Re: [Asterisk-Users] Dial via sip gateway?

Bob, I have a question into mediatrix for this, but maybe you have figured it out. I am trying to map a SIP user to a specific PSTN line. I have my extensions.conf file as you show below, but on the 1204, it just grabs whatever line is available, whereas I want extension 101 to always be port1 on 1

[Asterisk-Users] OT:Linux(or *BSD) SNMP tools (Was: Re: rtp sound quality?)

On Saturday 31 January 2004 21:31, you wrote: <<>> > I am just a low level c hack. Before I go out and write any thing to do > this snmp admin stuff, > are there any linux tools I could use to do this? Net-SNMP (http://freshmeat.net/projects/net-snmp/ , formerly UCB-SNMP or something) is very h

Re: [Asterisk-Users] rtp sound quality?

Rich Adamson wrote: Thanks Bob, that fixed it. Any other hints/issues/default values that I should muck with, or is that about it? Seems like it works pretty good; excellent echo cancellation, etc. I haven't done anything with the box as yet for dialing outbound. Anything to be concerned with, sp

RE: [Asterisk-Users] determining legal VoIP service

In order to sell it to ITXC, the minimum capacity is 1E1/T1. We are buying/selling termination without capacity limit. If you have some good routes, please let me know Jeff Chen U&M Network, Canada Tel:1-416-324-8066 Fax: 1-416-324-8261 www.mutualphone.com Yahoo messenger ID: jeffcheny2k -

Re: [Asterisk-Users] Dial via sip gateway?

Rich Adamson wrote: I'm having a brain fart What's the proper syntax for dialing out via a sip g/w (Mediatrix)? Been trying stuff similar to: exten => _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1}) where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did even try the IP. Ric

Re: [Asterisk-Users] How to delay dialing

On Saturday 31 January 2004 11:23, John Baker wrote: > On Sat, 2004-01-31 at 10:21, Tilghman Lesher wrote: > > I'll look at it again, but since I don't have a VoiceTronix card > > installed in any of my machines yet, I can't test it directly. > > I've got one I might be able to loan you for a coupl

Re: [Asterisk-Users] Dial via sip gateway?

On Sat, 31 Jan 2004, Rich Adamson wrote: > I'm having a brain fart > > What's the proper syntax for dialing out via a sip g/w (Mediatrix)? > > Been trying stuff similar to: > exten => _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1}) > where 3091 is alias for the port on the Mediatrix. Sniffer in

[Asterisk-Users] PCI expansion slots.

Hello, Did anyone use PCI expansion slots such as: http://www.cyberresearch.com/store/product/311.2.htm I want to know how well does it work with Asterisk FXO/FXS cards? Also, does FXO/FXS drivers work automatically (meaning seemlessly recognize the expansion slots) without any Power/Bandwidth

[Asterisk-Users] Dial via sip gateway?

I'm having a brain fart What's the proper syntax for dialing out via a sip g/w (Mediatrix)? Been trying stuff similar to: exten => _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1}) where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did even try the IP. Rich ___

RE: [Asterisk-Users] Question on setting up asterisk with hunting lines

The answer is B. Hunting lines is nothing else but a call forward on busy. So line 1 forwards to 2 line 2 forwards to line 3 and so forth. In your dialplan just set up the 5 lines with the same incoming context. Regards. Alfred. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAI

Re: [Asterisk-Users] Re: retrans_pkt: Maximum retries exceeded on call

2004-01-26 14:12 markster * channels/chan_sip.c (1.284): Don't send VMWI when we're not registered Yes that was fixed on the 26th. On Sat, 31 Jan 2004, Clif Jones wrote: > I noticed this too and it is a pain to look at. I saw it because some > of my SIP phones were turned of

Re: [Asterisk-Users] Re: retrans_pkt: Maximum retries exceeded on call

That sending notify to endpoints that aren't registered has been fixed recently but we have one more bug that causes 0.0.0.0 bkw On Sat, 31 Jan 2004, Clif Jones wrote: > I noticed this too and it is a pain to look at. I saw it because some > of my SIP phones were turned off and > the NOTIFY's f

Re: [Asterisk-Users] rtp sound quality?

