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FS does not respond to second 401 Unauthorized that has new nonce during registration with gateway
by Mervyn Yeo 25 Mar '11

25 Mar '11

Hi Everyone, I'm trying to register FS with a gateway but it's been unsuccessful so far with the last message being 401 Unauthorized. I'm quite certain that the configuration in the gateway xml file is correct. The same username, password, etc has been used with X-Lite, Zoiper, Yealink, Cisco phones and it's been able to register. I've compared SIP traces of FS's registration and all the others and it seems that FS is not responding to the last 401 message from the other end even when the it's been challenged with a new nonce. What I found out was that the second 401 has an additional domain parameter as well. Are all the other clients being lenient and is not following the RFC or is this a bug in FS? SIP trace http://pastebin.freeswitch.org/15831 Configuration http://pastebin.freeswitch.org/15832 Mervyn

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Event-lock, sync/async on outbound ESL and execute() tagging question
by Dmitry Sytchev 24 Mar '11

24 Mar '11

Hi All I'm trying to understand setEventLock and setAsyncExecute parameters on outbound ESL socket. My aim is to have ability to choose execution method of freeswitch dialplan command via execute(). I'm using async full, but I want to use sync mode for some applications for which I don't want to receive events, for example. Is there a way to run application in sync mode and just get return value, and discard its events? I want this because If later I will run some app in async mode, I'll get previous events like 'CHANNEL_EXECUTE_COMPLETE' and so on. Another approach is to have ability to mark calls to execute or API with some tag to distinguish between events from previous execute() or api() calls, like with BGAPI job id. What can be done to achieve this? Or maybe I should just process all events by myself if I use async? And finally, could anybody explain the Event-Lock / AsyncExecute difference? Seems I can't really understand the wiki notes about that. -- Best regards, Dmitry Sytchev, IT Engineer

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mod_conference and max-members
by Peter Steinbach 24 Mar '11

24 Mar '11

Hello, when we open a conference and set the value of "max-members" to 5 then only 5 members will be able to join the conference. This is fine. But just in case the conference moderator wants to invite an additional person: Is it possible to temporarily disable this function and to add (invite) another member to the conference? What we have tried so far: * 5 members dialed into the conference * changed number of members temporarily to 6 (via XML-Curl) * Invited another member into the conference (potential 6th member) * Freeswitch requests the new conference.conf XML with "max-members" == 6 * but still it shows "Conference is full" in the logs to the potential 6th member Is there any other way to overcome this, maybe by sending some api command via event socket? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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member_id of conference
by Madovsky 24 Mar '11

24 Mar '11

Is conference <conf> list the only way to get a specific member_id ? Thanks much

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voicemail sounds format for internal and external sip profiles
by Madovsky 24 Mar '11

24 Mar '11

Is there a way to configure voicemail sound format to internal and external profiles independently ? I'd like for example set ".spx" for inbound calls and ".wav" for outbound calls. thanks

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Re: [Freeswitch-users] Bug ID: FS-2868 Status?
by Kim Culhan 24 Mar '11

24 Mar '11

On Wed, March 23, 2011 9:15 am, Michal Bielicki wrote: > The problem is a bug in solaris perl development package that I haven't been doing > anything about since I do not use perl for roughly a decade. > Its not the hash module, its the perl module. > > Am 23.03.2011 um 11:44 schrieb A E [Gmail]: > >> Hello FreeSwitchers! >> >> Am attempting to compile the latest (off of Git) code for FS on Solaris 11 Express >> on an 64-Bit x86 machine. However, I'm running into the error that Bruce >> McAllister has reported on 23rd Nov 2010. The bug is still sitting unresolved. So >> I'm assuming there's a backdoor trick to fixing this OR that some other version of >> FS code doesn't have this issue. Could someone please let me on the secret on >> which build compiles on x86 Solaris v10+ and/or what the backdoor trick to bypass >> this is? Should I not compile the hash module where this bug seems to be >> originating from? I'm seeing the same error here with the perl module commented-out in modules.conf Please describe the workaround you are using to avoid the problem caused by the perl module. thanks -kim

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Gawk.exe crashes when building on Windows 7 64bits w/ VS
by Frankie Yiu 24 Mar '11

24 Mar '11

I still got the compiliation error of Gawk.exe. Could that be other setting? Thanks, Frankie ---------- Forwarded message ---------- From: Jeff Lenk <jeff(a)jefflenk.com> To: freeswitch-users(a)lists.freeswitch.org Date: Thu, 24 Mar 2011 06:48:30 -0700 (PDT) Subject: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows 7 64bits w/ VS http://freeswitch-users.2379917.n2.nabble.com/file/n6204157/New_Bitmap_Imag… See attached image. I recall an issue with this in the past but I cant find the thread. Try building with the locale set to english once to generate the sofia gawk files then you should be able to switch back. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Re-Gawk-exe-crashes-when-buil… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Event Lock and Bind Meta App
by Moe Navid 24 Mar '11

24 Mar '11

Hi all, I noticed when bridging the call in event socket and adding event-lock:true, the lock prevents the bind_meta_app to get executed while on the call it waits till bridge is done and then executes. Any suggestions or work arounds? Thanks Moe

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Re: [Freeswitch-users] invite in conference
by Madovsky 24 Mar '11

24 Mar '11

for those who need caller_id_name effective in conference from an invite, it has to specify caller_id_name in brackets conference testconf dial {originate_timeout=30,caller_id_name='bibi baba'}sofia/default/1000@softswitch 1234567890 FreeSWITCH_Conference ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, March 24, 2011 11:12 AM Subject: invite in conference I'm experimenting hot invite in conference. so I used the example on wiki conference testconf dial {originate_timeout=30}sofia/default/1000@softswitch 1234567890 FreeSWITCH_Conference while the conference testconf is running. I noticed that: - my TTS pronounces the extension number rather than caller_id_name (if someone enters in conference normally the TTS is right) question: - how to hangup the channel in case of answer machine or busy for example while the caller is in conference ? Thanks Franck

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Core dump on unimrcp
by Seven Du 24 Mar '11

24 Mar '11

I'm using a recent version of FS unimrcp, and tried to connect to a mrcp V1 server from a China vendor. Core dump after the log: http://pastebin.freeswitch.org/15830 I traced the core and found that in mod_unimrcp.c around line 1824 if (schannel->type == SPEECH_CHANNEL_SYNTHESIZER) { descriptor = mrcp_application_sink_descriptor_get(channel); } else { descriptor = mrcp_application_source_descriptor_get(channel); } where mrcp_application_sink_descriptor_get(channel) returns NULL. configurations is fairly "standard". anyone could help take a look is appreciated. Let me know if I need open a jira with update the the latest version and backtraces. Thanks. Seven.

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