Re: [asterisk-users] Fast detection of unreachable SIP clients?
On Monday 06 November 2006 16:41, Matt wrote:
> This should work.. please make sure you have qualify=yes on in
> your sip.conf file for each of your sip entries.
Now it works. Thank you!
>
> On 11/6/06, Dmitry Ivanov <[EMAIL PROTECTED]> wrote:
> > Hello!
> >
> > I have this in my dialplan:
> >
> > Dial(SIP/${ext}, 300);
> > switch(${DIALSTATUS}) {
> > case BUSY:
> > Busy();
> > break;
> > default:
> > Hangup();
> > };
> >
> > This means that SIP phone will ring for max. five minutes if
> > phone can be contacted. When SIP phone is turned off or
> > there is no connectivity, calling party hears many many
> > seconds of silence. I want Dial() to return CHANUNAVAIL if
> > there was no SIP response from the phone within 1 or 2
> > seconds. In this case, calling party will hear "out of
> > range" message similar to mobile networks. Is this possible?
--
Dmitry Ivanov
Network engineer
Telecentrs Riga, Latvia
[EMAIL PROTECTED]
(+371) 7160235
Weather at Riga Intl (EVRA/RIX): Tuesday 07 November 2006 09:50,9
km/h S,2°C,1003 hPa,,,
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[asterisk-users] Fast detection of unreachable SIP clients?
Hello!
I have this in my dialplan:
Dial(SIP/${ext}, 300);
switch(${DIALSTATUS}) {
case BUSY:
Busy();
break;
default:
Hangup();
};
This means that SIP phone will ring for max. five minutes if
phone can be contacted. When SIP phone is turned off or there is
no connectivity, calling party hears many many seconds of
silence. I want Dial() to return CHANUNAVAIL if there was no SIP
response from the phone within 1 or 2 seconds. In this case,
calling party will hear "out of range" message similar to mobile
networks. Is this possible?
--
Dmitry Ivanov
Network engineer
Telecentrs Riga, Latvia
[EMAIL PROTECTED]
(+371) 7160235
Weather at Riga Intl (EVRA/RIX): Monday 06 November 2006 10:50,29
km/h SSE,-4°C,1004 hPa,Broken clouds at 396 meters;Overcast
clouds at 640 meters,, Snow; Snow;Heavy Snow
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Re: [Asterisk-Users] Huawei EP201S
On Wednesday 03 May 2006 11:08, Tomislav Parčina wrote: > Has anybody used Huawei EP201S IP phone? I need 9 SIP phones that > cost around 100USD, and those phones are one of options. > > Can anybody suggest anything else that costs around 100USD? We have >10 Grandstream Budgetones 101 & 102 here in office. They work most of the time but overall quality is poor. The worst thing is that BT-10x are not well suited for mass deployment. Apparently, acceptable IP phones under 100 USD do not exist. ___ --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
Re: [Asterisk-Users] mISDN: No DID/extension information returns busy to caller
On Friday 28 April 2006 11:35, Ralf Schlatterbeck wrote: > I don't see the call at all in asterisk. Maybe your telco does not route these calls with incomplete number to you? ___ --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
Re: [Asterisk-Users] test numbers in different countries!
On Wednesday 26 April 2006 11:48, Dmitry Ivanov wrote:
> On Wednesday 26 April 2006 07:52, Jason Frisch wrote:
> > How about using time announments? I list of these
> > for each country would be great!
>
> I have some test numbers on my switch in Latvia:
>
> +371 7160201 -- echo
> +371 7160202 -- music :)
> +371 7160203 -- time
+371 7160204 -- SayDigits(${CALLERIDNUM});
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Re: [Asterisk-Users] test numbers in different countries!
On Wednesday 26 April 2006 07:52, Jason Frisch wrote: > How about using time announments? I list of these > for each country would be great! I have some test numbers on my switch in Latvia: +371 7160201 -- echo +371 7160202 -- music :) +371 7160203 -- time Do you mean something like this? ___ --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
Re: [Asterisk-Users] Two asterisk process in one hardware.
