RE: [Asterisk-Users] Advanced Conferencing optionswithout-of-treemodules?

A couple comments. I'm not a programmer, my C is passable, but my web development would have to grow by leaps and bounds to be considered poor. I pulled the Meetme2 from here: http://www.areski.net/asterisk-meetme/about.php?s=0 The app needs a minor tweak to compile against 1.0.5. I stumbled d

[Asterisk-Users] Problem with Asterisk Unable to allocate channel

Hello, yesterday when i wasnt in the office the asterisk server stop working, it was registering the sip terminals but cant make calls, because im not in the office i told the people to reboot the server to make the server works again but today i found this lines in the full log, can anybody tell m

Re: [Asterisk-Users] How does the g.729 registration program work?

Paul Fielding wrote: - Original Message - From: "Kristian Kielhofner" <[EMAIL PROTECTED]> Paul Fielding wrote: I thought that it was 3 different times with different nics, but with the same nic didn't count.*shrug*. No matter - if we can just copy the license keys that's much easier.

Re: [Asterisk-Users] How does the g.729 registration program work?

- Original Message - From: "Kristian Kielhofner" <[EMAIL PROTECTED]> Paul Fielding wrote: I thought that it was 3 different times with different nics, but with the same nic didn't count.*shrug*. No matter - if we can just copy the license keys that's much easier... :) Yes, that is t

[Asterisk-Users] Re: Grandstream and VLANs

"Wojciech Tryc" <[EMAIL PROTECTED]> writes: > I am having hard time to get VLAN tagging working with Grandstream > 101 phone. As soon as I enable tagging on the switch and configure > the phone to tag packets with corespodning VLAN ID #. I can not even > get IP anymore from my DHCP. Just a couple

[Asterisk-Users] snom220 *8 hangup

Good day all We have a snom 220 set as a switchboard phone I also configured *8 so that if the operator is somewhere else and it rings she can just go *8 on the nearest phone,Grandstrams bt-100 and snom 190.But If she does this she only speaks for about 30s and it will cut off the caller? Any ideas

RE: [Asterisk-Users] how to increase max number of simulatneousoutgoing calls

I just dealt with this similar type of problem with Teliax. You have to request unlimited simultaneous calls from them. They default to only 2 initially, I'm guessing to prevent you from eating up a lot of minutes. (this is with the PAYG plan btw, the monthly plans are probably limited to one

RE: [Asterisk-Users] setting up fromuser

Khan, This is essentially the same thing I was trying to make clear upon using 7960 phones. The fromuser variable in sip.conf is the entry that gets written to the SIP header "from" variable. There should be a way to make this fromuser dynamic to support these various situations. Does anyone el

Re: [Asterisk-Users] How does the g.729 registration program work?

Paul Fielding wrote: I thought that it was 3 different times with different nics, but with the same nic didn't count.*shrug*. No matter - if we can just copy the license keys that's much easier... :) Paul Yes, that is the case. But on the same hardware why not just copy the license files?

[Asterisk-Users] SNOM Call Diversion

I am just playing with a SNOM 190. Overall, I'm very impressed with the quality of the unit and the feature set. I am running the latest firmware (snom190-SIP 3.57u) and the asterisk CVS from last night (1/3/05).   The only problem that I've encountered so far is with Call Forwarding, which

[Asterisk-Users] chan_capi compile error on FC3

I want to use the Diva Server BRI card with chan_capi-0.3.5. Unfortunately, I got following errors when I make the chan_capi on Fedora Core 3. (gcc version 3.4.2 20041017 (Red Hat 3.4.2-6.fc3), kernel-2.6.9-1.667) FYI, I use the CVS Asterisk. Dose anyone find the solution to fix this problem? BT

Re: [Asterisk-Users] Advanced FollowMe or Forwarding Application Suggestions

On Mon, 28 Feb 2005 15:23:13 -0600 (CST), Kevin LaFata <[EMAIL PROTECTED]> wrote: > There is one final application I've been trying hard to find to replace > something we already use with another provider. It's kind of an advanced > "FollowMe" application. (freedomvoice.com) > > What I was wonder