> > Anyone have any thoughts as to why ringback and MOH are choppy but > > conversations are fine? Anything else I can look at to isolate the > > issue? > > First guess (rather likely): Silence supression > > Second guess (unlikely): Non optimal "Voice frames per TX" as it is > called in Grand

Re: [Asterisk-Users] Re: retrans_pkt: Maximum retries exceeded on call

I noticed this too and it is a pain to look at. I saw it because some of my SIP phones were turned off and the NOTIFY's for "no voicemail" reached maximum re-transmissions. Duh! Nobody was there to answer it. I didn't check to see what the log level was but if it only shows up on -vvv console o

RE: [Asterisk-Users] 8 lines - best approach

Thanks for the overwhelming response guys. Just wait for some time (6 weeks) and some of you will get to test it for sure. Watch out this mailing list for the announcement. For now, let's keep it a little secret. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Beha

Re: [Asterisk-Users] Compiling while * is running

On Sat, Jan 31, 2004 at 07:43:46PM -0600, Brian West wrote: > Nope I do make install all the time with asterisk running without ONE > problem. As I said, this behaviour is specific to some implementations of dynamic loadable modules. It depends what OS (and in some cases what version of the OS) yo

Re: [Asterisk-Users] rtp sound quality?

Hi Rich! > Anyone have any thoughts as to why ringback and MOH are choppy but > conversations are fine? Anything else I can look at to isolate the > issue? First guess (rather likely): Silence supression Second guess (unlikely): Non optimal "Voice frames per TX" as it is called in Grandstream

Re: [Asterisk-Users] Compiling while * is running

Nope I do make install all the time with asterisk running without ONE problem. bkw On Sat, 31 Jan 2004, William Waites wrote: > While your problem is most likely bad RAM as other > replies have suggested, there is another thing to > keep in mind. > > Some implementations of dynamic module loadin

Re: [Asterisk-Users] 8 lines - best approach

Rob Fugina writes: > On Sat, Jan 31, 2004 at 12:00:49PM -0800, Asterisk wrote: > > How about a 16 port FXO/FXS card (your choice of FXO/FXS modules or any > > mix of them) for $999. Will that be a good option? > > I know (me) someone (me) who'd make a (me) really good (me) beta tester (me)... I

RE: [Asterisk-Users] 8 lines - best approach

well quit with the suspense already and tell us who! :-) -Original Message- From: Rob Fugina [mailto:[EMAIL PROTECTED] Sent: Saturday, January 31, 2004 7:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 8 lines - best approach On Sat, Jan 31, 2004 at 12:00:49PM -0800, Asterisk

Re: [Asterisk-Users] 8 lines - best approach

On Sat, Jan 31, 2004 at 12:00:49PM -0800, Asterisk wrote: > How about a 16 port FXO/FXS card (your choice of FXO/FXS modules or any > mix of them) for $999. Will that be a good option? I know (me) someone (me) who'd make a (me) really good (me) beta tester (me)... -- Rob Fugina, Systems Guy [EMA

Re: [Asterisk-Users] rtp sound quality?

> >pstn -> sip gw -> * -> C7960 > > > >When I dial into * via the pstn, I hear the ivr menu just fine (good > >quality). I press 3000 (valid extn), and I begin to hear ringing however the > >ring back is very very choppy. > > > >I answer the C7960, and speech is clear in both directions. Place the

Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?

Not sure what your question is, but the Polycom IP Phones use SUBSCRIBE: "SoundPoint® IP supports shared call appearances (SCA) using the SUBSCRIBE-NOTIFY method in the "SIP Specific Event Notification" framework (RFC 3265)." This from the admin guide at http://www.polycom.com/common/pw_item_show

Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?