On Tuesday 25 April 2006 00:57, Juan Salas wrote: > Hello. > > Has anybody knows how run two asterisk process > in one hardware? (each one with its own configuration?) It is possible. 1) Use different UDP ports for SIP/IAX/RTP 2) Use different log files and astdb files But most users do not need this. If you need separate phone systems for multiple customers, use separate contexts instead. ___ --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
Re: [Asterisk-Users] Playback(something,noanswer) on Zap?
On Thursday 20 April 2006 19:22, Kevin P. Fleming wrote:
> A better solution is to set the PRI hangup cause before dropping the
> incoming call; if you set the hangup cause to 'number not assigned'
> then your telco's switch will play its normal intercept message to
> the caller.
Thank you! This works!
context from-e1 {
_X. => {
AGI(pub2ext.agi);
PRI_CAUSE=1;
Hangup();
};
};
Now caller hears voice from his/her telco (not from my telco) saying
that number is not available. This is even better.
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Re: [Asterisk-Users] Using ISDN MSNs for dialing out of Asterisk
On Tuesday 18 April 2006 14:58, Christian Gröger wrote:
> Hi,
> I am using Asterisk with misdn connected to an ISDN Line, so I have
> several numbers I can use...
>
> I know that I can use misdn like this in my extensions.conf:
>
> exten => _0.,1,Dial(mISDN/1/${EXTEN:1})
>
> But how can I use another number/MSN of my ISDN connection... it
> always uses the default number, but i'd like to use another MSN for
> calling... Can somebody help me please?
Not sure about mISDN... But we have E1 PRI and 100 numbers. We can send
any of these 100 numbers as caller id and it will reach remote end. If
we send anything else then our telco sets caller id to our default
number (first of 100). I use AGI to set caller id for outgoing calls.
If you want to use other available numbers as caller id, try to set
caller id in your extensions.conf.
Set(CALLERID(number)=1234567)
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[Asterisk-Users] Playback(something,noanswer) on Zap?
Hi! Our telco routes multiple numbers through PRI to our Asterisk. Not all of these numbers are in use. I have noticed recently that someone keeps calling unused phone number from outside world. I called them and asked why do they call dead number. The person on the far end explained that she keeps calling this number because she hears "busy" tone every time... Most telcos these days provide verbal in-band notification in case if number does not exist. Those nice female voices. People do expect this behaviour from their phones. People no longer accept beeps as "number does not exist" signal. First thing I've tried was Playback(invalid). The problem was that asterisk answered incoming call. This should not happen when caller does not reach his/her destination. Next, I tried Playback(invalid,noanswer). This time, Asterisk did not answer the call. But there was no sound! Apparently, Playback(invalid,noanswer) does not work with Zap/PRI. Is this bug? ___ --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
Re: [Asterisk-Users] Grandstream Budgetone and Mac mini?
On Tuesday 18 April 2006 18:26, The VoIP Connection wrote: > The switch in the Budgetone is 10Base-T. If the PC NIC cannot > auto-detect or otherwise handle that, it will be a problem. Yes! Looks like Mac mini cannot handle 10HD :) This is what I see when BT-102 is connected to Alcatel Omnistack switch. Gigabit port is in 10Mbps half duplex mode: omni2# show interfaces status Flow Link Back Mdix Port Type Duplex Speed Neg ctrl State Pressure Mode -- - --- --- e1 100M-Copper-- -- -- -- Down -- -- e2 100M-Copper-- -- -- -- Down -- -- .. g2 1G-Combo-C Half10Enabled Off Up Disabled Off .. ___ --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] Grandstream Budgetone and Mac mini?
Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --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
Re: [Asterisk-Users] Set caller ID for outgoing PRI calls
On Tuesday 28 March 2006 16:33, Tomislav Vojvodic wrote: > Is that what you were asking? My question is: how can I set specific caller id for outgoing PRI calls? pgpqCyPf2WORz.pgp Description: PGP signature ___ --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
Re: [Asterisk-Users] Set caller ID for outgoing PRI calls
On Tuesday 28 March 2006 15:40, Tomislav Vojvodic wrote: > It seems it's 'normal' behaviour since I heard exactly the same thing > happening in Croatia. If caller id is set to some number, telco > overrides it to first caller id.. (even if that number belongs to > your block (right?)) No. It sets Caller id to the first number only if it does not belong to my block. This is why I wish to set caller id myself when originating calls from office extensions. pgpLuMdDg4o23.pgp Description: PGP signature ___ --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] Set caller ID for outgoing PRI calls
Hallo! Finally we have E1 PRI connected to our Asterisk box. Now I have one question. My internal extensions (_XXX) are SIP phones connected to Asterisk. Our telco routes some public numbers (_71602XX and others) to our Asterisk via E1. Some internal extensions can be reached from outside using public numbers (e.g. 7160234 -> 200), and some others cannot. Everyone can call outside numbers from our network. How can I set caller id to something meaningful when they are calling outside? For example, 201 -> 7160201, 202 -> 3678685, etc. If I send incorrect caller id, my telco overrides it to first number from my block (7160200). pgpJwGHhvEuQ4.pgp Description: PGP signature ___ --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
Re: [Asterisk-Users] How to change Budgetone dialtone?
On Tuesday 07 March 2006 15:49, Lee Archer wrote: > Download the IP Phone Custom Ringtones Generation Tool > Unzip and read the readme Ringtone != dialtone. pgpEye3ebfT7t.pgp Description: PGP signature ___ --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] How to change Budgetone dialtone?
Good day! Is is possible to change dialtone (and other tones as well) in BT-102? pgpH067L1PT2h.pgp Description: PGP signature ___ --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
Re: [Asterisk-Users] (newby) Is PING a good indicator of latency?
On Wednesday 01 February 2006 14:12, Cosmin Prund wrote: > As the subject line says: Is PING a good indicator of network > latency? If not, how can I measure latency? Only if ICMP Echo has the same Class of Service (DSCP, 802.1p, priority/class in routers/shapers etc.) as VoIP traffic across the network. -- "The PSTN will never be a slave to you. You must be a slave to it." ___ --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
Re: [Asterisk-Users] Grandstream Budgetone mass deployment?
On Monday 30 January 2006 21:48, [EMAIL PROTECTED] wrote: > On Mon, 30 Jan 2006, Dmitry Ivanov wrote: > > I have created dynamic "CGI-like" TFTP server so I will create > > config files on-the-fly. Now we use this system (dynamic tftp > > server and Perl "CGI" script) for country-wide Sipura 3000 > > configuration. BTW, if anyone is interested I can send sources of > > this TFTP server. > > you know you can provision sipura 3000 via http, right? Yes, and I did it before TFTP. But some other equipment requires TFTP, and we decided to use single server. -- "The PSTN will never be a slave to you. You must be a slave to it." ___ --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
Re: [Asterisk-Users] Grandstream Budgetone mass deployment?
On Monday 30 January 2006 13:03, Phil Blundell wrote: > Personally I'd be a bit wary of mass Budgetone deployment for other > reasons, but the remote configuration stuff shouldn't be a problem. What reasons do you mean? > Grandstream use basically the same configuration file system for the > Budgetones as they do on the Handytones and the GXP-2000. Obviously > you need some way to make the files in the first place: when we > deployed our GXP-2000s I ended up writing a little Python script to > create the Grandstream config files (and the associated Asterisk > config entries) from input data in a Gnumeric spreadsheet. I have created dynamic "CGI-like" TFTP server so I will create config files on-the-fly. Now we use this system (dynamic tftp server and Perl "CGI" script) for country-wide Sipura 3000 configuration. BTW, if anyone is interested I can send sources of this TFTP server. -- "The PSTN will never be a slave to you. You must be a slave to it." ___ --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] Grandstream Budgetone mass deployment?
Hello! I am considering mass deployment of Budgetones 102. According to their website, remote provisioning (configuration via TFTP) is possible. Anyone has experience with this? Is this really working? ___ --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
Re: [Asterisk-Users] Asterisk Archives: BUG?