Re: [Asterisk-Users] Advanced FollowMe or Forwarding ApplicationSuggestions

On Mon, February 28, 2005 4:50 pm, Alistair Cunningham said: > My company, , does Asterisk consultancy. We can develop, > install, and support such an application if you like. Rather poor form for a non-biz list don't you think? Next time, how about just contacting him off list yourself and avoid

[Asterisk-Users] strange CDR problem

I'm using all sip, no zap, or iax, so I don't know if the same problem exists when using zap, or iax. When device 1 calls device 2, and device 2 blind transfers to device 3, the following happens: The cdr record shows that 1 called 3, and ignores completly the fact that 1 called 2 and 2 called 3. I

RE: [Asterisk-Users] Advanced Conferencing options without-of-treemodules?

Please do share I am very interested in your mods and the web interface. -Original Message- From: Dan Austin [mailto:[EMAIL PROTECTED] Sent: Monday, February 28, 2005 7:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Advanced Conferencing o

Re: [Asterisk-Users] music on hold trouble

I too am having the same problem with CVS from last night. From my debugging, * never attempts to start MOH. Anyone else found this? - Original Message - From: Krystian Filiks To: asterisk-users@lists.digium.com Sent: Monday, February 28, 2005 1:46 PM Subject: [A

Re: [Asterisk-Users] How does the g.729 registration program work?

I thought that it was 3 different times with different nics, but with the same nic didn't count.*shrug*. No matter - if we can just copy the license keys that's much easier... :) Paul - Original Message - From: "Kristian Kielhofner" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing L

[Asterisk-Users] what is phone: Linux Telephony channel

need hardware ? can dial to PSTN? help me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aste

Re: [Asterisk-Users] DSL splitters and FXO signalling (hangup detectionproblem)

ADSL splitters should not effect disconnect signals from the telco. You don't mention your country, so we cann't even guess based on your location. Telco standards vary widely across this globe we call Earth. Lyle - Original Message - From: "Soner Tari" <[EMAIL PROTECTED]> To: "Asterisk

Re: [Asterisk-Users] where is voice conduits

on 2/28/05 09:49, Andrew Thompson at [EMAIL PROTECTED] wrote: > > There was a thread a month or two ago on here about voiceconduits. The > general gist was they are not yet open for public business. Are there any voice conduits customers out there? if not, maybe I ought to just walk away. --

Re: [Asterisk-Users] Advanced Conferencing options with out-of-treemodules?

Dan Austin wrote: < CLIP It turned out to be a compile issue, despite no warnings or errors. I've updated CBMysql to support app_MeetMe2, with conferences created based on the database contents, no pre-defined rooms required. Next comes adding a web frontend to the MeetMe2 gui to allow for sche

RE: [Asterisk-Users] Conecting to asterisk server through NAT usingIAX

I found out that the problems are caused by wrong ip address. My setup looks like this: asterisk is running on the server with the ip 192.168.1.252. In adddition to above address the asterisk is running also on virtual ip for HA services (192.168.1.251). When I use the virtual ip it os not working

RE: [Asterisk-Users] AMP with FC3

Yes it works great.  You need to follow the AMP doc and http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3.  Make sure you install the Fedora updates before you make any of the config changes.     good luck From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of steven cSent: Mon

RE: [Asterisk-Users] Conecting to asterisk server through NAT usingIAX

I am still having problem with connecting the iax client to my server. This time I tried firefly. The only config is the ipaddress of the server username and password. The entry in the iax.conf looks like this: [client1] type=peer usernamename=client1 secret=test context=sip host=dynamic allow=al

[Asterisk-Users] AMP with FC3

Hi All,   Does anyone know if AMP can work with Fedora Core 3?  I've tried to install AMP, and * on FC3, but found FOP failed to start when you use the command amportal start. Please advice if anyone got the solution.   Chichi   Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced s