> So, what hardware or use is the SUBSCRIBE method used for in > chan_sip.c? I asked this question a while ago, and got resounding > silence. Maybe someone who is better at de-tangling C code than I am > could take a peek. John, Not sure, but seems to me it came in about the time Olle and Sn

[Asterisk-Users] SUBSCRIBE in chan_sip - anyone?

So, what hardware or use is the SUBSCRIBE method used for in chan_sip.c? I asked this question a while ago, and got resounding silence. Maybe someone who is better at de-tangling C code than I am could take a peek. JT ___ Asterisk-Users mailing list

Re: [Asterisk-Users] rtp sound quality?

Rich Adamson wrote: pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the

Re: [Asterisk-Users] TE410P E1 PRI problem

On Sat, 31 Jan 2004, Tomica Crnek waxed: > Hi everyone! > > Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk > is connected to our national telecom company here in Croatia. When I call from > outside over this trunk to my company I get 'error in connection' r

RE: [Asterisk-Users] rtp sound quality?

> >pstn -> sip gw -> * -> C7960 > > > >When I dial into * via the pstn, I hear the ivr menu just fine (good > >quality). I press 3000 (valid extn), and I begin to hear ringing > however >the ring back is very very choppy. > > Where are you getting timing from? Zaptel device? Ztdummy? The * syste

RE: [Asterisk-Users] rtp sound quality?

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson >pstn -> sip gw -> * -> C7960 > >When I dial into * via the pstn, I hear the ivr menu just fine (good >quality). I press 3000 (valid extn), and I begin to hear ringing however >the ring back is

Re: [Asterisk-Users] Multiple Line Appearances

My point in my last paragraph was that you don't want to do this on the user-agent side; you want to control this on the server side. To use asterisk parlance: exten => 1234,1,Dial(SIP/jane&SIP/bill) This means that when extension 1234 is called, that the (single) phones named "jane" and "bill

[Asterisk-Users] rtp sound quality?

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold,

RE: [Asterisk-Users] Internal Lines Dialing Out

Thanks to both who replied, it works!!! I cannot believe i missed that, talk about being knocked down a couple notches:) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Rodgers Sent: Saturday, January 31, 2004 3:19 PM To: [EMAIL PROTECTED] Subject: Re:

Re: [Asterisk-Users] 8 lines - best approach

What card would that be? I would be interested to test it out. David - Original Message - From: "Asterisk" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, January 31, 2004 12:00 PM Subject: RE: [Asterisk-Users] 8 lines - best approach > How about a 16 port FXO/FXS card (y

Re: [Asterisk-Users] Caller ID Presentment on PRI...

Hiya, > Is anybody out there currently able to set CIDName to be something > different than the reverse lookup name? My goal is not to spoof the > White House, btw, but it makes a fun example. Beware the three-letter agencies. Beware even more the two-letter ones. yeah, impersonating the White H

[Asterisk-Users] echo cancellation disabled

Hello   I get these entries in my event log   Jan 31 19:21:08 gateway kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Do I have to change anything for enable echo cancellation?   Regards Deepak    

Re: [Asterisk-Users] Internal Lines Dialing Out

As the prevous poster pointed out, replace all instances of $EXTEN with ${EXTEN} and it should start working for you. Steve. On Saturday 31 January 2004 12:32, Bruce Marler wrote: > Thanks for the tips, i tried it though and i still get the same thing. > > basically what happens is I pick up

RE: [Asterisk-Users] Internal Lines Dialing Out

It's ${EXTEN} not $EXTEN On Sat, 2004-01-31 at 14:32, Bruce Marler wrote: > Thanks for the tips, i tried it though and i still get the same thing. > > basically what happens is I pick up the phone, hear dialtone, dial the > number, get a slight pause, here dial tone again (when i would expect it

[Asterisk-Users] The future of VoIP regulation (in the US)

