Re: [Asterisk-Users] Distinctive Ring Detection not working
I have a similar problem in Australia and I think it has to do with chan_zap.c Currently Digium are investigating it for me as it is in association with one of their TDM400P cards. Gonzalo Servat wrote: Hi there. I'm having a strange issue with the distinctive ring detection in Asterisk (I have a FXO card). It certainly seems to be enabled as I can see the Asterisk console spitting out the cadences (same cadence every time: 0,0,0) but the problem is that it is not waiting 2 seconds after Starting switch on Zap/1-1 like it used to, long enough to determine the cadences, presumably the reason why it is always 0,0,0 as it hasn't had enough time to detect the ring pattern. My zapata.conf looks like the following: [trunkgroups] [channels] language=es context=incoming-landline signalling=fxs_ks usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 busydetect=yes busycount=8 dring1=334,146,0 dring1context=secondnumber channel = 1 I've looked through some of the chan_zap.c code to try and increase the wait period, but after making a couple of attempts at fixing it decided to leave it alone before I break something :-) Another thing I've noticed is that if I *don't* add a dring pattern for 0,0,0, when a call comes in, it tries to find the dring pattern for 0,0,0, fails to do so, so it tries to go to context ,s,1 (notice the missing context name as the first argument), fails to do so and it supposedly hangs up the chan, then detects the ringing again (it's still the same call, only in its 5th ring by now) and successfully detects a pattern different to 0,0,0. This is the only way to have it somewhat working, although it's pretty unreliable. It's coming up with quite a few different patterns, still, I shouldn't have to do it this way. A lot of people hang up after the 4th or 5th ring. Does anyone have any ideas on this? Any suggestions would be greatly appreciated. Cheers, Gonzalo ___ --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 -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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]how to send fax using Spandsp
Compile CVS HEAD and it's all built in. Andy Kuo wrote: Hi, I've been trying to get fax going for the last few days. I can receive fax with Spandsp 0.0.2pre19 on Asterisk 1.0.9 with rxfax, but when I tried sending the received fax file to a fax machine, I either get line error or just a blank page. Is anyone using Spandsp to send fax to fax machines on PSTN? I've run out of things to try now, and I'd really appreciate if anyone can share some ideas/experiences here. Thank you. AK ___ --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 -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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] Distinctive Ring Detection in AU
Has anyone got distinctive ring detection working for PSTN lines in Australia. I am using the latest CVS and have got zapata.conf set up thus: but it appears that the chan_zap modules is not going anywhere near that piece of code and all it returns is the default 0,0,0 [channels] context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes pulsedial = no rxgain = +10% txgain = 0 immediate = no busydetect = yes busycount = 6 progzone = au musiconhold = default usecallerid = yes sendcalleridafter=2 callerid = asreceived usedistinctiveringdetection = yes dring1=0,0,0 dring1context=default dring2=296,235,146 dring2context=default dring3=296,275,266 dring3context=default useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 4 -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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] Why n s priority in CVS but not in release?
I think one of the most important and flexible features of * is the ability to restructure the dial plan by using the n and s priorities. I cannot, for the life of me, see why they only exist in the CVS strand and not in the release strand; even the CVS of the release strand doesn't have these priorities. -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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] Why n s priority in CVS but not in release?
strand, branch, same difference. I mean the one that is available as: Çommands to get the current snapshot from the release branch of CVS: # cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds as opposed to: Commands to check out code from our CVS repository: # cvs checkout zaptel libpri asterisk So, when does a feature cease to be a feature, and what does it become? Kevin P. Fleming wrote: Howard Lowndes wrote: I cannot, for the life of me, see why they only exist in the CVS strand and not in the release strand; even the CVS of the release strand doesn't have these priorities. 'strand' ? We don't add features to the released versions of Asterisk. That's why this was not added to 1.0.x. ___ --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 -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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] Why n s priority in CVS but not in release?
To quote your earlier post: We don't add features to the released versions of Asterisk. That's why this was not added to 1.0.x. This implies that n s priorities are features since they are not in 1.0.x. So, when do they cease to be features and become a standard part of the released version. I have certainly found them to be both essential and stable. Kevin P. Fleming wrote: Howard Lowndes wrote: So, when does a feature cease to be a feature, and what does it become? I don't understand the question... ___ --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 -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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] Why n s priority in CVS but not in release?
Great :) Kevin P. Fleming wrote: Howard Lowndes wrote: This implies that n s priorities are features since they are not in 1.0.x. So, when do they cease to be features and become a standard part of the released version. I have certainly found them to be both essential and stable. They become part of a released version when the branch containing them becomes a release itself! We are currently beta testing 1.2 releases, which are made from the CVS HEAD branch, which contains these features. That means that the first actual release that will contain these features will be 1.2.0, scheduled for release within the next two weeks. ___ --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 -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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] Could someone look at channels/chan_zap.c
I'm banging my head against a brick wall trying to get CallerID
recognised in Australia.
I have CLID presentation enabled and I know that it works. I also have
distinctive ring tones enabled in zapata.conf
Around about line 5924 in channels/chan_zap.c is where the caller ID and
distinctive ring tone recognition starts for Bellcore FSK signalling
5924 } else if (p-use_callerid p-cid_start ==
CID_START_RING) {
5925 /* FSK Bell202 callerID */
5926 cs = callerid_new(cid_signalling);
and at line 5961 there is this comment:
5961 /* Let us
detect callerid when the telco uses distinctive ring */
but what follows appears to have no resemblence to identifying CLID.
The problem is that I cannot see, or work out what is supposed to go on
after that. I am getting distinctive ring tones but an not getting CLID.
Any help out there, or anyone who can explain what the code is supposed
to be doing?
--
Howard.
LANNet Computing Associates - Your Linux people http://lannet.com.au
--
When you just want a system that works, you choose Linux;
When you want a system that works, just, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.
___
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Re: [Asterisk-Users] Format of extensions.conf
It's also catering for the fact that ${FROM_DID} might be a string with
embedded spaces, and it's assuming, probably not unreasonably, that
${CALLERIDNUM} doesn't have embedded spaces.
David Tillman wrote:
In my (inherited) extensions.conf I have some lines of the format:
exten = o,2,GotoIf($[foo${FROM_DID} =
foo]?from-pstn,s,1:from-pstn,${FROM_DID},1)
and some lines like:
exten = s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4)
Note the quotes around the foo${FROM_DID} and foo in the
first, but the lack of quotes in the second.