[Asterisk-Users] about dial parameter L

quesion 1: If I write extension.conf exten => 100,1,dial(sip/100,20,L(0)) I will listen ring but I don't access when other one dial 100. who have this experience ? quesion 2: and I look at this extension.conf in voip-info.org exten => _908.,1,Dial(Modem/ttyI0:${EXTEN:1}) Could this modem

[Asterisk-Users] New SMS gateway command

I thought it might be of interest to the group to pass on information about a replacement SMS command we have developed for sending text messages from Asterisk. The FASTSMS command will route text messages to mobile phones in 154 countries. Applications include voicemail notifications, missed call

Re: [Asterisk-Users] Dial out through Broadvoice

> > So, what exactly is happening again? You can rx calls but not tx calls > > over Broadvoice? Correct? > > > > Can you rx calls over any other VoIP provider or PSTN? > > > > Could you post your current configs again? > > I was unable to tx could rx all day no problem i was getting an error:

RE: [Asterisk-Users] Advanced Conferencing options with out-of-treemodules?

>The combination of applications CBMysql and MeetMe2 seem to >address our goals. I have MeetMe2 working. CBMysql is >another story, the code looks simple enough and has been >modified to leverage MeetMe2, but * restarts everytime it >tries to launch CBMysql. I cannot find any examples of how >to

Re: [Asterisk-Users] Dial out through Broadvoice

On Monday February 28 2005 6:16 pm, Roger Hanson wrote: > - Original Message - > From: "Gabriel Gunderson" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial > Discussion" > Sent: Monday, February 28, 2005 4:49 PM > Subject: Re: [Asterisk-Users] Di

Re: [Asterisk-Users] I can't load modules (ztdummy, wcfxo.o)

Adam Goryachev wrote: On Mon, 2005-02-28 at 21:35 +0100, Androtech wrote: Hi to everybody, seems that I cannot load the zaptel modules: ztdummy says the following: [EMAIL PROTECTED] misc]# modprobe ztdummy /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.22/misc/

Re: [Asterisk-Users] I can't load modules (ztdummy, wcfxo.o)

On Mon, 2005-02-28 at 21:35 +0100, Androtech wrote: > Hi to everybody, > seems that I cannot load the zaptel modules: > > ztdummy says the following: > > [EMAIL PROTECTED] misc]# modprobe ztdummy > > /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_unregister > /lib/modules/2.4.22/misc

Re: [Asterisk-Users] Dial out through Broadvoice

- Original Message - From: "Gabriel Gunderson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, February 28, 2005 4:49 PM Subject: Re: [Asterisk-Users] Dial out through Broadvoice Am i not providing some helpfull info?

[Asterisk-Users] What about Asterisk and handling switchtype qsig (zaptel) ?

Hi all, does anybody know anything about the development of qsig? I only found something in zaptel and libpri (switchtypesettings, facility...), but nothing about Asterisk using it. Is there anything yet implemented in Asterisk, for example caller name and how does it works (automatically filling

[Asterisk-Users] Zap channel calling back after hangup (due to polarity CID detection)

Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine. I bought the card mainly to get caller ID to work properly in Sweden, and that works just fine. However, if the called or calling party hangs up after I hangup my SIP channel, polarity CID detection kicks in and dials a co

Re: [Asterisk-Users] HELP NEEDED! - Asterisk GUI

Time Bandit wrote: Its now up at http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig I would be interested in any feedback. Hope it helps. I've checked with IE and the numbers on the right point to extensions defined in the file you are editing. That's a pretty nice feature

[Asterisk-Users] DSL splitters and FXO signalling (hangup detection problem)

Hi all, My * is connected to the same line as my ADSL, of course with an attached splitter. The problem is that * cannot detect remote side hangups and continues to service the Zap FXO channel. I tried ks, ls, and gs, without success, and I think I've read most of the relevant documentation and

Re: [Asterisk-Users] Dial out through Broadvoice

> Am i not providing some helpfull info? If not tell me > what i am missing and i will get it. I am sure I have missed somethins but i > do not know what/ > I greatly apreciate all the help so far. > John Millican The service might just be down. I was up and working just fine and a few hours ag