Listers, Some of you may find this link of interest It was pointed out to me by one of our clients. Howard White ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/li

RE: [Asterisk-Users] Internal Lines Dialing Out

>> > exten => _.,1,Dial(Zap/1/$EXTEN) exten => _.,1,Dial(Zap/1/${EXTEN}) Gotta put the name of the variable in brackets for it to work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNS

[Asterisk-Users] Dial app does not indicate ringing to calling party

I hope somebody has seen this before... I'm trying to use a Dial command on a inbound call to ring multiple destinations.The calls come in to me from the provider on IAX2, and one of the destinations I try to ring is a IAX2 to call to my cell phone. When I add the IAX2 destination into the Dia

RE: [Asterisk-Users] Internal Lines Dialing Out

Thanks for the tips, i tried it though and i still get the same thing. basically what happens is I pick up the phone, hear dialtone, dial the number, get a slight pause, here dial tone again (when i would expect it to be dialing), and then I dial the # again and it works, it seems that it is passi

RE: [Asterisk-Users] 8 lines - best approach

How about a 16 port FXO/FXS card (your choice of FXO/FXS modules or any mix of them) for $999. Will that be a good option? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Saturday, January 24, 2004 5:35 PM To: [EMAIL PROTECTED] Subje

Re: [Asterisk-Users] Compiling while * is running

While your problem is most likely bad RAM as other replies have suggested, there is another thing to keep in mind. Some implementations of dynamic module loading have problems if a loaded module is overwritten on the disk. What this means is that it is safest to stop Asterisk just before running

Re: [Asterisk-Users] Compiling while * is running

On Friday 30 January 2004 17:57, Greg Boehnlein wrote: > Speaking of Binary packages, has anyone had the chance to test the > Asterisk 0.7.1 RPMS that I built last weekend? I'm using them on an up-to-date Fedora Core 1. So far so good. I don't have my Digium hardware yet, so I'm still just usin

Re: [Asterisk-Users] SIP gateway question

Hi Bob, > >The 1204 then sends "one" more packet to * with both the source and destination > >ports one digit greater then what was used for the rtp session. I'm assuming > >that's a bug in their code; anyone seen something like that before? > > > That would be RTCP (RTP + 1) > > >3. Has anyone p

Re: [Asterisk-Users] SIP gateway question

Rich Adamson wrote: The 1204 then sends "one" more packet to * with both the source and destination ports one digit greater then what was used for the rtp session. I'm assuming that's a bug in their code; anyone seen something like that before? That would be RTCP (RTP + 1) 3. Has anyone played wi

Re: [Asterisk-Users] Words for Allison(?)

On Sat, 2004-01-31 at 18:24, Rob Fugina wrote: > In the mean time, I've seen references to bug #'s, here on the list and > in the CVS logs. I've yet to stumble across the bug tracking system, > though -- can you give me a nudge in the right direction? http://bugs.digium.com/ F _

Re: [Asterisk-Users] Words for Allison(?)

Rob Fugina said: > On Fri, Jan 30, 2004 at 10:48:35PM -0500, John Todd wrote: > > > In the mean time, I've seen references to bug #'s, here on the list and > in the CVS logs. I've yet to stumble across the bug tracking system, > though -- can you give me a nudge in the right direction? > > Thanx,

Re: [Asterisk-Users] asterisk with big number of extentions.

On Sat, 2004-01-31 at 10:36, WipeOut wrote: > Fran Boon wrote: > > OK, so what success have people had with which clustering technologies? > > I'm more interested in resilience than performance. > I would think that failover clustering would be far easier than a load > sharing or processing cluste

Re: [Asterisk-Users] Words for Allison(?)

bugs.digium.com On Sat, 2004-01-31 at 12:24, Rob Fugina wrote: > On Fri, Jan 30, 2004 at 10:48:35PM -0500, John Todd wrote: > > You may consider putting together concrete lists of words so that I > > or others may keep them on short lists so that when we have Allison > > do various recordings we

Re: [Asterisk-Users] Words for Allison(?)

