search for: ect

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 XXXXXXXXXXXXXX, but it is there. But sometimes i get this: calldate | clid | src |...

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

...Good C Knowledge, I would them , time permitting , to possible work together on this over IRC. I guess it would proable teaching , acualy it is teaching. Also, Anyone know of any good C Tutorials. Although codecs aren't very large to work on, I'm just feeling overwhelmed by HWND classes ect ect. Any Help , or tutoring would be greatly apriciated. So just e-mail me, or over on IRC on #vorbis (I'm cardboard) just Private Message me or something. Thanks to everyone on their helping me get CVS working and answering my constant barrage of newbie spam. Thanks for reading, ***** Ant...

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*...

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 Reported...

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 video codec thread and I have 30 gigabytes of HD space total, sitting around doing nothing :) So if some sort of storage is needed for any sort of videos, anything. I'm the man (I think :)...

...192.168.192.163, RULE=192.168.199.22, RULE=*:179, And here''s the tcpdump result for the traffic : 09:57:36.875357 192.168.199.22.1843 > 192.168.192.163.179: P 1280098168:1280098187(19) ack 1407875708 win 34752 <nop,nop,timestamp 58101584 523141495>: BGP (KEEPALIVE) (DF) [tos 0x2,ECT] 09:57:36.875730 192.168.192.163.179 > 192.168.199.22.1843: . ack 19 win 5792 <nop,nop,timestamp 523201215 58101584> (DF) 09:57:37.166515 192.168.192.163.179 > 192.168.199.22.1843: P 1:20(19) ack 19 win 5792 <nop,nop,timestamp 523201505 58101584>: BGP (KEEPALIVE) (DF) [tos 0x2,ECT...

...Good C Knowledge, I would them , time permitting , to possible work together on this over IRC. I guess it would proable teaching , acualy it is teaching. Also, Anyone know of any good C Tutorials. Although codecs aren't very large to work on, I'm just feeling overwhelmed by HWND classes ect ect. Any Help , or tutoring would be greatly apriciated. So just e-mail me, or over on IRC on #vorbis (I'm cardboard) just Private Message me or something. Thanks to everyone on their helping me get CVS working and answering my constant barrage of newbie spam. Thanks for reading, ***** Ant...

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

...s in to one of the boxes and is answered. After 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...

> -----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.

...ypads. (Believe it or not, before the advent of cell phones some businesses provided a "phone patch" interface to allow making phone calls from their 2-way radios. This is still quite common in the railroad industry, AFAIK.) In my case, I need it because it's how my PBX does disconnect notification to the voice mail system. When the line is hung up, it sends a "D". I expect these digits are very rarely used, which is probably why no one thought to document that Background() ignores them.

> -----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 busycount=10 but every voicemail > > has an annoying busy signal tacked onto the end of it. > > CVS HEAD has a feat...

...nstallation 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 delay when I do a 'net user/group'. It takes may be 20 seconds before prompting me the password and another 20 seconds or so to give the result(which is correct). Still struggling with the winbind thing and hope some experts can hel...

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 the channels are free again. Is it possible and with what card vendor (mISDN vs.zaptel) and how do I do that? Thanks in advance, Christophorus

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 0x...

...Fedora 23 x86_64 Linphone (linphone-3.6.1-10.fc23.x86_64) Firefox 43 (firefox-43.0.3-1.fc23.x86_64) Google Chrome (google-chrome-stable-47.0.2526.111-1.x86_64) SIP.js 0.7.2 I set up my SIP configuration to have two SIP accounts. Account 1000 is the Linphone and 1001 is the webrtc: [general] faxdetect=no vmexten=*97 context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes useragent=FPBX-2.11.0(11.20.0) disallow=all allow=g723 allow=ulaw allow=gsm allow=alaw allow=g729 allow=speex allow=g722 allow=h264 allow=h263p all...