RE: [Asterisk-Users] Asterisk Behind NAT

This has already been mentioned, but I remembered this froma little while back (sorry forget the original poster): Thanks to Pau (the original person to pose the question on this list), it's fixed. The firewall was getting in the way. I needed to open up UDP ports 1 to 2 for RTP traffi

[Asterisk-Users] Re: Strange text on Asterisk console

In article <[EMAIL PROTECTED]>, Kristian Kielhofner <[EMAIL PROTECTED]> wrote: > Howard Lowndes wrote: > > On Tue, 2005-03-01 at 08:03, Kristian Kielhofner wrote: > > > >>Tony Mountifield wrote: > >> > >>>I've just set up a new box with FC1+updates and the latest Stable > >>>Asterisk from CVS. > >

Re: [Asterisk-Users] Grandstream and VLANs

Colin Anderson wrote: Somewhere along the way, you should get enough information to make a deduction about what the GS is doing. It wouldn't suprise me if GS's VLANing is poopoo; everything about these phones seems to be "sacrifice quality at all cost". My users hate them. HTH. Colin, Like many th

Re: [Asterisk-Users] Grandstream and VLANs

I need tagging as I will have also PC's on the hub connected to the same port on the switch. PCs will be on a separate VLAN. As I have to tag one device I prefer to tag phones (apparently supported). Again, Cisco phones no problem (as expected) :) Anyone tried Grandstream's VLAN tagging? W -

Re: [Asterisk-Users] Asterisk Behind NAT

[EMAIL PROTECTED] wrote: Hi, I am working on exact same problem now and open to any suggestions. So far I : 1. Made my NAT device to forward port 5060 to Asterisk server. 2. Added line 'nat=yes' to the sip.conf for the user that is on outside. At the moment, outside phone registers with Asterisk,

Re: [Asterisk-Users] Asterisk Behind NAT

- Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, February 28, 2005 9:50 PM Subject: Re: [Asterisk-Users] Asterisk Behind NAT Hi, I am working on exact same problem now and open to any suggestions. So far I : 1.

Re: [Asterisk-Users] Strange text on Asterisk console

Howard Lowndes wrote: On Tue, 2005-03-01 at 08:03, Kristian Kielhofner wrote: Tony Mountifield wrote: I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Why are you using FC1 when FC3 is out? Better yet, why are you using FCx at all? Why not? What are you, som

Re: [Asterisk-Users] Weird behaviour on incoming DIDs

--- Michael Loftis <[EMAIL PROTECTED]> wrote: > > > --On Monday, February 28, 2005 08:46 -0800 beonice > <[EMAIL PROTECTED]> > wrote: > > > -- snipped -- > > When _I_ dial > > either DID, I get exactly the same behaviour that > I > > have specified (the call is answered, and then I > play >

Re: [Asterisk-Users] Asterisk Behind NAT

On Feb 28, 2005, at 16:48, Steve Clark wrote: What is the nat box? Linux, BSD, etc. Linux. Gibraltar firewall. http://www.gibraltar.at/ ---sambo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aste

RE: [Asterisk-Users] Grandstream and VLANs

As it should. As a stupid work-around, you could possibly put *everything else* on a seperate VLAN from the phones and you would have kind of a reverse-VLAN which would have the net same effect, this would be fine for 10 PC's but 100? 200? fuggedaboutit. -Original Message- From: Wojciech

Re: [Asterisk-Users] Asterisk Behind NAT

Hi, I am working on exact same problem now and open to any suggestions. So far I : 1. Made my NAT device to forward port 5060 to Asterisk server. 2. Added line 'nat=yes' to the sip.conf for the user that is on outside. At the moment, outside phone registers with Asterisk, but I can only place

Re: [Asterisk-Users] Re: Strange text on Asterisk console

On Mon, 28 Feb 2005, Tony Mountifield wrote: In article <[EMAIL PROTECTED]>, Carlos Chavez <[EMAIL PROTECTED]> wrote: On Mon, 28 Feb 2005 20:58:48 + (UTC), Tony Mountifield wrote I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Asterisk is started with the de