On Fri, Jan 30, 2004 at 10:48:35PM -0500, John Todd wrote: > You may consider putting together concrete lists of words so that I > or others may keep them on short lists so that when we have Allison > do various recordings we can find them in a single place. In fact, a > bugnote would be the op

Re: [Asterisk-Users] SNOM 200 question

Are you sure that the snom isn't negotiating the GSM codec? I think that this is negotiated by default unless you have disallow/allow statement. To determine whether this is the problem, put the following into the [general] section of you sip.conf: disallow=all allow=ulaw allow=alaw As for the c

Re: [Asterisk-Users] Re: Grandstream Firmware ?

Yes. We're currently testing 1.04.45 before making it available on our web site (www.telappliant.com/grandstream). Tan telappliant.com - Original Message - From: "Mike Machado" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, January 31, 2004 12:08 AM Subject: Re: [Asterisk-

Re: [Asterisk-Users] Multiple Line Appearances

How were you able to integrate this with asterisk? Or did you drop asterisk in favor of ser? John On Thu, 2004-01-29 at 12:44, John Todd wrote: > At 12:20 PM -0500 1/29/04, Jeremy McNamara wrote: > >MLS Drop for SysAdmin wrote: > > > >>Has anyone successfully implemented concurrent appearance of

[Asterisk-Users] Are there any list moderators?

I'm just curious... I have several times posted a message accidentally using the wrong account - since the address I use for this list isn't my default one. I often re-post using the correct account, and get a notification on the first that my message is 'pending approval'. I don't expect my wr

Re: [Asterisk-Users] How to delay dialing

Oops... sorry about that, stupid webmail system defaults to HTML (don't use it very often so frequently overlook this. I usually send mail using 'apple mail', which seems to screw up even plain text e-mails, but... well... have to retaliate against Outlook Express users some way!! Anyways - doe

Re: [Asterisk-Users] asterisk php status viewer

Looks interesting I will check it out and see what I can do with it =) On Sat, 2004-01-31 at 08:17, Brancaleoni Matteo wrote: > since I was annoyed this morning, I > wrote this simple php script to output > channel status from asterisk manager. > > > that's very bad written, nor commented... > I

Re: [Asterisk-Users] smtp question

Yep thats what I was thinking but apparently that isn't what is going on. We have have 41,000 bounces that have been blocked from said mail server. The from is the mail server and no return path is set. EVIL bkw On Sat, 31 Jan 2004, Walt Reed wrote: > The headers From:, Reply-To: etc generall

Re: [Asterisk-Users] How to delay dialing

I've got one I might be able to loan you for a couple of months, if you have a great desire to hack away. John On Sat, 2004-01-31 at 10:21, Tilghman Lesher wrote: > First, turn off your HTML in all posts to the list. > > On Saturday 31 January 2004 02:03, Terence Parker wrote: > > Thanks again f

[Asterisk-Users] TE410P E1 PRI problem

  Hi everyone!   Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anythi

RE: [Asterisk-Users] Music on Hold Warnings

Tilghman Thanks for the help. You were spot on, yup the bitrate was screwed. "NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?!" And the machine does seem to be heavily underload - Asterisk = 100% CPU. MOH is working great now. Thanks. -Original

Re: [Asterisk-Users] How to delay dialing

First, turn off your HTML in all posts to the list. On Saturday 31 January 2004 02:03, Terence Parker wrote: > Thanks again for the patch - and the updated patch too! > > Actually, I was looking more closely at the chan_vpb.c file > earlier after the first patch and tried recompiling * again > af

Re: [Asterisk-Users] Caller ID Presentment on PRI...