Are these syntacticly equivalent?
In the first, is it just comparing to see if the string foo still equals
foo after the vars are interpolated?
Thanks,
-dave
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--
Howard.
LANNet Computing Associates - Your Linux people http://lannet.com.au
--
When you just want a system that works, you choose Linux;
When you want a system that works, just, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.
___
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[Asterisk-Users] Can someone please explain caller line identification
This is not a newbie question, and my problem may be related to Australia only or may be wider based. I have a PSTN line that has Caller ID presentation enabled. It used to work fine until recently, in as much as I could identify inbound CLID. There is/was a patch to * that suggested that in /channels/chan_zap.c the variable DEFAULT_CIDRINGS should be changed from 1 to 2 to suit Australian conditions and I had this done and everything worked. Recently I upgraded my kernel from 2.6.12 to 2.6.13 and did a clean recompile of * to suit, and to get the updated modules. BTW, * is CVS-HEAD of about 15 Sept. Since I recompiled * I have lost inbound CLID recognition but have gained the distinctive ring recognition ability which I previously didn't have. I still have the Australian mod in the chan_zap.c file, but I now note that the documentation indicates that this variable only applies to outbound CLID and not inbound CLID, or that is how I am reading the comment: /* Typically, how many rings before we should send Caller*ID */ /* #define DEFAULT_CIDRINGS 1 this needs to be set to 2 for Australia */ #define DEFAULT_CIDRINGS 2 [the 3rd 4th lines are my mod and are not in the original code] I also have a Wait(2) at the start of the relevant amswering dial plan as also recommended. I am not sure how CLID works technically, and the callerid.c code appears somewhat esoteric, so I would appreciate any assistance, esp from an Australian connection who has got inbound CLID working. -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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] Re: Asterisk Evening in Melbourne Australia!
Is there any chance anyone could discuss my post under Can someone please explain caller line identification. I live in Albury so I have no chance of getting to ML tonight. jurgen wrote: Just a quick reminder - this is happening *TONIGHT*. Hope to see all local Asteriskers come out (except PaulH, who went to great lengths to avoid us this time). jurgen On 14/10/05, jurgen [EMAIL PROTECTED] wrote: Hi all, Come out come out! If you're involved in Asterisk and live around the Melbourne area, please come out and join us for an evening of geeking out with Asterisk, socialising and generally having fun. Please note, people who have before, the venue has changed from last time because it was invaded by an annoying DJ. Date and time: Thursday October 20th at 7pm. Location: Mitre Tavern: http://www.melbournepubs.com/v/487/ If it's a warm evening, we'll be outside in the courtyard, but if it's not so warm, look for us inside. I'll bring along an old skool Telecom 9600 PABX phone and put it on the table. If anyone else has some classic technology, bring it along for a laugh. We've been thinking about doing a more geeky, less social evening as well, so we'll be talking about that - plus whatever else everyone has been up lately. Questions? Send them to [EMAIL PROTECTED], or give me a ring on 0415 276 127. See you there! ...jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ --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 -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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] Configuring TDM400 in Australia
Koolstart - see attached Rudolf Ladyzhenskii wrote: Hi, all I have installed TDM400 with 1 FXS and 1 FXP ports. Now I am goig through documentation on how to configure it. It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one do I use? Can someone send me sample zaptel.conf file for Australia? This will save me some time and will be used as a working example. Thanks, Rudolf ___ --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 -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ; $Log: zapata.conf,v $ ; Revision 1.4 2005/09/30 07:51:50 lannet ; Change the fax line to enable international dialling ; ; Revision 1.3 2005/09/13 16:46:08 lannet ; Define distinctive cal 0,0,0 and tie it to contect default ; ; Revision 1.2 2005/06/14 09:37:44 lannet ; Adjust RX TX volumes down by 5 points each ; ; Revision 1.1 2005/06/13 02:59:54 lannet ; Initial revision ; [trunkgroups] [channels] context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes pulsedial = no rxgain = +10% txgain = 0 immediate = no busydetect = yes busycount = 6 progzone = au musiconhold = default usecallerid = yes callerid = asreceived usedistinctiveringdetection = yes dring1=0,0,0 dring1context=default dring2=296,235,146 dring2context=default dring3=296,275,266 dring3context=default useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 4 callerid = House191 context = extn-standard signalling = fxo_ks rxgain = 0 txgain = 0 callwaiting = yes callwaitingcallerid = yes threewaycalling = yes transfer = yes hidecallerid = no usecallingpres = yes cancallforward = yes callreturn = yes usedistinctiveringdetection = no useincomingcalleridonzaptransfer = no faxdetect = no group = callgroup = 1 pickupgroup = 1 adsi = yes mailbox = 111 channel = 1 callerid = Fax192 context = extn-super rxgain = -5% txgain = +5% channel = 2 # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of 1. For a secondary, use 2, and so on. # To not use this as a sync source, just use 0 # # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) # # The framing is one of d4 or esf for T1 or cas or ccs for E1 # # Note: d4 could be referred to as sf or superframe # # The coding is one of ami or b8zs for T1 or ami or hdb3 for E1 # # E1's may have the additional keyword crc4 to enable CRC4 checking # # If the keyword yellow follows, yellow alarm is transmitted when no # channels are open. # #span=1,0,0,esf,b8zs #span=2,1,0,esf,b8zs #span=3,0,0,ccs,hdb3,crc4 # # Next come the dynamic span definitions, in the form: # dynamic=driver,address,numchans,timing # # Where driver is the name of the driver (e.g. eth), address is the # driver specific address (like a MAC for eth), numchans is the number # of channels, and timing is a timing priority, like for a normal span. # use 0 to not use this as a timing source, or prioritize them as # primary, secondard, etc. Note that you MUST have a REAL zaptel device # if you are not using external timing. # # dynamic=eth,eth0/00:02:b3:35:43:9c,24,0 # # Next come the definitions for using the channels. The format is: # device=channel list # # Valid devices are: # # em : Channel(s) are signalled using EM signalling (specific # implementation, such as Immediate, Wink, or Feature Group D # are handled by the userspace library). # fxsls : Channel(s) are signalled using FXS Loopstart protocol. # fxsgs : Channel(s) are signalled using FXS Groundstart protocol. # fxsks : Channel(s) are signalled using FXS Koolstart protocol. # fxols : Channel(s) are signalled using FXO Loopstart protocol. # fxogs : Channel(s) are signalled using FXO Groundstart protocol. # fxoks : Channel(s) are signalled using FXO Koolstart protocol. # sf : Channel(s) are signalled using in-band single freq tone. # Syntax as follows: #channel# =
[Asterisk-Users] CallerID presentation in Australia