Re: [Asterisk-Users] Advanced FollowMe or Forwarding Application Suggestions

Kevin, My company, Integrics Ltd, does Asterisk consultancy. We can develop, install, and support such an application if you like. For the lists of extensions for each DID, this can be in the Asterisk extensions.conf, or an external database. The valid responses from the called party could also

RE: [Asterisk-Users] Re: Strange text on Asterisk console

The point is, only the coloured text is affected: the normal text is OK. Anyone actually seen this behaviour before? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org _

Re: [Asterisk-Users] Asterisk Behind NAT

sammy ominsky wrote: Hi all, I've done quite a bit of reading, and I see that it's going to be difficult, but as a last-ditch effort before implementing a suggestion I don't like at all, I figured I'd ask... Has anyone successfully put an asterisk box on an internal network behind a NAT device

Re: [Asterisk-Users] Grandstream and VLANs

Yes, I guess I will have to start looking at the packets. BTW: if I set the port to which grandstream is plugged to untagged vlan and leave the default VLAN 0 on the phone then everything works just fine W - Original Message - From: "Colin Anderson" <[EMAIL PROTECTED]> To: "'Asterisk

[Asterisk-Users] Re: Strange text on Asterisk console

In article <[EMAIL PROTECTED]>, Carlos Chavez <[EMAIL PROTECTED]> wrote: > On Mon, 28 Feb 2005 20:58:48 + (UTC), Tony Mountifield wrote > > I've just set up a new box with FC1+updates and the latest Stable > > Asterisk from CVS. > > > > Asterisk is started with the default safe_asterisk script

[Asterisk-Users] Re: Strange text on Asterisk console

In article <[EMAIL PROTECTED]>, Kristian Kielhofner <[EMAIL PROTECTED]> wrote: > Tony Mountifield wrote: > > I've just set up a new box with FC1+updates and the latest Stable > > Asterisk from CVS. > > Why are you using FC1 when FC3 is out? Better yet, why are you using > FCx at all? Thanks for

RE: [Asterisk-Users] Grandstream and VLANs

I'd try as "The Tyrant" always suggests and make it as simple as possible i.e. -Isolate the phone to a seperate switch that supports VLAN -Plug in a PC to the same switch -Turn OFF VLAN-ing on the switch, PC & phone -Assign a static IP to the PC & phone -Fire up Ethereal on the PC, start recording

Re: [Asterisk-Users] Strange text on Asterisk console

On Tue, 2005-03-01 at 08:03, Kristian Kielhofner wrote: > Tony Mountifield wrote: > > I've just set up a new box with FC1+updates and the latest Stable > > Asterisk from CVS. > > Why are you using FC1 when FC3 is out? Better yet, why are you using > FCx at all? Why not? What are you, som

RE: [Asterisk-Users] Asterisk Behind NAT

Title: [Asterisk-Users] Asterisk Behind NAT <>   Like you I read about and NAT and the problems.  After a few days unsuccessful battling I gave up.  Instead of using SIP directly, we've taken SIP numbers with a VoIP service provid

Re: [Asterisk-Users] Asterisk Behind NAT

On Mon, 28 Feb 2005 16:03:30 -0500, sammy ominsky wrote > Hi all, > > I've done quite a bit of reading, and I see that it's going to be > difficult, but as a last-ditch effort before implementing a > suggestion I don't like at all, I figured I'd ask... > > Has anyone successfully put an asteris

[Asterisk-Users] Advanced FollowMe or Forwarding Application Suggestions

Our company is at the point now where we're almost ready to switch over to an Asterisk server for a number of telephony applications. There is one final application I've been trying hard to find to replace something we already use with another provider. It's kind of an advanced "FollowMe" applicat

[Asterisk-Users] Newbie---Inquiring.

