with Asterisk I have followed steps which is provided in the following link, http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk I have tried to call "2000" from FreeSWITCH, but I have received the following message in Asterisk console "NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user "Velusamy" <sip:Velusamy@192.168.6.94<sip%3AVelusamy(a)192.168.6.94> >;tag=69Q648H9NjrSK" I have read "Using Authentication" topic in the link, But I did understand that topic.. They have mentioned "HOSTNAME.DOMAIN.COM" in that topic. Which hostname I have to specify here? Please help me.... Regards, Velusamy.K

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Multiple gateways: Priority the first bridge with prefix
by Edmar Cruz 28 Jul '09

28 Jul '09

Hi FS Users, I just want to try multiple gateways. It works actually like this... <action application= "bridge" data="sofia/sip/51(a)222.333.444.555 | sofia/sip/63(a)111.222.333.3333"/> But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-p… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Dial plan contexts
by Jason White 28 Jul '09

28 Jul '09

Has anything changed in the handling of dial plan contexts recently? As of rev. 14363, the context setting in the Sofia profile seems to be overriding the context setting in the user's definition in the directory. As per the default configuration, I have user-context set to public in my internal profile, my user has its context set to "default", but calls made from the phone registered to that user ID end up in public context when they reach the dial plan. Either something has changed or there's something wierd in my configuration that I haven't tracked down. I haven't made any changes to any of the profiles or users recently, though, and it was working under an older revision.

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Operation has no matching challenge
by Jesse Peterson 27 Jul '09

27 Jul '09

Hello, I am getting some SIP registration problems with FreeSWITCH 1.0.4pre10 and my ITSP (Vitelity). The error in the logs is such: 2009-07-27 08:54:48.217815 [ERR] sofia_reg.c:1460 vitelity-out Registration Failed with status Operation has no matching challenge [904]. failure #56 While these errors are happening the gateway state (via "sofia status") is FAIL_WAIT. With the ever-increasing back-off wait (60, 90, 120, 150, ..., seconds) the registration never resumes. Now one might suspect that there is something wrong with the configuration/ authorization but this problem is intermittent: a simple "sofia profile external restart" restores the registration and all is well (state turns to REGED) and of course the initial registration succeeds just fine, too. Quite an annoying problem as you never quite know when your gateway is registered when you pick up the receiver of your phone giving the impression of unreliable service! I suspect this to be the same problem, but with a different error message, that has been reported before[1][2]. Thoughts? Anything I should try? Thanks, - Jesse [1] http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011269… [2] http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/00627…

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linksys spa962+spa932 blf, hold and intercept of ringing extensions
by Vladimir Elizarov 27 Jul '09

27 Jul '09

Hello. I am trying to configure the linksys spa-932 (at http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932 stated that he works with freeswitch). I said "Server-Type option set to" RFC3265_4235 "" added to the unit 1 key 1 string: "fnc = blf + sd; sub = 1002(a)pbx0.test.lan; nme = test". The button blinks orange. if call on 1000 (spa962). This subscription runs spa932 and starts to show the status of the phone 1002. Thanks.

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IAX configurations
by velusamy velu 27 Jul '09

27 Jul '09

Dear All, I have tried to call a Asterisk extension from FreeSWITCH. I have done the following configurations, * I have enabled mod_iax module in modules.conf.xml file. * Next I have configure following extension in dialplan. <extension name="voipjet"> <condition field="destination_number" expression="^(222)$"> <action application="bridge" data="iax/222:222@192.168.6.94/$1"/> </condition> </extension> * Next I have configured a 222 user in sip.conf file at Asterisk machine. * I wrote dialplan for that extension in extension.conf file. When I tried to call 222 from FreeSWITCH, I have received following error in Console. "[ERR] switch_core_session.c:255 switch_core_session_outgoing_channel() Could not locate channel type iax" What would be the problem? Is there any configuration I missed? Please help me ..... Regards, K.Velusamy.

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