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Hi, since version 1.4.12 the libpri package supports ETSI Explicit Call Transfer feature: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.12 Does anyone know, how to use this feature in the dialplan? I can not find any hints in the asterisk doc. Best regards, -Thorsten-

I have a simple cdr configured with the default tables, here is a row of a good cdr report calldate | clid | src | dst | dcontext | channel | ect ..... ect .... 2008-08-29 10:16:49 | "C. BOUTON" <40> | 40 | XXXXXXXXXXX | phonesystems | SIP/40-08776938 | ect ..... ect .... I have replaced the number by

Hi all, i'm a bit confused, can i setup samba (3.0.30) with LDAP backend, and have the "posix/local linux" users and groups reside in the /etc/groups /etc/shadow ect. ect (the standard linux files) ??? or do i have to put them in ldap also ?? (is there a choice?) Greets, Collen

Hello, I'm just letting everyone know that I've begun the preliminary part of putting together a working comment system. I have a few idea's on how to implement this system. 1. What I would like to do is grab the General and Input SDK's for Winamp first. Then I would like to add to our existing Winamp plug-in the ability to simply add comments via the built in ID tag

Hey Guys Im trying to find out how to transfer a call with ECT (explicit call transfer) ? Im currently transferring a call as following: exten=>2,1,Dial(capi/720****:078888****,18) exten => 2,2,Goto(2-${DIALSTATUS},1) exten => 2-NOANSWER,1,Dial(capi/720****:07979****) exten => 2-CHANUNAVAIL,1,Goto(1,1) exten => 2-BUSY,1,Dial(capi/720****:07979****) If I wanna transfer a call with

https://bugzilla.netfilter.org/show_bug.cgi?id=928 Summary: ECN: --ecn-tcp-ece and --ecn-ip-ect is not supported Product: nftables Version: unspecified Platform: x86_64 OS/Version: Debian GNU/Linux Status: NEW Severity: normal Priority: P5 Component: nft AssignedTo: pablo at netfilter.org

Hello, I've been real busy latley and have been unable to spend on time on vorbis [or IRC for that matter :)] I'll be getting some spare time soon so I will be back in action. Anyhow, I found this /excellent/ website of audio testing for soundcards, Pro DAC, I/O Video cards ect. The operator has this GREAT suite of tools. http://www.pcavtech.com Second, I was browsing over the

Hi, I have successfully shape bandwidth using htb.init using ip address , but when i try to shape zebra BGP using their ip address and BGP port it can''t match the class for BGP and always get the default class. Here''s my htb.init script in the bridge : #eth0-2:50.bgp RATE=128kbit RULE=192.168.192.163 RULE=192.168.199.22 RULE=*:179 #eth1-2:50.bgp RATE=128kbit

(Just a copy of the message I sent to Vorbis, better to send it here.) Hello, I'm just letting everyone know that I've begun the preliminary part of putting together a working comment system. I have a few idea's on how to implement this system. 1. What I would like to do is grab the General and Input SDK's for Winamp first. Then I would like to add to our existing Winamp

I have been looking around for VoIP providers but have not found a good listing. Is there no "yellow pages" for VoIP providers? Google mostly returns services like Vonage, Packet8, NuFone, ect. None seam to be very reseller friendly and none offer LNP or local DID's for my area. Anyone know of a list (even a partial one) Jeromie Reeves

...some interaction with the caller, the box decides that the call needs to be handled by the other box. I don't want just to relay the call through to the second box using IAX or SIP or an additional PSTN channel. What I would like to do is to redirect the call in the PSTN so that it ends up connected only to the second box. That is why each box would have its own telephone number as well as the global access number for all boxes. Obviously, as well as the ability to redirect the call transparently to a channel on the second box, I also need the second box to be able to identify that this call...

> -----Original Message----- > From: David Brodbeck [mailto:DavidB@mail.interclean.com] > Okay, the problem appears to be that I'm tone deaf. ;) > > I finally thought to turn on debugging on the channel. The > PBX is sending > "D", not "*". The programmer of the previous voice mail system (whose > configuration I was cribbing from) seems to have

> -----Original Message----- > From: Gilad Ben-Yossef [mailto:gilad@codefidence.com] > I'm prbably stupid, but wont this do what you want? > > > exten => 1,1,Goto(bye,s,1) No, because I wanted to match on "D", not "1". Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't

...it out. The extension was working, but > Background() > > ignores the tones A through D by default. I didn't realize > this because I > > wasn't waiting for message playback to finish. > > Please enter a bug in Mantis for this; it should very likely be > corrected, as I don't see any reason to ignore A-D in Background(). http://bugs.digium.com/bug_view_page.php?bug_id=0003538 A simple fix is included, though I don't have a deep understanding of the Asterisk code, so it's possible it has side effects I'm not aware of.

> -----Original Message----- > From: Paul Rodan [mailto:asterisk@glitch.cc] > I'll ask a stupid question, how does a user hit an alpha > letter from his touchtone? > > I know that the Cisco 7960's support entering alpha letters, > and it could > potentially do it (maybe), but how does the average end user > enter an a b c or d from their touchtone phone?

> -----Original Message----- > From: Chris Wade [mailto:clwade@sparco.com] > Brian Roy wrote: > > I think that my PBX does this too. Is there any way I can get the > > Zaptel drivers to disconnect on that tone too? I would love > to replace > > my existing voicemail with * but I can't get my PBX to signal a > > disconnect properly. I have to use

I have seen the same thing in my two installation of beta2 and rc4. Some how, I forgot what I have played around in beta2 and was later been able to do a 'getent passwd DOMAIN\\username' but can't repeat that again. in my latest installation of RC4. I am wondering if it has anything to do with the timing of locating the correct domain controller to logon. As I am experiencing very long

Hi list, I am searching for a possibility to do a certain call transfer method which is called "path replacement" in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and

Hi I have done everything richard told to do ECT . below is my trace, anyone can help ? -- DAHDI/i1/09123278669-4 answered DAHDI/i1/88050048-3 -- Native bridging DAHDI/i1/88050048-3 and DAHDI/i1/09123278669-4 PRI Span: 1 Adding facility ie contents to send in FACILITY message: PRI Span: 1 ASN.1 dump PRI Span: 1 Context Specific/C [1 0x01] <A1> Len:11 <0B> PRI Span: 1

...ox or Google Chrome into asterisk, out to the SIP softphone. The problem arises when I try to make asterisk send a call into the browser. When using Firefox 43 I can receive the call normally (this required patching around ASTERISK-25659) and all is well. However, in Google Chrome, the call is rejected with a message of "Failed to set remote video description send parameters.." as shown in this SIP trace in the browser console: Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | received WebSocket text message: INVITE sip:8dgpkoa2 at 192.0.2.210;transport=wss SIP/2.0 Via: SIP...