[asterisk-users] Correct settings for UK (BT) FXO

then get: == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' Is there anything I can do to make this more robust? Thanks, Brian Candler. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] 9 becomes 99 ? And other strangeness

er dialling just the '9'. -- Starting simple switch on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Dial("Zap/1-1", "Zap/1|10") in new stack [Sep 14 13:24:20] WARNING[4622]: app_dial.c:1077 dial_exec_full: Unable to create channel of type 'Zap

[asterisk-users] Getting 'i' functionality on internal extensions

Hello, In the process of finding my way around, I tried to get Asterisk to give a recorded message if an invalid extension is dialled by a locally-attached phone (FXS port on TDM400P) Here's what I'm trying: -- extensions.conf -- [internal] exten => 611,1,Answer() exten

Re: [asterisk-users] 9 becomes 99 ? And other strangeness

On Thu, Sep 14, 2006 at 09:00:57AM -0500, Rich Adamson wrote: > >[outbound] > >exten => _9.,1,Dial(Zap/4/${EXTEN:1}) NOTE HERE > >exten => _9.,2,Congestion() > >exten => _9.,102,Congestion() > > > > Try replacing the first step above with: > exten => _9.,1,Dial(Zap/4/w${EXTEN:1})

Re: [asterisk-users] 9 becomes 99 ? And other strangeness

On Thu, Sep 14, 2006 at 10:37:59AM -0500, Rich Adamson wrote: > Brian Candler wrote: > >On Thu, Sep 14, 2006 at 09:00:57AM -0500, Rich Adamson wrote: > >>>[outbound] > >>>exten => _9.,1,Dial(Zap/4/${EXTEN:1}) <<<< NOTE HERE > &

Re: [asterisk-users] Getting 'i' functionality on internal extensions

On Thu, Sep 14, 2006 at 10:23:09AM -0500, Eric ManxPower Wieling wrote: > exten => _X.,1,Playback(pbx-invalid) > exten => _X.,2,Goto(s,1) OK, I can see that would match anything (and therefore implicitly anything less specific than the dialplan). However, can you explain to me why 'i' doesn't in

Re: [asterisk-users] 9 becomes 99 ? And other strangeness

On Thu, Sep 14, 2006 at 11:49:14AM -0400, Steven Totaro wrote: > Are you dialing from a sip phone? Nope; a POTS phone plugged into an FXS port on a TDM400P. > How is DTMF configured in asterisk? Whatever its default are - I've not overridden anything to do with DTMF. More precisely: I moved all

Re: [asterisk-users] 9 becomes 99 ? And other strangeness

On Thu, Sep 14, 2006 at 10:37:59AM -0500, Rich Adamson wrote: > Try the above an see what the result is. If it does not address the > problem, at least one item has been removed from the list of > possibilities. ;) OK, I can now replicate this without using outbound dialing at all, with a tiny d

Re: [asterisk-users] Getting 'i' functionality on internal extensions

On Thu, Sep 14, 2006 at 10:23:09AM -0500, Eric ManxPower Wieling wrote: > exten => _X.,1,Playback(pbx-invalid) > exten => _X.,2,Goto(s,1) The problem with this is that all extensions now take 3 seconds longer to answer. For example, with this extensions.conf: [internal] exten => 611,1,Answer() ex

Re: [asterisk-users] How to download asterisk 1.3 development version

On Fri, Sep 15, 2006 at 04:10:30PM +1000, Boneym wrote: > I would like to test asterisk 1.4 development version , can anyone >send me a link to it . Thanks in advance. Try this: (1) Open your web browser (2) Enter "www.asterisk.org" (3) Click on the link marked "downloads", at the top of

Re: [asterisk-users] 9 becomes 99 ? And other strangeness

On Thu, Sep 14, 2006 at 08:33:43PM -0500, Eric ManxPower Wieling wrote: > Turn off relaxdtmf in zapata.conf if that does not help play with the > rxgain, if that does not help, play with the txgain. If the volume is > too loud or too soft on zap channels, Asterisk can sometimes miss or see > do