On Sat, 31 Jan 2004, David C. Troy wrote: > What I am wondering is whether a PRI has to have some feature turned on > in order to inject the textual Caller ID name data, or whether this is > just how PRI's work in general. And if so, who controls the reverse > mapping, and how does that work with

Re: [Asterisk-Users] Caller ID Presentment on PRI...

> What I am wondering is whether a PRI has to have some feature turned on > in order to inject the textual Caller ID name data, or whether this is > just how PRI's work in general. And if so, who controls the reverse > mapping, and how does that work with CLEC phone numbers, etc? Nope -- that's h

[Asterisk-Users] Caller ID Presentment on PRI...

Hey folks, I have a T100P card with a PRI; when doing outbound dialing over the PRI, I can use SetCIDNum(2024561414|a) to force caller ID to display as "The White House" on a land line. This is apparently done as a reverse lookup by Verizon, as I do not hand the PRI the words "The White House" -

[Asterisk-Users] Using an additional modem to get CallerID information

There are two stages in the process of getting callerID information from a standard modem, to be used in Asterisk. The first stage is actually capturing the information from the modem, the second stage is importing the captured data into Asterisk. Capturing the caller ID details from the modem I

RE: [Asterisk-Users] determining legal VoIP service

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Walker Haddock > Sent: Friday, January 30, 2004 5:52 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] determining legal VoIP service > > > Can anyone recommend who we can consult with that c

Re: [Asterisk-Users] Internal Lines Dialing Out

On Fri, 30 Jan 2004, Steve Rodgers wrote: > Oops! I forgot the leading underscore. Use this version below. > > Steve. > exten =>_ NXX,1,Dial(Zap/1/$EXTEN) > exten => _1NXXNXX,1,Dial(Zap/1/$EXTEN) And reaching us wot is in the rest of the world...? ;-) Steve __

Re: [Asterisk-Users] determining legal VoIP service

On Fri, 30 Jan 2004, Dustin Goodwin wrote: > Actually I believe this is one of the few things that can be done > without worrying about the state(s) PUC coming down on your head. Since > your users are in another country the state PUC cannot consider you > providing a telephone service in the

[Asterisk-Users] SIP gateway question

Just received a Mediatrix 1204 fxo sip gateway and playing with the initial config's, etc. It's working, but have a ways to go before it could be considered usable. The box was not designed to "register" like sip phones do. The incoming pstn line is an ordinary 2-wire analog US pots line, and I'm

Re: [Asterisk-Users] smtp question

The headers From:, Reply-To: etc generally ARE things like MAILER-DAEMON. The envelope sender used in the SMTP conversation and Return-Path: should be <>. On Sat, Jan 31, 2004 at 01:20:02AM -0600, Brian West said: > Correct me if I'm wrong here but when a message bounces and the mailer/mta > gene

[Asterisk-Users] asterisk php status viewer

since I was annoyed this morning, I wrote this simple php script to output channel status from asterisk manager. that's very bad written, nor commented... I wrote that just for fun and if someone will use that / improve it , just lemme know. http://asterisk.espia-net.net (wrote with php 4.3.3

Re: [Asterisk-Users] newbie thinclient env

On Saturday 31 January 2004 12:13, you wrote: > Hi! > > > - asterix on the LTSP servers ? > > It is not advisable to run Asterisk on a server that runs X-Windows. You > could give it a try on one machine, but I might very well turn out that > you'll not like the resulting voice quality ("choppy sou

Re: [Asterisk-Users] determining legal VoIP service

I heard this argument when in SA by a group using Net2Phone- it was not accepted by the head of the Foreign Department of Telkom with whom I also spoke. Also of importance is that in many countries there have been police raids and confiscation of equipment used by CallShops because the CallShops ar

Re: [Asterisk-Users] availability of the "ExtensionState" was "Asterisk Manager Interface notes"

mattf wrote: Hello, I was referring to the availability of the "ExtensionState" Action (see the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20manager%20API ), even though I don't actually use it. Ok. Has anyone used the"ExtensionState" Action sucessfully? I can't get it to wor