Hello and thanks for your advise, time, and help!!! :) :) Digium itemizes the following Linux type systems for use with Asterisk and TDM cards: Red Hat 8.0+ or Fedora Debian (2.4 or greater) Gentoo I was a user of BellLabs Unix a long time ago (~1985), but I consider myself now

Re: [Asterisk-Users] Strange text on Asterisk console

On Mon, 28 Feb 2005 20:58:48 + (UTC), Tony Mountifield wrote > I've just set up a new box with FC1+updates and the latest Stable > Asterisk from CVS. > > Asterisk is started with the default safe_asterisk script with a > console on TTY9. > > The coloured text on this console is made up of wei

[Asterisk-Users] No such host when trying to register

As I got to compile 1.0.6 and got it to run but having the same problem as before I thought of creating a new mail thread about this instead of continuing with one where topic is about something else. (Sorry) So, I can't do register anymore. It worked just a couple of days before and I haven't

RE: [Asterisk-Users] Asterisk Behind NAT

I know this is possible using IAX easily, although I guess that is not an option for you. I have no firsthand experience, but believe some have got it working via careful setup e.g noreinvites and other things. If you setup a linux router, you could maybe have a separate DMZ to the * box, but sti

Re: [Asterisk-Users] Grandstream and VLANs

Yes :) It's not DHCP as the phone won't work even with statically assigned IP. It basically looks like Grandstream is tagging and/or reading the tagged packets incorectly. W - Original Message - From: "Colin Anderson" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial

RE: [Asterisk-Users] x101p + Nortel ATA2

[EMAIL PROTECTED] wrote: > Hi, > Does anyone have any experience connecting Asterisk to a > Meridian system using an ATA2 and x101p? The basics work -- I > can make outbound calls, receive inbound, and use flash to > transfer calls, but certain things do not work, specifically > with calls from in

Re: [Asterisk-Users] Strange text on Asterisk console

Tony Mountifield wrote: I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Why are you using FC1 when FC3 is out? Better yet, why are you using FCx at all? Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on th

RE: [Asterisk-Users] Recommendation for dialplan in case of DDoS atta cks?

On Tue, 2005-03-01 at 07:11, Colin Anderson wrote: > >How about a combination of GotoIF, and app_dbodbc (or app_db): > > >exten => 700,1,playback(ddos-on) > >exten => 700,2,DBput(DDOS/yes) > > >exten => 701,1,playback(ddos-off) > >exten => 701,2,DBdel(DDOS/yes) > > >[mymainaa] > >exten => s,1,DB

Re: [Asterisk-Users] T.38 fax summary

Hi, my name is Pedro Caria I'm new to this list. I live in Portugal and find myself in the position to talk often to various parts of the world, very often the Telco line has a delay superior to 1s, I also fax in the same conditions, so to my experience faxes do work with delays far superior to

[Asterisk-Users] Asterisk Behind NAT

Hi all, I've done quite a bit of reading, and I see that it's going to be difficult, but as a last-ditch effort before implementing a suggestion I don't like at all, I figured I'd ask... Has anyone successfully put an asterisk box on an internal network behind a NAT device and been able to conn

[Asterisk-Users] Strange text on Asterisk console

I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on this console is made up of weird characters instead of normal. Please see http://www.softins.co.uk/dsc00018.j

RE: [Asterisk-Users] Grandstream and VLANs

>I can not even get IP anymore from my DHCP Hate to ask the obvious, but is the DHCP server on the same VLAN? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

RE: [Asterisk-Users] Recommendation for dialplan in case of DDoS atta cks?

>Are these inbound or outbound calls? (both?) I am pretty confused >about all of this... Sorry, I should have been more specific. The primary Asterisk box that connects with the PRI is the one I am concerned about being DoS'd - the remote IAX peer runs off of a cable modem with a dynamic IP, I h

Re: [Asterisk-users] Asterisk 1.0.6

Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been released. There is also a new tarball for Asterisk-sounds. They are available for download on the digium FTP site: ftp://ftp.asterisk.org/pub/asterisk/ ftp://ftp.asterisk.org/pub/zaptel/ ftp://ftp.asterisk.org/pub/libpri/ Chan

[Asterisk-Users] Grandstream and VLANs

I am having hard time to get VLAN tagging working with Grandstream 101 phone. As soon as I enable tagging on the switch and configure the phone to tag packets with corespodning VLAN ID #. I can not even get IP anymore from my DHCP. I have to reset the phone to factory default. I've tried differe

[Asterisk-Users] I can't load modules (ztdummy, wcfxo.o)

Hi to everybody,seems that I cannot load the zaptel modules:   ztdummy says the following:   [EMAIL PROTECTED] misc]# modprobe ztdummy /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_unregister/lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_transmit/lib/modules/2.4.22/misc

Re: [Asterisk-Users] Recommendation for dialplan in case of DDoS atta cks?