Re: [asterisk-users] Cisco 7940 Problem (Mess)

On Tue, Sep 19, 2006 at 12:12:03AM +0900, Gary Guthary wrote: > This is going to be an exercise in 'Networking' for sure... > > The only catch is that per the phone's network settings: > > The phone uses a static IP of something like 192.168.0.220 with a Gateway of > 192.168.0.1. - Standard c

Re: [asterisk-users] Alcatel OXO Sip

On Tue, Sep 19, 2006 at 02:53:17PM +0200, Christian Gatti wrote: > It the question why does asterisk has problems with SIP/2.0/udp or > SIP/2.0/UDP > > if (strcasecmp(via, "SIP/2.0/UDP")) { > ast_log(LOG_WARNING, "Don't know how to respond via '%s'\n", via); > return -1; > } > > This

Re: [asterisk-users] Alcatel OXO Sip

On Tue, Sep 19, 2006 at 04:41:42AM -0700, Steve Langstaff wrote: > Oops! Yes, I was confusing strcmp and strcasecmp, sorry - too little > coffee. > > However, it's funny that the strcasecmp of "SIP/2.0/UDP" fails to match > with "SIP/2.0/udp". cat

[asterisk-users] DSL router with integrated SIP proxy?

ox? All experience and advice welcomed :-) Many thanks, Brian Candler. P.S. Sorry if this is off-topic because I didn't mention Asterisk. Let's just assume that the remote soft-switch that these telephones are going to talk to is Asterisk, OK? :-) __

[asterisk-users] ATA with wireless client

Sorry, one other equipment query: does anyone know of an ATA with wireless hardware which can act as a *client* to another wireless network? The Linksys units have an integrated wireless access point, but I want something which will work as a client onto an existing wireless network - so you can i

[asterisk-users] extensions.conf strangeness

gt; outbound ;include => invalid <<< comment out this line include => test then it works: dialling 611 from a SIP phone gives "Hello World". Could someone please explain to me why the dialplan seems to behave differently for calls originating from SIP and zaptel lines

Re: [asterisk-users] extensions.conf strangeness

On Thu, Sep 28, 2006 at 09:44:07AM -0500, Eric ManxPower Wieling wrote: > You need the [general] and [global] sections Well, if you read the attachment, you would see that I had a [general] section: [general] autofallthrough=no In what way would adding an empty [global] section alter the anomoly

[asterisk-users] Difference between "SIP Server" and "Outbound Proxy"

Proxy" or "SIP Server" setting is for, and I couldn't find that info on voip-info.org or google. If anyone could explain, clearly and precisely, how this setting is used by the client (in terms of how it affects the SIP exchanges), I'd be very grateful! Many thanks,

Re: [asterisk-users] extensions.conf strangeness

On Fri, Sep 29, 2006 at 07:49:00PM +0200, Michael Neuhauser wrote: > The order of include statements is important in 1.2, I don't know if > this still holds for trunk/1.4. Could you please try to include the > 'invalid' context as the last one (i.e., AFTER "include => test", not > before) in both i

Re: [asterisk-users] extensions.conf strangeness

On Sun, Oct 01, 2006 at 03:55:45PM +0100, Marco Mouta wrote: > >[invalid] >> exten => _X!,1,Answer() >> exten => _X!,2,Background(pbx-invalid) >Are you sure that your invalid context is correctly written? >I've never heard about this pattern match _X! >As far as i know the

Re: [asterisk-users] newbie: intelligent handling of two ISDN lines (4 B-channels)

On Tue, Oct 03, 2006 at 08:41:47PM +0200, Ekkard Gerlach wrote: > I want to bundle two ISDN-lines (4 B-channels alltogether) with > asterisk. I need it for remote administration of my customers. Sometimes > the 2 B-channels of the first ISDN lines are busy (speech or data > connection), then aster

Re: [asterisk-users] T38Passthrough and 1.4 Beta

On Thu, Oct 05, 2006 at 09:44:08AM +1000, David Hindmarsh wrote: > Does anybody have T38passthrough working using the 1.4 Beta? > > If so what did you need to do to get it working? > > I have 2 SPA-2100 that cannot get a T38 call going via 1.4. Have a look at issue 7679 on bugs.digium.com (Sipur

[asterisk-users] "set verbose 4" in SVN trunk?