This is not a newbie question... I have CLID presentation enabled on my PSTN service for some months now and it has worked fine until the other day - I could discover the CALLERID and hence could divert the telemarketers to voicemail. I then did a routine update of Linux and recomplied * - something I have done a few times before when there is a new kernel upgrade. Previously I had done make make install; this time I did make clean make make install and now - no inbound CLID recognition. My code is CVS-HEAD of about 15 Sep and I have checked that the Australian mod in chan_zap.c has been made. Another odd thing. With the previous compiles I was not able to get the distinctive ring recognition to work. Now I have dring - but no CLID. Has anyone any thoughts on this? -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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] DNS SRV supported phones
Grandstream supports DNS SRV Justin Richards wrote: Have you found any information yet about this? I am looking for good and affordable phones that can use DNS myself, but not for failover, simply for ease of use by some non-computer savvy family members. So far, I am afraid I'm going to be limited to USB/software phones. I would greatly appreciate any advice you can share. Thanks! On 7/21/05, *Anish Basu* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I am looking to use DNS SRV records for load balancing and failover across multiple Asterisk servers. The Asterisk servers share the exact same configuration via mySQL replication. I would like to know which particular SIP phones support DNS SRV and would like to hear of any success stories. Many SIP phones claim to support DNS SRV, yet there is usually very little documentation on how to configure it to do so. Any input is appreciated. Thanx in advance. Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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] Distinctive Ring Tones
This is an Australian situation. I have a PSTN connection that has CLID presentation enabled and has two numbers assigned to it, the primary number with the standard ring cadence: 400,200,400,2000 and the secondary number with the alternative cadence: 200,400,200,400,200,1600 CLID presentation is working fine and in zapata.conf I have: usecallerid = yes usedistinctiveringdetection = yes I am trying to set up the dring and dringcontext variables in zapata.conf and am trying to identify the returned codes for the two ring tones. Unfortunately, what gets returned in the asterisk console (verbose) is: -- Starting simple switch on 'Zap/4-1' -- Detected ring pattern: 0,0,0 for both ring cadences. I have looked at the code, chan_zap.c, and can see where this gets zeroed out, but not being a C programmer I am at a loss to identify what is not happening to get the correct numbers for the two situations. All cluesticks welcomed. -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --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] Undocumented exten syntax?
On Fri, 2005-03-18 at 08:34, Asterisk wrote:
John Goerzen wrote:
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten = s,1,SetVar(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,SetGlobalVar(EMERGENCY=1)
exten = s,n,SetVar(SET_EMERG_FLAG=1)
exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
Now, I have several questions:
* What is the n priority and how can they use it for several
different items? Don't they need an increasing integer there?
n stands for the next available sequence after the previous one. This is
used so that you can insert a step in the dial plan without having to
renumber all the following steps.
* What is the (checkavail) doing?
Thats a label. If you use n, you lose the ability to say goto 102
You can mix explicit priorities (123) with relative priorities (n)
* What does s+2 mean?
This priority + 2, thus n == s+1
ok. That's fooled me.
I've tried looking in docs and the wiki but can't figure it out.
Thanks!
-- John
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[Asterisk-Users] Does zapateller work in Australia?
as asked. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Providing a dialtone
On Thu, 2005-03-10 at 08:05, Martijn van Oosterhout wrote: Hi, I see applications for signalling busy, congested, ringing, progress etc, which I understand can be provided either in or out of band. But all I want to do is generate a dialtone. This obviously can only be done in band. There is code for generating the tones when you have a physical line, like the alsa channel, or a zap channel. But I'm just thinking of if they've selected an option that allows them to dial a normal number, to also provide a normal dialtone. Should I just record one and use Background()? I have a similar problem in as much as I want to provide a Facility dialtone to a zap channel under certain situations (call forward active) in the same way a Stutter dialtone is sent to a zap phone when there is a message waiting. Providing dialtone to SIP phones is probably not possible - I guess it is very phone dependent. Thanks in advance, -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] GotoIf problem
On Wed, 2005-03-09 at 05:29, kurt x wrote:
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is
this venture. When I dial and access voicemail with an ani of 3000
the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) . In additon when I dial with a 10 digit
ani the s,3,Gotoif does not work. It also goes through each line(
1,2,3,4,5,6,7)
Any help is greatly appreciated.
Have you tried removing the quotes?
Thanks
Kurt
Asterisk CVS-HEAD-07/14/04-16:28:29 built by
[EMAIL PROTECTED] on a i686 running Linux
[globals]
${ext}=0
SetGlobalVar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM}}) = ${DIGITS}]?s|5)
exten = s,4,GotoIf($[${CALLERIDNUM} = 3000]?s|6)
exten = s,5,Voicemail(u${ext})
exten = s,6,Background(pbx-invalid)
exten = s,7,Hangup
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Re: [Asterisk-Users] Call Forward or DND
On Tue, 2005-03-08 at 09:14, Anton Krall wrote: Guys. Im trying to implement some kind of call forward or DND, I checked the wiki and there are some examples of call forwards but I was wondering if anybody has implemented one that will let you forward calls to SIP, IX or ZAP channels alike? For example, forwardto another extension, to an outside number or directly to voicemail (ala DND). Yes. Give them an extension to prompt for, and store, the call forward number. Also an extension to cancel call forward. Use the * internal database to store the call forward numbers against the extension that they are setting it up from. If the call forward number is the same extension that they are setting up from, then assume they want DND and code the dial plan to check for this and do a redirect to voicemail. Thx! ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Call Forward or DND
On Tue, 2005-03-08 at 10:48, Anton Krall wrote:
Nice idea.. Now, also We would need to check the number of digitsentered, if
more than X, then call is an outside number, is less than X, then its an
internal extension..
Simple.
SetGlobalVar(DIGITS=4)
GotoIf($[${LEN(${EXTEN}) ${DIGITS}]?s-ext:s-int)
Sounds good?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: Lunes, 07 de Marzo de 2005 04:43 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Forward or DND
On Tue, 2005-03-08 at 09:14, Anton Krall wrote:
Guys.
Im trying to implement some kind of call forward or DND, I checked the
wiki and there are some examples of call forwards but I was wondering
if anybody has implemented one that will let you forward calls to SIP,
IX or ZAP channels alike? For example, forwardto another extension, to
an outside number or directly to voicemail (ala DND).