On Mon, 2005-02-28 at 14:20 -0600, Kristian Kielhofner wrote: > His suggestion was basically the same thing, only in mine you would dial > an extension to "activate" DDOS mode instead of running the database put > from the command line. > > How about monitoring your hosts with "iax2/sip sh

[Asterisk-Users] Asterisk-OH323 no ringing

Hello, I'm using Asterisk stable (1.0.3) with Asterisk-oh323 (0.6.5). Everything is working fine, well, except that : when a call is made from an h323 device (gnomemeeting for example), the caller does not hear any ringing at all, he suddenly hears the person who answers the phone. That can be q

Re: [Asterisk-Users] Recommendation for dialplan in case of DDoS atta cks?

Colin Anderson wrote: How about a combination of GotoIF, and app_dbodbc (or app_db): exten => 700,1,playback(ddos-on) exten => 700,2,DBput(DDOS/yes) exten => 701,1,playback(ddos-off) exten => 701,2,DBdel(DDOS/yes) [mymainaa] exten => s,1,DBGET(TRUE=DDOS/yes) exten => s,2,Do this exten =) s,102,

[Asterisk-Users] Cannot compile (app.c)

First of all. Asterisk was not functioning very well lately as I couldn't register. Output from *CLI: knivby*CLI> sip reload Feb 28 21:17:22 WARNING[143771648]: chan_sip.c:1310 create_addr: No such host: ipkund1.rixtelecom.se Strange, because I can ping the host. Have also tried changing the ho

Re: [Asterisk-Users] HELP NEEDED! - Asterisk GUI

> Its now up at > http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig > > I would be interested in any feedback. Hope it helps. I've checked with IE and the numbers on the right point to extensions defined in the file you are editing. That's a pretty nice feature. The problem

RE: [Asterisk-Users] Recommendation for dialplan in case of DDoS atta cks?

>How about a combination of GotoIF, and app_dbodbc (or app_db): >exten => 700,1,playback(ddos-on) >exten => 700,2,DBput(DDOS/yes) >exten => 701,1,playback(ddos-off) >exten => 701,2,DBdel(DDOS/yes) >[mymainaa] >exten => s,1,DBGET(TRUE=DDOS/yes) >exten => s,2,Do this >exten =) s,102,do something

Re: [Asterisk-Users] Recommendation for dialplan in case of DDoS atta cks?

Primary * box detects DD0S -> runs: asterisk -rx "database put PANIC DDOS YES" and have your dialplan look for that database family/key being set to determine which path it takes. When the primary * box detects that the DD0S is over -> runs: asterisk -rx "database del PANIC DDOS" On Tue, 2005

Re: [Asterisk-Users] Recommendation for dialplan in case of DDoS atta cks?

Colin Anderson wrote: I'm trying to formulate a strategy for our interconnected Asterisk IAX peers to failover to the PSTN in the event of a DDoS. We currently use them like this: DID--->PRI--->Primary Asterisk--->IAX--->On-site Asterisk--->SIP This works fine, and everyone is happy. One of my conc

[Asterisk-Users] Recommendation for dialplan in case of DDoS atta cks?

I'm trying to formulate a strategy for our interconnected Asterisk IAX peers to failover to the PSTN in the event of a DDoS. We currently use them like this: DID--->PRI--->Primary Asterisk--->IAX--->On-site Asterisk--->SIP This works fine, and everyone is happy. One of my concerns, though, is if

Re: [Asterisk-Users] FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format

Bastian Schern wrote: Yes, of course. Bastian Bastian, If you look back in the archives, you will see that many, many, many people have gotten tripped up on the "make linux26" issue. Sorry to offend you. Remember that your original post never mentioned key details that would help. Speaking o

[Asterisk-Users] Advanced Conferencing options with out-of-tree modules?