In SVN trunk, I see "set verbose 4" and "set debug 4" no longer work: asterisk1*CLI> set debug 4 No such command 'set debug' (type 'help' for help) asterisk1*CLI> set verbose 4 No such command 'set verbose' (type 'help' for help) I'm probably being obtuse, but I can't find what th

Re: [asterisk-users] Bandwidth requirements

On Wed, Oct 04, 2006 at 07:51:14PM -0400, hugolivude wrote: >Age old question it seems but I haven't been able to get a handle on >it yet. Let's assume I'm using a g729 codec. If I wanted to handle >20 simultaneous calls, how much bandwidth would I need? Is there a >general formu

Re: [asterisk-users] "set verbose 4" in SVN trunk?

On Thu, Oct 05, 2006 at 03:08:16PM +0700, Pryakhin Dimitry wrote: > I think its changed to core verbose asterisk1*CLI> core debug 10 Core debug was 1 and is now 10 asterisk1*CLI> core verbose 10 Verbosity was 4 and is now 10 Thank you. Yes I was being obtuse :-) Regards, Brian.

Re: [asterisk-users] Re: extensions.conf strangeness

On Wed, Oct 04, 2006 at 12:20:40AM -0700, Martin Joseph wrote: > >Are there any debug tools which can show the "thought process" as a > >dial-plan is processed - for example, what patterns are tried and in what > >order? > > You can say show dialplan from the command line... > > Don't know if thi

Re: [asterisk-users] Bandwidth requirements

On Thu, Oct 05, 2006 at 10:53:14AM +0200, raphael Jacquot wrote: > Brian Candler wrote: > > >However on ADSL, you have to add the 15% ATM cell tax. And you would be > >wise > >to add 20% headroom (i.e. so your line is not more than 80% full) > > ATM cell tax is ac

Re: [asterisk-users] Re: extensions.conf strangeness

On Thu, Oct 05, 2006 at 04:07:14PM +0200, Michael Neuhauser wrote: > I've created and attached a one line patch (for 1.4 branch, r44464) that > should give you the info you need (sort of). But be aware that I haven't > tested it on 1.4 (only on 1.2, but things are different there). Only use > this

Re: [asterisk-users] Newbie h/w Q, and confirming basic concepts

On Thu, Oct 05, 2006 at 07:22:16PM -0700, Mike Morris wrote: >I'm preparing for my first asterisk install, and would like to ask a >hardware question & confirm my understanding of some basics: > * The Q: I'm looking for 2 FXO ports to have asterisk answer 2 >incoming lines. The

Re: [asterisk-users] Re: verbose logging to file in 1.4

On Fri, Oct 06, 2006 at 02:42:41PM +0200, Benko wrote: > 2006/10/4, Benko <[EMAIL PROTECTED]>: > >Hello! > > > >How can i change the verbose logging level to a file in 1.4? > >In 1.2 i was used to set the verbose level via asterisk -Rx 'set > >verbose 5' but in 1.4 it is always reset to OFF again,

Re: [asterisk-users] Asterisk RT on Disk On Module Performance and Durability

On Fri, Oct 06, 2006 at 05:10:34PM -0500, Erick Perez wrote: >Example: if I setup system XYZ with asterisk, then load this magical >utility/procedure that counts how many writes the filesystem has done >to / or to /,/tmp,/var and after 24 hours the utility/procedure says: >10thousan

[asterisk-users] VOIP with PSTN backup

I'm looking for a way to set up a VOIP network in branch offices where one or more phones have "lifeline" capability, i.e. can place calls if the IP network or VOIP service dies, or even if power goes down. (I'm thinking of business continuity here, not just emergency services) This seems to limit

Re: [asterisk-users] VOIP with PSTN backup

On Mon, Oct 09, 2006 at 11:01:30PM +0700, Peter Lindquist wrote: > Brian, > > Take a look at www.intertex.se I believe they have what you are looking for. Thanks - that one is on my shopping list already :-) The unit is limited to 5 users, and they are very coy about letting you know how much th

Re: [asterisk-users] isdn cross-over ...