Yes.
Give them an extension to prompt for, and store, the call forward number.
Also an extension to cancel call forward.
Use the * internal database to store the call forward numbers against the
extension that they are setting it up from.
If the call forward number is the same extension that they are setting up
from, then assume they want DND and code the dial plan to check for this and
do a redirect to voicemail.
Thx!
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--
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--
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--
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--
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RE: [Asterisk-Users] Call Forward or DND
On Tue, 2005-03-08 at 11:43, Anton Krall wrote:
Wow, too professional for me hahaha can you explain to me the last part of
the goto?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: Lunes, 07 de Marzo de 2005 06:22 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call Forward or DND
On Tue, 2005-03-08 at 10:48, Anton Krall wrote:
Nice idea.. Now, also We would need to check the number of
digitsentered, if more than X, then call is an outside number, is less
than X, then its an internal extension..
Simple.
SetGlobalVar(DIGITS=4)
GotoIf($[${LEN(${EXTEN}) ${DIGITS}]?s-ext:s-int)
If the length ${LEN()} of the dialout extension ${EXTEN} is greater
than some parameter you define ${DIGITS} then go to ? somewhere
in the dial plan that handles external call outs, otherwise : go to
somewhere in the dial plan that handles internal call outs.
The SetGlobalVar() is not necessary if you have defined DIGITS = 4 in
the [globals] section of your dial plan.
s-ext and s-int can be stated:
either as a priority point in the current extension in the current
context,
10
or as a priority point in a different extension in the current context,
1234|10
or as a priority point in a different extension in a different context.
other-context|1234|10
Check out the GotoIf() application.
Sounds good?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard
Lowndes
Sent: Lunes, 07 de Marzo de 2005 04:43 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Forward or DND
On Tue, 2005-03-08 at 09:14, Anton Krall wrote:
Guys.
Im trying to implement some kind of call forward or DND, I checked
the wiki and there are some examples of call forwards but I was
wondering if anybody has implemented one that will let you forward
calls to SIP, IX or ZAP channels alike? For example, forwardto
another extension, to an outside number or directly to voicemail (ala
DND).
Yes.
Give them an extension to prompt for, and store, the call forward number.
Also an extension to cancel call forward.
Use the * internal database to store the call forward numbers against
the extension that they are setting it up from.
If the call forward number is the same extension that they are setting
up from, then assume they want DND and code the dial plan to check for
this and do a redirect to voicemail.
Thx!
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--
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--
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--
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Re: [Asterisk-Users] newbie questions
On Tue, 2005-03-08 at 14:57, Brian Nehring wrote: I actually got X-Lite talking to the server, finally. I didn't have to change any of my Asterisk servers... I just kept fooling around with X-Lite and watching the diagnostics log and it finally worked. I can't really say what fixed it, I don't even feel like I changed anything. Oh well, thanks for all the advice, the Diagnostics Log in X-Lite and running Asterisk with -vgcd helped quite a bit. [EMAIL PROTECTED] also looks great, I'm going to install that tomorrow, hopefully the GUI will ease some of the learning curve. Is Xlite running on Windows or Linux? -Brian ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] newbie questions
On Tue, 2005-03-08 at 16:56, Brian Nehring wrote: Xlite for OS X actually. bummer, I've been wanting to get it running under Linux. On Tue, 08 Mar 2005 15:00:24 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: On Tue, 2005-03-08 at 14:57, Brian Nehring wrote: I actually got X-Lite talking to the server, finally. I didn't have to change any of my Asterisk servers... I just kept fooling around with X-Lite and watching the diagnostics log and it finally worked. I can't really say what fixed it, I don't even feel like I changed anything. Oh well, thanks for all the advice, the Diagnostics Log in X-Lite and running Asterisk with -vgcd helped quite a bit. [EMAIL PROTECTED] also looks great, I'm going to install that tomorrow, hopefully the GUI will ease some of the learning curve. Is Xlite running on Windows or Linux? -Brian ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] voicemail volume
On Mon, 2005-03-07 at 09:02, David Newman wrote: On Sun, 6 Mar 2005, Marty Mastera wrote: The full text of the bug you reference above indicates that pstnVMgain was (or is) part of an ongoing feature request/bug report and has not been implemented for use at this time (and may never be). Right. So -- what can I do to boost volume of PSTN - * voicemail? Assuming you are using a zap interface for the PSTN connection, could you try increasing the rx gain. Is your incoming volume low anyway? thanks dn ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Stutter Tone
On Sat, 2005-03-05 at 14:10, Anton Krall wrote: I think I have something misconfigured regarding voicemails. They work great, I have this setup: Sip.conf [ext1] Context=phones Mailbox=201 Voicemail.conf [home] 201,password,name,[EMAIL PROTECTED] Voicemail delivery and all works great but when I check sip extension ext1 (analog phone using a Granstream ATA 286), the stutter tone signaling message waiting does not work. Anything wrong with contexts or something? Stutter tome works with Zap but does it work with SIP phones unless they have their own stutter tome which activates when they get a NEW MESSAGE header. My HOP 1002 SIP phones certainly don't have that. Thx Guys ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Sending Voicemail's to two email addresses
On Thu, 2005-03-03 at 06:32, Randy Johnson wrote: Is there a way to send a voicemail to two different email addresses when a caller leaves a message? Does address1, address2 work or does it get confused about the ,? Thanks a bunch! Randy ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] CallerID in Australia Analogue PSTN PhoneSystem
