search for: netzquadrat

Did anybody figure out how to make dial detect a busy on a zaptel channel on a pri interface when using overlap dialing? According to the documentation dial should return to priority n+101, if the called party is found to be busy. I can see a DISCONNECT message with "user busy" coming from the network when I turn on pri debugging, but the dial application does not seem to notice.

I see in the changelog that ENUM support has been added.. anyone know what this is? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

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...fixing it. By now, I have seen a E400p filled up with ppp dial-up connections running stable and smoothly. Let me say that I am truly impressed with the kind of commitment Mark has shown in the process resolving this issue. This is unheard of for most vendors I worked with in the past. Thilo -- [netzquadrat] GmbH | Thilo Salmon Ronsdorfer Str. 74 | Fon: +49 211 302033 0 40233 Duesseldorf | Fax: +49 211 302033 22

Hi all, Hope somebody can help-I really am stumped as to why this won't work. I realise that this isnt an Asterisk problem (Please dont bash me on the list) and I have emailed the SER list but I havent received a reply and maybe someone on this list can help...Once this problem is solved I am going to use Asterisk for voicemail etc with SER (I have it set up) I currently have SER set up and

Hello, Does anyone know what driver to use with an "Eicon Diva Server 4BRI" card? My modem.conf: -------------- [interfaces] context=remote driver=i4l msn=16453 device => /dev/ttyds01 -------------- I get the following error: -------------------------- == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem Driver) WARNING[8192]: File chan_modem.c, Line 383

Did anybody ever come across an affordable, residential cable/dsl router with support for QoS? The ones I've seen so far (Netgear, D-Link and W-Linx) do not seem to support it. I noticed that even email can damage a G.711 stream on an 128kbit uplink, leave alone file-sharing applications. I understand this is strictly related to *, but nevertheless of interest to many of us. Thilo

Hi, did anybody ever try to make asterisk 'switch' calls on pri interfaces? My goal is to set up * to 1. wait on incoming calls until an extension is matched, 2. dial out and then 3. "pass through" the signaling until the call is either established or released I found that asterisk would first pick up the incoming call and then dials out. For my application however this

I have a number of local users who can dial out on a pri channel using the fantastic new overlap dialing feature. I would like to add a speed dialing feature, such as 1. User picks up and dials out (dial startet with option 'H') 2. User hangs up call with '*' 3. Dialed number is stored in a variable 4. User dials a two-digit extension followed by the # sign to save the stored

What would you suggest for a usb phone? I'm thinking a real phone - not an adapter. The kind of thing which is small enough so you can take it on a trip to place VoIP calls from your laptop. Something that ties into asterisk on linux and (more importantly to be honest) a softphone on XP or even its own client. Browsing the web I found that most places either sell them by the truckload or

An application I am running provides a dial tone to my users, read 9 digits, checks whether or not the called party number should be allowed and then dials out using overlap dialing on a pri channel. I.e. exten => _XXXXXXXXX,1,AGI(pm-check-destination.agi) exten => _XXXXXXXXX,2,Dial,Zap/g1/BYEXTENSION|60|CH The AGI-Skript takes about 0.3 to 0.5 seconds (it does a number of rather complex

In order to bill some of my customers I am trying to convince * to write a full set of cdrs. I found that * does not log cdrs outgoing calls on a zap pri channel when incoming and outgoing calls share the same span. So far I tried to play around with the 'C' option for the dial application and tried to set overlapdialing to 'no'. Interesting enough calls coming in on IAX2 or SIP

Is anybody aware of a database containing the types of nat implementation in todays soho/consumer routers? I think it would make sense for the community to have this database in order to avoid symmetric nats. If one such thing does not exist how about starting this database? A stunclient for linux can be found at http://sourceforge.net/projects/stun/ I can contribute this information for two

Hi everyone, any chance * could be used as a b2bua without forcing the media stream through the same box? I would love to do some computing on incoming calls, do things like setting another callerid and the forward the call to another sip UA - all without any audio traversing the * box. Any ideas? Thanks, Thilo

Hi, 1) is there a function like "zap destroy channel" to destroy sip channels? Zap/10-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/-081aee08 (pstn-out s 7 ) Ring Dial Zap/g1/0123456789|50|g Zap/8-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/-081aee08 (pstn-out s

Szenario: UA(Grandstream) => PROXY(SER) => GATEWAY(*) => PSTN After sending the SIP ACK From Gateway (*) ACK sip:123456@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK86c0bd474ea746b5 From: "Me" <sip:123456@mydomain.de>;tag=0f63d269bc25545d To: <sip:100@mydomain.de>;tag=as05df60b5 Contact: <sip:100@192.168.0.1> Call-ID:

Hi, Some ISDN Information has already been made available to asterisk eg. {$CALLERID}. Since my C knowledge is very basic yet I'm trying hard to find out how asterisk gets this Information from libpri. I guess it could be found in chan_zap.c ?? Does anybody have more in deep knowledge of the asterisk and libpri sources. Any hint is apreciated thnx Moritz

Hi, we are using ser as registrar and proxy, * as gateway. Can someone explane me the * NOTICE Message "chan_sip.c:7380 do_monitor: Disconnecting call 'SIP/sipgate.de-08352520' for lack of RTP activity in 0 seconds" We got a lot of these messages and Call Hung ups right after the Notice. Greets Markus

While using Blindtransfer #Extension everything works fine. But how do i activate announced transfer with an Grandstream GPX2000 ? Greets Markus

Hi, is it possible to send 2 CallingNumbers with libpri/asterisk => wct4xxp? I see it while receiving Calls. < Calling Number (len= 6) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) < Presentation: Presentation permitted, user number not screened (0) '0712345678' ] < [6c 0d 01 83 30 32 31 31 37 37 30 35 39 38