On Mon, Oct 09, 2006 at 10:55:51AM -0500, Ejay Hire wrote: > Hi. A "cross-over" cable won't work, the isdn network provides signalling > and adressing functions. > > When I was studying for my CCIE, an ISDN simulator cost an arm-and-a-leg, > around $1k used from ebay. Couldn't you put two ISDN p

Re: [asterisk-users] Understanding NAT Traversal

On Tue, Oct 10, 2006 at 05:03:30PM -0400, hugolivude wrote: >I understand how sitting behind a NAT could cause problems for a SIP >UA. The SIP UA would create SIP mesages using IP addresses from >inside the network (i.e. 192.#.#.# or 10.#.#.#) and these IP addresses >are of course

Re: [asterisk-users] Understanding NAT Traversal

On Tue, Oct 10, 2006 at 05:03:30PM -0400, hugolivude wrote: >Similarly, why do we need a timeout on a SIP registration? Does this >work the same way as a "heartbeat" enabling disconnected UA to be >unregistered? Yes, that's the purpose: so that if you unplug a SIP phone without giving

Re: [asterisk-users] Extension and Voice Mail setup

On Wed, Oct 11, 2006 at 04:06:06AM -0400, Ahmed Ndaula wrote: > I am absolutely new to asterisk for the Voice Over IP. I have set up my > own server using asterisk, successfully connected and be in position to > test the voice over IP by connecting to the digium server and testing > the echo sys

Re: [asterisk-users] Understanding NAT Traversal

On Wed, Oct 11, 2006 at 09:21:38AM -0800, Mojo with Horan & Company, LLC wrote: > Conceivably, if only one SIP UA were in use behind a NAT router, then > when it constructed a call and needed to receive RTP streams, it would > configure port mappings in the router via the UPnP protocol, so extern

Re: [asterisk-users] Call bridged, but no sound

On Thu, Oct 12, 2006 at 02:26:16PM +0200, Norbert Zawodsky wrote: > As soon as the connection is up and the receiver is lifted on both > sides, the leds of the DSL Modem between Asterisk and my ISP, and the > leds of the switch between Asterisk and the SNOM phone start rapidly > flashing. So I assu

Re: [asterisk-users] Call bridged, but no sound

On Fri, Oct 13, 2006 at 01:35:04AM +0200, Norbert Zawodsky wrote: > I've set canreinvite=no on the channel to the SIP provider and it > immediately worked. O.k., I'm happy about that but I want to > *understand* what's going on here. > . > My setup is: > > Asterisk is connected on one side via eth

Re: [asterisk-users] Connection question...

On Thu, Oct 12, 2006 at 01:50:53AM -0300, Danko Miocevic wrote: > Dovid, the thing is that my server is on one internet connection connected > directly... without nat... but my phones are behind a nat and they are in > their own network, separated from the server. I just wanted to manage the > c

Re: [asterisk-users] VoIP+RJ11 Phone existed?

On Fri, Oct 13, 2006 at 04:35:06AM -0700, Crazy Boy wrote: >I want to buy a phone. That phone must have two ports. One is Ethernet >port (to connect to my Asterisk server) and second is RJ11 port (to >connect with my traditional PSTN exchange). I searched in internet, >but unable to

Re: [asterisk-users] GPL Softphones

On Fri, Oct 13, 2006 at 05:03:05PM +0100, Dumpolid Exeplish wrote: >How can I download the Sourcecode? To what? > went to the site Which one? > but there wasnt a >single .zip or tar file Most projects have a "Download" link you can click. For example, if you were talking about ekiga:

Re: [asterisk-users] Psst... Top secret information: Codename Pineapple

On Fri, Oct 13, 2006 at 07:00:54PM -0500, Eric ManxPower Wieling wrote: > >>>* Phones = stations, regardless of where they are > Asterisk = SIP Server, Phone = SIP Client > > >>>* Trunks = trunks to other SIP servers, bilateral > Asterisk and the other server is "peer to peer" > > >>>* Services =

Re: [asterisk-users] VoIP+RJ11 Phone existed?