On Thu, 2005-03-03 at 08:14, Rod Bacon wrote: I have the openline card with a recent CVS and I can't seem to get CallerID to work. Debugging shows; vpb/1-1: New call for context [local_pstn] Caller ID disabled Do you have caller line identification presentation enabled for your telco line? In most cases it's not enabled by default, and Telstra/Sensis charge you AU$6/month for the privilege. vpb.conf is... [general] type = v4pci cards = 1 [interfaces] echocancel = on board = 1 txhwgain = 12 txgain = 12 context = local_pstn mode = fxo group=1 callerid = on channel = 1 channel = 2 channel = 3 channel = 4 Any ideas? - Original Message - From: James Andrewartha [EMAIL PROTECTED] Sent: Tuesday, January 04, 2005 11:50 PM Subject: [Asterisk-Users] CallerID in Australia Analogue PSTN PhoneSystem Howard Lowndes wrote: Is there anyone using * in AU that has successfully extracted the CLID from an incoming analogue PSTN phone call, and would like to spread the word? Yes, I have. I'm using a Voicetronix OpenLine4 card. I did have an issue with the config file, you have to set callerid=yes before each channel, unless you're running CVS from 2004/12/13 21:04:12 or later. What hardware are you using? chan_vpb has useful debugging info for callerid at debug level 4. James Andrewartha DAA Sysadmin ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] n priority not in 1.0.6
Does anyone know why the n priority in the dial plan is not recognised in 1.0.6 It seems strange to me that it should be so. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Call waiting in Australia
Has anyone had problems with Call Waiting signals causing Zap channel or bridging hangups in AU. I was on a call the other day (Zap channel to PSTN) and the call suddenly hung up on my side. I dialled the calling party and got the call again, it seems that the bridge had dropped and that the other party had not lost the connection. As soon as I got the bridging again the other party mentioned that they had had a call waiting signal immediately before I went off the air. Any one had similar experiences, or have fixes? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Two offices connection
On Mon, 2005-02-28 at 20:38, Azhar Chowdhury wrote: I would like connect two offices where one office have 4 PSTN Analog lines and another office without any PSTN. Both the offices will have two separate Asterisk server with TDM400P cards (4 ports FXS FXO). My questions is that how to configure Asterisk to forward the PSTN calls directly to another Asterisk which has the TDM400P card without pressing the extension number. Use the I'net to connect the two offices and IAX2 Diagram like following ---PSTN line1 --[Asterisk]__WAN__[Asterisk ]-Phone Set1 ---PSTN line x -[TDM400P] [TDM400P] Phone Set1 So, call coming from PSTN should go directly to Phone Set1 without any Extension. Is it possible, if so,please let me know how to configure both Asterisk server? Thanking you, Azhar -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] 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, 2005-03-01 at 06:40, 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 concerns, though, is if we get DDoS'd - which happens probably once every couple of years. I'd like to have the dialplan failover to PSTN to shunt calls to the PSTN---User's cell number in the case of a DDoS attack. My current thinking is K.I.S.S - just put the user's cell as the next step in the dialplan. However, I'd like for this to be controllable - when things are working OK, I don't want the calls being routed to the cells *at all*. I also don't want to have an extensions.conf and an extensions_emergency.conf and do the _emergency as an commented out include. I'd like for it to be automatic i.e. Asterisk detects Internet latency is above a certain threshold, then automagically does the cell thing. Any suggestions? I fooled around in Google for about a half hour on this, and of course the Wiki was no help. TIA ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] 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,DBGET(TRUE=DDOS/yes) exten = s,2,Do this exten =) s,102,do something else My comment: Good suggestion, but requires user intervention. I'm lazy and I want it to be totally transparent. I'm not avaliable most of the time and training someone to do it is not reliable, even my MCSE monkey would have trouble figuring out that we are being DoS'd (NOT my hire!) -and- 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 My comment: Better suggestion, and looks to be workable. What would be a good way to detect latency? A cron job that pings a known host with, say, 2K of data and pipes it back to a shell script? If so, what kind of frequency would be considered effective? Every 30 seconds, 1 minute? You would probably need to have 2-3 positive samples over a period of, say, 3 minutes before you triggered the change, otherwise it runs the risk of being too volatile. A short packet ping would probably be better than a long packet transfer. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] 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, some sort of Debian nut? 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.jpg for an example. If I do asterisk -rv on a normal login, either via the console or an xterm, the text appears correctly. Does anyone have any ideas what is causing this and how to fix it? Thanks Tony -- Kristian Kielhofner ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Jumb between macro's and variables
On Mon, 2005-02-28 at 05:58, Riphagen, Ferdy wrote:
Hello All,
I have a macro and want to jump to another macro if a conditition is true or
false.
Asterisk is jumping to the next macro, but then the {ARG1} variable is not
working anymore.
Try SetVar(SAVEARG=${ARG1}) in one macro then reference it in the next.
part of config:
[macro-default]
exten = s,1,DBGet(do-not-disturb=DND/${ARG1})
exten = s,2,GotoIf($[${do-not-disturb} = YES]?macro-do_not_disturb,s,1)
...
[macro-do_not_disturb]
exten = s,1,Wait(2)
exten = s,2,Answer
exten = s,3,Playback(pls-try-call-later)
exten = s,4,Voicemail(u${ARG1})
exten = s,5,Hangup
In the asterisk log I see this error for s,4 in macro-do_not_disturb:
-- Executing VoiceMail(SIP/201-6440, u) in new stack
Feb 27 19:44:32 WARNING[1202]: app_voicemail.c:1540 leave_voicemail: No
entry in voicemail config file for ''
What is wrong?