I've been poking at setting up a proof-of-concept * server as a replacement for our commercial conferencing solution. I've been through the wiki and list archives, and think I have found a combination that provides the features we want/need. The combination of applications CBMysql and MeetMe2 see

Re: [Asterisk-Users] Dial out through Broadvoice

On Monday February 28 2005 1:17 pm, Roger Hanson wrote: > see bottom > > > >> Hello all, > > > >> When I call the Broadvoice number all is good. > > > >> When I try to call out through DISA on my broadvoice line i get > > > >> the > > > > > > > > following: > > > >> Executing Dial("SIP/147.135.0.1

RE: [Asterisk-Users] phpconfig

The key to getting the menu entries to appear on the pages is the fgetc/fgets edits. It caught me out until I read through the code. BTW - how did that "error" get into CVS anyway!! Take another look at the tutorial again to see if you have missed anything else. -- No virus found in this outgoi

Re: [Asterisk-Users] List tips for new subscribers

David Brodbeck wrote: -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] On February 23, 2005 10:21 am, [EMAIL PROTECTED] wrote: Oh I'm sorry. This is the first list I've joined where this is such a big issue! Forgive me for not having your superior understanding of mai

RE: [Asterisk-Users] Manager "Message: Originate failed" beinggenerated when callee does not pick up

Thanks for your help. From what you said it looks like I should not use Originate, but there is no alternative to the "Originate" action if I just want to make an outgoing call is there? This is what my code is sending to the Manager API: clientSocket.Send(Encoding.ASCII.GetBytes("Action: Origin

Re: [Asterisk-Users] FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format

Kristian Kielhofner schrieb: Bastian Schern wrote: Hello, I've got problems to install zaptel on a SuSE 9.1 System. The System has got a Linux 2.6.9 Kernel. If I try to load zaptel framework (modprobe zaptel) I get this message: FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zapte

Re: [Asterisk-Users] Asterisk + SER

it depends what you mean by billing and accounting. postpaid? prepaid? integrated into the dialplan or just for use later? you can use cdr_mysql or similar to dump everything into a DB and build billing apps on that, if you want as well. please read the stuff here: http://www.voip-info.org/wiki-A

Re: [Asterisk-Users] Suse 9.2 + CAPI Driver

On Mon, Feb 28, 2005 at 05:06:47PM -, Victor Alvarez wrote: > > Hello, > > I'm trying to install CAPI Driver for Suse 9.2 and I found the > documentation for this pretty old since It refers to Suse 8.2 ( > http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install ). This > is

RE: [Asterisk-Users] Asterisk + SER

Hey Thanks guys... But how can I use Asterisk for billing and accounting? Do you mean use the astcc module..? Please help... Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charles Wang Sent: Saturday, February 26, 2005 11:50 PM To: [EMAIL

Re: [Asterisk-Users] Dial out through Broadvoice

see bottom - Original Message - From: "John Millican" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, February 28, 2005 10:21 AM Subject: Re: [Asterisk-Users] Dial out through Broadvoice On Saturday February 26 2005 4:45 pm, John Millican

RE: [Asterisk-Users] List tips for new subscribers

> -Original Message- > From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] > On February 23, 2005 10:21 am, [EMAIL PROTECTED] wrote: > > Oh I'm sorry. This is the first list I've joined where this > is such a big > > issue! Forgive me for not having your superior > understanding of mail >

RE: [Asterisk-Users] Asterisk network architecture

>I'd like to find a way to have my asterisk server in a DMZ protected >from outside and not directly on the internal network. Is there any >recommended architecture ? One of my current installs is a DMZ with an * server protected from outside and inside with Monowall: http://www.m0n0.ch/wall/

[Asterisk-Users] Ring state patch

Hi   Does anyone know the procedure for installing the ring state patch for snom phones . I really need this.   Id appreciate any help. Geoffrey Sachs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://list

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