Re: [Asterisk-Users] P2P RTP without SIP re-invites

hi > > I guess this would work if both Alice and Bob were NAT'ed on the inside of the same > NAT box. The problem is that if Alice and Bob both have NAT=yes and CANREINVITE=yes > and they're on separate NAT'ed networks, the call is broken. So it's a dangerous > configuration. nope. I have a publi

Re: [Asterisk-Users] P2P RTP without SIP re-invites

Brancaleoni Matteo wrote: Hi. If both Alice and Bob are connected without NAT, have the same codec support and have canreinvite=yes * Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it goes directly from Alice to Bob Not all UAs support a re-INVITE and in publi

Re: [Asterisk-Users] P2P RTP without SIP re-invites

Hi. > If both Alice and Bob are connected without NAT, have the same codec support and > have canreinvite=yes > * Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it >goes directly from Alice to Bob > Not all UAs support a re-INVITE and in public scenarios, a lo

Re: [Asterisk-Users] newbie thinclient env

Hi! > - asterix on the LTSP servers ? It is not advisable to run Asterisk on a server that runs X-Windows. You could give it a try on one machine, but I might very well turn out that you'll not like the resulting voice quality ("choppy sound"). > - can i connect a phone to the thinclients via

RE: [Asterisk-Users] determining legal VoIP service

Hi, > -Original Message- > Actually I believe this is one of the few things that can be > done without worrying about the state(s) PUC coming down on > your head. Since your users are in another country the state > PUC cannot consider you providing a telephone service in > their juris

Re: [Asterisk-Users] asterisk with big number of extentions.

Fran Boon wrote: Anton wrote: you can do it with a well setup cluster OK, so what success have people had with which clustering technologies? I'm more interested in resilience than performance. I would think that failover clustering would be far easier than a load sharing or processing clust

Re: [Asterisk-Users] Firefly and asterisk*

Hi Adam, I just got it to work ;)) I added an entry at the bottom of my iax.conf : register => *MY_FIREFLY_NUMBER*:[EMAIL PROTECTED] [firefly] type=friend host=firefly.virbiage.com context=incoming then, when a firefly user calls me, he is taken to incoming/s. I'm not sure if type=friend is rig

[Asterisk-Users] newbie thinclient env

i have a lot of questions, maybe you've got some answers. I've got the following setup : 1 central server ( linux ) , 17 servers ( LTSP ) on different locations, where each server has a number of thinclients connected to it ( max 6 ) The central server / 17 LTSP servers are on the internet and s

Re: [Asterisk-Users] P2P RTP without SIP re-invites

Let's go through how SIP works in Asterisk compared with a SIP Proxy. Remember that Asterisk is not designed to be a SIP Proxy, it's designed to be a Multi-VOIP and PSTN PBX, a quite complicated task. (I'm not going into all details (ACK, TRYING, RINGING etc)) We have two SIP users, Alice and Bob.

Re: [Asterisk-Users] How to delay dialing

Tilghman, Thanks again for the patch - and the updated patch too! Actually, I was looking more closely at the chan_vpb.c file earlier after the first patch and tried recompiling * again after removing 'char' from char t* - since I gather that wasn't required again. Asterisk now compiles p

[Asterisk-Users] smtp question

Correct me if I'm wrong here but when a message bounces and the mailer/mta generates a bounce message shouldn't the from field have <> in it instead of an email addres (ie. [EMAIL PROTECTED]). The list was nailed with over 13,000 bounce messages(and they keep coming) from ONE list subscriber and I

Re: [Asterisk-Users] Cameron Palmer / voiceglo

At 23:40 30/01/04 -0700, you wrote: I found a message in the archives from Cameron Palmer on 23 Dec regarding his voiceglo SIP configuration. Unfortunately (for me), the archive has his email address removed. So, Cameron -- or anybody else using voiceglo with their * box -- please reply to me so th