Thanks,
--Ferdy
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Re: [Asterisk-Users] Minimal hardware requirements
On Wed, 2005-02-23 at 01:52, Mark Eissler wrote: On Feb 21, 2005, at 7:35 PM, Rudolf Ladyzhenskii wrote: Hi, all I am doing prrof of concept system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration. At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be enough to demonstrate? Scrap any analog connections. Get a VOIP SIP adapter to handle analog. Setup VOIP to PSTN termination via one of the many providers. If they are available - not so usually outside of capital cities (in AU at least) No need for TDM cards. Your system will work fine. You can have a problem if you make emergency calls over IP - the call is focussed on the PSTN drop off point, so may not be local. Consider the risk of bigger installations. I've done this on an old Pentium Pro with only 128MBs for a small system. You only need bigger hardware if you're going to add many more users. Two SIP phones is nothing. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Minimal hardware requirements
On Tue, 2005-02-22 at 11:41, John Novack ( Mozilla - portable ) wrote: Rudolf Ladyzhenskii wrote: Hi, all I am doing prrof of concept system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration. At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be enough to demonstrate? Thanks, Rudolf Depends. If you plan on using the TDM400 with one each FXS and FXO, the MB needs to have PCI Ver 2.2 slots, or the card won't be seen Any MB made after 2000 probably is OK Well, I have a PII 300 of about 2000 vintage and that didn't work with the TDM400P card. The PCI 2.2 spec was announced in Jan 2000 so it would take a while for it to filter thru to actual mobos. John Novack ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] How to announce the DNID to the called party
On Mon, 2005-02-21 at 15:50, dkwok wrote: How to announce the DNID to the called party who picks up the phone and say the correct greeting? I suppose it has to say to the called party before the call is bridged. So it has to do something before the dial command transfer the call. Any ideas? Check out the A option to the Dial command. David Kwok ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] festival text for weather report
On Thu, 2005-02-17 at 15:24, dean collins wrote: http://www.srh.noaa.gov/fwd/productviewnation.php?pil=OKXZFPOKXversion=0 can anyone suggest how I could set up [EMAIL PROTECTED] to read out allowed the following text when I dial extension 850? 815 PM EST WED FEB 16 2005 .OVERNIGHT...MOSTLY CLEAR. LOWS 30 TO 35. NORTHWEST WINDS 15 TO 20 MPH WITH GUSTS UP TO 30 MPH...DIMINISHING TO 10 TO 15 MPH LATE. .THURSDAY...PARTLY CLOUDY. COOLER WITH HIGHS AROUND 40. NORTHWEST WINDS AROUND 15 MPH. .THURSDAY NIGHT...PARTLY CLOUDY. LOWS IN THE MID 20S. WEST WINDS AROUND 15 MPH. .FRIDAY...PARTLY CLOUDY AND BRISK. HIGHS IN THE MID 30S. NORTHWEST WINDS 15 TO 25 MPH. .FRIDAY NIGHT...PARTLY CLOUDY AND BRISK. LOWS AROUND 17. NORTHWEST WINDS 15 TO 25 MPH. Basically you are going to have to parse this text with something (Perl springs to mind) so that you can break it into separate files in order to move back firth in it. Then you are going to need to adjust it so that MPH reads miles per hour etc. Goodluck. :) Theres $20 via paypal to the first person to help me complete this (Ill then post it on the the wiki so anyone can replicate it) (anyone wanting to add to that bounty email me) Also if it is not too difficult Id like it to skip to the next block each time you press 1 (eg go from overnight to Thursday) Also it doesnt need to be this particular web page that it connects to but something with current weather etc. Cheers, Dean __ ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zap/g0/ to a Telstra Mobile
On Thu, 2005-02-17 at 15:51, [EMAIL PROTECTED] wrote: I've installed a TDM400. Having a go with AMP. I would like incoming calls to be put throuhg to an extension (at my desk) and a mobile (cell phone) at the same time. Whichever picks up, gets the call.. This should be possible with AMP (call groups, 200|201|0*0408xx), but it didn't work, so I have created a custom-incoming in extensions-custom.conf What is happening is, The extension rings for about 5 secs (as long as it takes the TDM400 to dial the mobile number), then just the telstra mobile rings.. From asterisk -vvvr -- Goto (custom-incoming,s,1) -- Executing Dial(SIP/202-b424, Zap/g0/0408xxSip/200|30|t) in new stack -- Called g0/0408xx -- Called 200 -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- Zap/2-1 answered SIP/202-b424 This tend to indicate to me that the mobile system has picked up the call request on the zap channel and that * therefore thinks that the zap channel has picked up the call and will then bridge the zap channel to the sip 202 channel and kill off the ringing on the sip 200 channel. I don't know that there is much you can do about this as basically you are trying to get interaction on two different systems. At this stage the mobile is still ringing and has not been answered. Below are zapata.conf extensions-custom.conf Any thoughts anyone? Cheers Shane ---zapata.conf--- language=en context=from-pstn signalling=fxs_ks ;stripmsd=1 immediate=no overlapdial=yes faxdetect=no usecallerid=no echocancel=yes callprogress=yes busydetect=yes busycount=6 echocancelwhenbridged=no echotraining=800 rxgain=5.5 group=0 channel=2 channel=3 group=1 usecallerid=yes channel=4 ---extensions-custom.conf--- [custom-incoming] exten = s,1,Dial(Zap/g0/0408xxSip/200,30,t) exten = s,104,Voicemail(u200) exten = s,105,Hangup() ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] A hypothetical question...
On Wed, 2005-02-16 at 09:33, Stefan Gofferje wrote: Rod Bacon schrieb: I know this is casting a wide net, but If you were charged with building a large, public VOIP network with multiple PSTN gateways, the capacity to carry a lot of traffic and bill clients accurately, what pieces (brands, makes, models) would you use to assemble the solution? Assume that $$$ is not an issue. If money is no matter? Cisco... Callmanager Cluster, Gateways... ...and the reports are that you will be throwing $$$s at crap. Endpoints may be chosen freely. --Stefan -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 no one is available to take your call
On Wed, 2005-02-16 at 11:05, Greg Oliver wrote: OK - I can successfully make calls from SIp phone through an asterisk 323 channel to a Cisco Call Manager and out a MGCP controlled gateway. The problem is that if the call is not answered within ~5 seconds, * gives the message no one is available to take your call and disconnects the call. If I answer b4 the 5 seconds - everything is good. Anywhere I need to set to get around this. I have tried the t,T settings (even though the docs say no entry is forever) with no luck. Read the doco on the Dial command again. It's noting to do with the Tt option, it's the parameter before that that you need to set to the timeout Thanks, Greg Oliver ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 IP phone to use in Australia
On Wed, 2005-02-16 at 13:07, Shaun Ewing wrote: On Wed, 16 Feb 2005 12:20:00 +1100, Paul Hales [EMAIL PROTECTED] wrote: Regarding your quote about Polycom - I'm not sure what you mean by 'Polycom won't sell...' We have over 100 polycom's out and about, all hooked into our 3 Asterisk servers. I will admit that I haven't enquired with Polycom, but I've read numerous times on this list and other places (can't think of references off the top of my head) that they'll only officially sell the phones if it's to be used with an approved softswitch. Two points: 1. How are they to know 2. In AU that is contrary to the Trade Practices Act Not sure if that's still the case though. -Shaun ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 IP phone to use in Australia