On Sat, Oct 14, 2006 at 09:34:33AM +1000, Paul Hales wrote: > The Zulty's 4x5 does (or did) fwiw. Thanks. voipon.co.uk has them for GBP 299 or $557.64 (gulp) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSU

Re: [asterisk-users] Call bridged, but no sound

> I turned on sip debugging and noted folowing differences in the output > (1st='8904676', 2nd='890467610'): > > 1st: INVITE sip:s at 81.223.241.115 SIP/2.0 > 2nd: INVITE sip:01890467610 at 81.223.241.115 SIP/2.0 > > 1st: To: sip:8904676 at p1.voip.inode.at > 2nd: To: sip:890467610 at p1.voip.ino

Re: [asterisk-users] Multiple 'routes' to extension in different contextes. How to influence search oder?

On Mon, Oct 16, 2006 at 02:08:05PM +0200, Benoit Panizzon wrote: > [myself] > ; National Destinations > exten => _0z.,1,Dial(SIP/someisp/${EXTEN}); > exten => _0z.,n,Dial(Zap/g1/${EXTEN}); > > ; International Destinations > exten => _00z.,1,Dial(SIP/someisp/${EXTEN}); > exten => _00z.,n,Dial(Zap/g

Re: [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second

On Mon, Oct 16, 2006 at 04:47:31PM +0200, Giorgio Incantalupo wrote: > Hi, > every second I get on the console: > Remote UNIX connection > Remote UNIX disconnected > which gives no problem but makes console unusable. > Is there anybody who has encountered the same problem? How did you solve it? Ha

Re: [asterisk-users] Why is this happening?

On Mon, Oct 16, 2006 at 05:25:22PM -0400, Time Bandit wrote: > >Thanks for the answer, but I don't buy it. There are currently 0 > >calls up on that bridge, while another connection which has calls up > >on it is on Port 4569.. please try again. IAX2 is suppose to run on > >ONLY one port.. this i

[asterisk-users] chan_bluetooth, mobile handset as VoIP terminal?

I have been looking at chan_bluetooth, so far being unable to make it compile with Asterisk SVN trunk. I was wondering about the different ways it can be used. What I have read so far implies two possibilities: 1. Asterisk pretends to be a handsfree unit, and can use the cell phone for placing

Re: [asterisk-users] Why is this happening?

On Tue, Oct 17, 2006 at 11:20:02AM -0500, Mitch Miller wrote: > The "moving to another port" discussion is actually what happens with > sockets. A socket listens on a designated port (ex: port 80) and when a > connection is made to that socket, another socket begins to listen to > port 80 for N

Re: [asterisk-users] Orange Flash Light Mitel 5215 - Asterisk - working !

On Wed, Oct 18, 2006 at 11:43:41AM +0100, Marco Mouta wrote: > from those phones stills turning Flashing (ON/OFF) all day. It seems > to me could be bad configuration of voicemail server parameter: > Voice Mail Server: > > Could you explain me what it means # here ? I think # here is just a sh

Re: [asterisk-users] random one way audio and noise between SIP phones on same LAN

On Wed, Oct 18, 2006 at 01:44:06PM +0200, Giorgio Incantalupo wrote: > sometimes I have one way calls and noise between sip phones connected to > the same LAN so no nat/firewall is involved. I tried with different sip > phone models soft phones and the result is the same. I searched inside > eve

Re: [asterisk-users] Asterisk+SER help

On Wed, Oct 18, 2006 at 05:31:52AM -0700, Crazy Boy wrote: >I want to setup multiple SIP accounts. How can I do this? That depends what you mean by "setup multiple SIP accounts". I'm not a mind reader, but I can think of two possibilities: (1) You want to have multiple phones on the Asterisk

Re: [asterisk-users] Why is this happening?