On Tuesday 10 January 2006 13:06, Jean-Michel Hiver wrote: > http://lists.digium.com/pipermail/asterisk-users/ > > May 2016? November 2007? Woot? Some kind of delayed Y2K bug? Randal Law lives in future. ___ --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] Is it Wildcard 406
Hello! After many troubles, I have received my Wildcard 406. There is a label on antistatic bag stating that this is 406. The card itself is marked as 405. Kernel modules shows in dmesg that card is 405. Is 406 the same as 405 with additional board installed? ___ --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
Re: [Asterisk-Users] Merry Xmas to everybody!
On Friday 23 December 2005 10:22, Mauro Zanin wrote: > Hi everybody, > > no issues this time. Only stopped to say: Merry Christmas and Happy > New Year. Yes, Merry Christmas, Happy New Year and Hanukkah :) Just received nice postcard from Digium :) ___ --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] Codec selection in dialplan
Is is possible to select (preferred) codec in dialplan using extensions.ael? For example, use 711 for extension 6004 (which is not "physical" extension) and 729 for anything else? ___ --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
Re: [Asterisk-Users] How to record a call
On Thursday 22 December 2005 07:36, Stefan Reuter wrote: > http://www.voip-info.org/wiki-Asterisk+cmd+Monitor For Asterisk 1.2: http://www.voip-info.org/wiki/view/MixMonitor ___ --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
Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX
On Wednesday 21 December 2005 15:11, [EMAIL PROTECTED] wrote: > Is Asterisk able to provide virtuals IPBX ? > I mean one hardware server which handle one IPBX per > enterprise . Yes, just create separate context for each enterprise. ___ --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] [offtopic] Asterisk <-IP-> Siemens HiPath 4000
Hello! Is it possible to connect Siemens HiPath 4000 to Asterisk? What equipment required on Siemens side? I mean IP not E1. Sorry for asking here. Siemens-related websites use "salesperson language". There is no technical information. ___ --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
Re: [Asterisk-Users] Asterisk 1.2 Released!
On Thursday 17 November 2005 12:58, Dmitry Ivanov wrote: > On Thursday 17 November 2005 12:32, Anton Krall wrote: > > Drumroll > > > > TADA!! > > Already compiling it :) Unlike beta2, it works for me. I hear no noise during echo test. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 Released!
On Thursday 17 November 2005 12:32, Anton Krall wrote: > Drumroll > > TADA!! Already compiling it :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/asterisk-users
Re: [Asterisk-Users] Asterisk hobby box
On Tuesday 15 November 2005 06:26, [EMAIL PROTECTED] wrote: > Hi. I'm setting up an Asterisk hobby box for me to play around with. > Is it possible to use a regular 56k modem and a regular home phone > for it? Yes, but forget G.711. BTW, some SIP-phones have built-in modem :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/asterisk-users
Re: [Asterisk-Users] Which Wildcard?
On Tuesday 08 November 2005 11:34, Hugh Jackman wrote: > Hi, > > Digium have Wildcard for FXO/FXS connections (i.e., telephone lines) > and E1/T1 cards such as the TE110p. There're a few things you might > want to consider: > > 1) TE110p is much more expensive > 2) it is too much for a "small shop". Concurrently supports upto 15 > incoming and 15 outgoing calls (or 30 incoming calls). Hence, the > number of clients can be up to 100, depending on your service needs > and configuration. > 3) you do need echo cancellation or your VoIP phone users will > suffer. The lastest Digium E1 card support hardware echo > cancellation. The builtin software echo cancellation is quite > incapable! > > Hope that helps! > > H. Thank you! 100 clients is not enough. Just ordered Wildcard 406 :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/asterisk-users
[Asterisk-Users] Which Wildcard?
Hello! We consider purchasing Digium Wildcard for E1 connectivity. Wildcards are pretty expensive pieces of silicon for small shop like ours. And we have no previous experience with E1 communications. What Wildcard do we need? How can we estimate our needs? How many clients (approx.) can share one E1 in practice, for example? What about hardware echo cancelation? Do we really need it? Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/asterisk-users
[Asterisk-Users] Noise in Echo()
Hello! First problem with 1.2-beta2. All I hear during Echo() is noise. No matter which codec selected. However, when using ulaw noise sounds better than g723 :) My equipment is Sipura SPA-3000. Works fine with 1.0.9 amd 1.0.7. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/asterisk-users