On Wed, 2005-02-16 at 13:14, Shaun Ewing wrote: On Wed, 16 Feb 2005 09:23:21 +0800, Stuart Elvish [EMAIL PROTECTED] wrote: Definitely agree - don't even try using the Grandstream for a receptionist (among other things the phone probably won't hold out physically for more than a few weeks if it makes it that far). :-) They have recently been ticked as well, plus the firmware has become some what stable that having been said I am not sure when the last update came out and it does have a couple of quirks. We have the system time out (or send the dialed digit string) after 4 seconds of no dialing which works well - but that depends on the user environment and what they expect from the phone system. The other problem is that Grandstream don't display any type of alpha caller id - they are purely a digit based caller ID presentation (it tries to present an alpha sequence but it doesn't work at all). The lack of alpha caller ID is a downside. We're using the alpha string for all sorts of things, eg: to display the trunk a call came in on Private Line, a queue QUEUE: Sales, in addition to the name of the caller where supported. It's certainly noticeable when absent. Don't get me wrong - they are still the bottom of the range / basic phone IMHO and Cisco do seem to work a lot better, but are also more expensive and my boss won't pay for one. They are more expensive, which is a downside to the Cisco phones. I bit the bullet and bought a few varying models, but it was a bit of a financial hit. I have the final say on company purchases, so there is no boss to contend with. What sort of setup is involved for the Cisco as far as config files etc? I am used to plug and play phones (Zyxel, Grandstream, HOP etc) which require minimal configuration and have no licensing issues with them. I know for the Polycom you need to get a TFTP server for XML config files running, and I believe you need something similar for Cisco phones. You'll need a TFTP server to get the SIP firmware on the phone. For small deployments you can configure the options on the phone itself, but for anything more than 2 phones, I'd recommend a TFTP server. With a working HOP 1002 phone they have their own web server inbuilt and you can upgrade the firmware with that. Mind you, on a couple of occasions, particularly when trying to upgrade the firmware remotely, it's screwed up and I had to reset back to default and use the Palmtool software (Windows only - barf) to fix things up. Stuart -Shaun ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Re: Festival Woes
On Tue, 2005-02-15 at 15:55, Brian Dingman wrote: Wow. I posted that a long time ago. Thanks. Festival doesn't seem very stable to me though. Works fine for me, but I think the non-US accents need some work. On Tue, 15 Feb 2005 15:14:47 +1100, Rod Bacon [EMAIL PROTECTED] wrote: SIOD ERROR: wrong type of argument to car : wholeutt Try changing your festival.scm to the following: (Note the extra () on the 4th last line). (define (tts_textasterisk string mode) (tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe functions. (let ((wholeutt (utt.synth (eval (list 'Utterance 'Text string) (utt.wave.resample wholeutt 8000) (utt.wave.rescale wholeutt 5) (utt.send.wave.client wholeutt))) ___ 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 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 IP phone to use in Australia
On Tue, 2005-02-15 at 17:13, Rudolf Ladyzhenskii wrote: Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I use the HOP 1002 from IP Trading in Sydney - I think they call it the Vision - $149 inc GST. Get back to me if you need to upgrade the software - which you may need to do if it's not running V 1.41 I need actual handsets. I need standard handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by receptionist. The main problem is that I am in Australia and I need to get phones that can be sourced in Australia. (correct power supplies, certified for australia, etc..) I did look at supported h/w list and I am going to go through all of those companies, but I have no idea on how good/bad those phones are. I really need some advise here. Thanks, Rudolf /***/ Rudolf Ladyzhenskii Senior Design Engineer Open Networks Pty. Ltd. Level 26, 35 Collins Street, Melbourne VIC 3000 e-mail: [EMAIL PROTECTED] phone: +61 3 9656 5107 fax: +61 3 9656 5122 web: www.opennw.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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 IP phone to use in Australia
On Tue, 2005-02-15 at 17:26, Paul Hales wrote: The Asterisk meeting in Melbourne Thursday night would be a good place to discuss this! Not if: 1. You don't know about it 2. You're not Melb based. Regards, regards, PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf Ladyzhenskii Sent: Tuesday, 15 February 2005 5:14 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which IP phone to use in Australia Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need standard handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by receptionist. The main problem is that I am in Australia and I need to get phones that can be sourced in Australia. (correct power supplies, certified for australia, etc..) I did look at supported h/w list and I am going to go through all of those companies, but I have no idea on how good/bad those phones are. I really need some advise here. Thanks, Rudolf /***/ Rudolf Ladyzhenskii Senior Design Engineer Open Networks Pty. Ltd. Level 26, 35 Collins Street, Melbourne VIC 3000 e-mail: [EMAIL PROTECTED] phone: +61 3 9656 5107 fax: +61 3 9656 5122 web: www.opennw.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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 IP phone to use in Australia
On Tue, 2005-02-15 at 17:43, Adam Goryachev wrote: On Tue, 2005-02-15 at 17:13 +1100, Rudolf Ladyzhenskii wrote: Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need standard handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by receptionist. The main problem is that I am in Australia and I need to get phones that can be sourced in Australia. (correct power supplies, certified for australia, etc..) I did look at supported h/w list and I am going to go through all of those companies, but I have no idea on how good/bad those phones are. I really need some advise here. Personally, I quite like the polycom phones such as the IP300 and IP600 I've never really bothered with the IP500... They are pretty good as far as admin functionality (control based on FTP files) as well as very functional (user side) and very good quality (low fail rate, look/feel good, excellent audio, etc) There are a few issues I have with them though, the main one being that I can't disable call waiting on the phone. There are workarounds for this though (in asterisk dialplan). ...which is something to be said for the HOP 1002 - you can disable call waiting. Just my 0.02c worth. Regards, Adam -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 IP phone to use in Australia
On Tue, 2005-02-15 at 18:05, Adam Goryachev wrote: On Tue, 2005-02-15 at 17:54 +1100, Howard Lowndes wrote: On Tue, 2005-02-15 at 17:43, Adam Goryachev wrote: Personally, I quite like the polycom phones such as the IP300 and IP600 I've never really bothered with the IP500... There are a few issues I have with them though, the main one being that I can't disable call waiting on the phone. There are workarounds for this though (in asterisk dialplan). ...which is something to be said for the HOP 1002 - you can disable call waiting. Have you actually used the polycom phones? If so, how do they compare to the HOP 1002, or, would you call the polycom IP600 and HOP 1002 exactly equivalent in all respects except for the call waiting factor? Unfortunately I have never used, or even seen the polycom phones, so I cannot comment on the comparison. I do know that the HOP 1002 serve my purpose and are quite robust. There was a date issue with the software pre v1.41.007 and I have found out how to get a brand name to display on the screen. I have also discovered that, under SIP at least, the phone will only display the caller ID number and not the caller ID name, though that latter is not often sent anyway except for calls from mobiles as MOBILE. Basically they are very robust, almost brick shithouse robust. :) The online manual is about 47 pages of Chinglish which is an Alexander (downer). (Oz joke there for all you yanks) The only down side that I can see is that the 2 port hubbing is only 10 mbps which shouldn't really be a problem for most users who connect their PC in line, but could be a real bummer for the power user PHBs who want to do gaming. I've not seen/used the HOP 1002, I just find it hard to accept that it would be as good as the polycom IP600 phones Note: I would be *pleasantly* surprised if you say it is as good! Regards, Adam -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Who makes these phones?