On Wed, Oct 18, 2006 at 09:11:15AM -0400, Matt wrote: > >In the case of you example the IAX2 registration came in from the source > >port on the far device of 1207. > > > >Connections don't just move between ports. > > I understand all this. However, here is my question. > > MY on 4569 OTHER SI

Re: [asterisk-users] Linksys PAP2 dial plan help please

On Fri, Oct 20, 2006 at 03:09:56PM +0300, [EMAIL PROTECTED] wrote: >I have a Linksys PAP2-NA connectd to my asterisk. I would like the >device to add 2 characters in front of the dialled number always when >it send the call to my asterisk. I dont know how to do that. I will >summari

Re: [asterisk-users] Linksys PAP2 dial plan help please

On Fri, Oct 20, 2006 at 04:46:52PM +0300, [EMAIL PROTECTED] wrote: >Let me put is clear. I'm using TRIXBOX which many out here feel - its >for kids- but i do like it an use it for sometime. >As all i do have extensions and trunks configured on it. I want one of >my extensions to use

Re: [asterisk-users] spandsp and freebsd

On Mon, Oct 23, 2006 at 02:32:55PM +0300, Giedrius Augys wrote: > >Hi, >I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1. I get >error: configure: error: "Can't build without libtiff" . But I have >installed tiff from port tiff-3.8.2. I understand that the problem is >

Re: [Asterisk-Users] rxfax problem

On Mon, Oct 23, 2006 at 08:23:12PM -0700, Lee Howard wrote: > >If you don't mind saying, what is missing for full t.38 support? > > > Steve giving Digium a royalty-free license to his GPL software or a > pure-GPL branch of the Asterisk codebase, take your pick. Why royalty-free? AFAICS there's

Re: [asterisk-users] UA - number assignment

On Tue, Oct 24, 2006 at 10:22:33AM +0300, Paul Ianas wrote: >My problem is simple and I've issued it about 3 weeks ago. I want the >UAs to authenticate with a number to the SIP server. Is this possible? > >For example, I configured an AT-RG613TX (Allied Telesyn Residential >Gateway

Re: [asterisk-users] UA - number assignment

On Wed, Oct 25, 2006 at 11:11:00AM +0300, Paul Ianas wrote: > Indeed, this is what I want to know. When somebody wants to make a call > (using a standard telephone, connected to a media gateway), he doesn't > know what user is in my Asterisk conf. He only knows that he wants to > call John, who has

Re: [asterisk-users] Call is not coming through sipgate.co.uk+Asterisk

On Wed, Oct 25, 2006 at 01:54:43AM -0700, Crazy Boy wrote: >My sip.conf file contents: ... >[250] >type=friend >username=250 >secret=danny >callerid="Danny" >host=dynamic >context=demo >register => 100:[EMAIL PROTECTED]/100 ... >My Extensions.conf fil

Re: [asterisk-users] ASterisk Start problem

> >I have installed 1.2.12.1 in FC5 with libpri.1.2.4 > > > >when i start > > > >iam getting the following error and it quits > > > > == Registered channel type 'Local' (Local Proxy Channel Driver) > > [chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325 > >__load_resour

Re: [asterisk-users] Simple example for call transfer.

On Wed, Oct 25, 2006 at 07:14:23PM +0300, Jonson Player wrote: >i hev a subscription to a international voip provider and I want all >calls for numbers _001xx to go through my voip provider. I >tried many settings in sip.conf , extensions.conf and iax.conf. Please >give me s

Re: [asterisk-users] Add second account to Xlite 3.0

On Wed, Oct 25, 2006 at 11:37:35AM -0700, Tielin Xu wrote: > I have been testing Xlite 2.0 and 3.0. The Xlite 2.0 is slow on > initiate time, but I can add second sip proxy account, which is very > critical to my testing. I installed Xlite 3.0, which I could not add > second account on SIP account