On Mon, 2005-02-14 at 10:10, Gary wrote: On Mon, 14 Feb 2005 09:53:36 +1100, PHP Mechanic wrote: http://www.broadbandphone.com.au/global/pnp.htm They look like they are all PA1688 based. The black one is a dead copy of the one sitting on my desk, made by Hirakawa Electronics according to the label underneath. The middle white one looks similar - dunno out the other white one. ...and yes, they are PA1688 based. Gary . ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Who makes these phones?
On Mon, 2005-02-14 at 13:52, Craig wrote: Message: 1 Date: Mon, 14 Feb 2005 09:53:36 +1100 From: PHP Mechanic [EMAIL PROTECTED] Subject: [Asterisk-Users] Who makes these phones? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; charset=iso-8859-1; reply-type=original http://www.broadbandphone.com.au/global/pnp.htm they are called a Kitty Ethernet Phone, seem to be available in 3 or 4 models but with identical Guts. The only info I have found on them is Gateway Technologies, supposedly the Chinese manufacturer website... http://www.ipgw.net/EN/index.htm I bought one off a guy who is flogging them in Au for about $90 each. That's cheaper than from IP Trading in Sydney. Nice looking, cheap ip phone. But information manual are next to useless. Try aredfox.com The only technical info I have been able to find is the 8 page manual that comes with it (copy on website) which tells you nothing. I haven't yet tried it live, still working out how to set it up. Seems to have features like talking speed dial etc, ...haven't found that yet - how did you find out about it. but haven't yet worked out how to drive the functions and manual is less than helpful. Would appreciate if anybody has already managed to get one of these working and would like to share the setup and how to use the functions on them. Contact me OL howard at lannet dot com dot au Regards, Craig ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Playing Dialtones
In AU we have a number of different dialtones defined for various purposes. From indications.conf: au ringcadance 400,200,400,2000 au dial413+438 au busy425/375,0/375 au ring413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au callwaiting 425/200,0/200,425/200,0/4400 au dialrecall 413+438 au record !425/1000,!0/15000,425/360,0/15000 au info425/2500,0/500 au std !525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100 au facility425 au stutter 413+438/100,0/40 au ringmobile 400+450/400,0/200,400+450/400,0/2000 With analogue handsets connected to a zapata interface the standard dialtone in the off-hook condition is dial, but if there is a message waiting then the stutter dialtone is sent out when the handset goes off hook. These emulate behavour that occurs on the PSTN system. With the PSTN service the facility dialtone is output if the user has a facility set, such as call forward immediate. I have a routine in my dialplan which allows the user to set up CFIM, but what I want it for * to output the facility dialtone when the use user picks up the handset for which a CFIM number is set. Identifying that a CFIM number is set is not a problem in the dialplan, but when I run * in verbose mode all I am seeing, when the handset is lifted, is: -- Starting simple switch on 'Zap/2-1' Does this mean that the context for that extension has started and that if I put a Playtones command in as the first command in the context, then I will get what I want to achieve, or do I need to tackle it from some other direction? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Autodetecting faxes
On Tue, 2005-02-08 at 09:49, Brian Dingman wrote: I am having the same problems. No matter what I try, * won't detect faxes. I have faxdetect=both in zaptel.conf and my extensions.conf looks like this: [fromPSTN] exten = s,1,Answer exten = s,2,DigitTimeout(2) exten = s,3,ResponseTimeout(10) exten = s,4,Wait(3) exten = s,5,Background(custom/ivr-greeting) exten = i,1,Wait(1) exten = i,2,Background(pbx-invalid) ; That's not valid, try again exten = i,3,Goto(s,5) exten = t,1,Background(goodbye) exten = t,2,Hangup exten = fax,1,Goto(fax,s,1) You do have an exten called fax...? On Sun, 06 Feb 2005 20:42:07 +, Adrian Chapman [EMAIL PROTECTED] wrote: What we found was that the fax/voice decision was being made before the intermittent beep--beep--beep fax tone was being generated, so it wasn't being detected. Changing the order of things in extensions.conf around a smidge got it all working nicely :- [inbound-from-pstn] include = default exten = s,1,Answer exten = s,2,Wait,1 exten = s,3,Playback(thank-you-for-calling-please-wait-a-moment) exten = fax,1,Macro(faxreceive) exten = s,4,Do the normal phone call gubbins The wait allows the start of the Playback to be heard by the caller - without it, we were finding the first word clipped. That second plus the duration of the Thank you for calling message gives enough time for the roughly 2.5sec duration between fax beeps to repeat, no matter when it last fell compared to the answer. We've not checked more into the three rings before answer, but there's been discussion (here? elsewhere?) that it's down to the wait for caller ID. Try turning that off. TBH, I *like* the three rings - as a caller, it psychologically gives you time to get your head in gear before the call's answered. Besides - If you're ringing from a mobile, it also gives you time to physically put the phone to your ear... -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ 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 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Autodetecting faxes
On Tue, 2005-02-08 at 09:49, Brian Dingman wrote: I am having the same problems. No matter what I try, * won't detect faxes. I have faxdetect=both in zaptel.conf and my extensions.conf looks like this: [fromPSTN] exten = s,1,Answer exten = s,2,DigitTimeout(2) exten = s,3,ResponseTimeout(10) exten = s,4,Wait(3) exten = s,5,Background(custom/ivr-greeting) exten = i,1,Wait(1) exten = i,2,Background(pbx-invalid) ; That's not valid, try again exten = i,3,Goto(s,5) exten = t,1,Background(goodbye) exten = t,2,Hangup exten = fax,1,Goto(fax,s,1) Apologies. I meant You do have a context called Fax...? On Sun, 06 Feb 2005 20:42:07 +, Adrian Chapman [EMAIL PROTECTED] wrote: What we found was that the fax/voice decision was being made before the intermittent beep--beep--beep fax tone was being generated, so it wasn't being detected. Changing the order of things in extensions.conf around a smidge got it all working nicely :- [inbound-from-pstn] include = default exten = s,1,Answer exten = s,2,Wait,1 exten = s,3,Playback(thank-you-for-calling-please-wait-a-moment) exten = fax,1,Macro(faxreceive) exten = s,4,Do the normal phone call gubbins The wait allows the start of the Playback to be heard by the caller - without it, we were finding the first word clipped. That second plus the duration of the Thank you for calling message gives enough time for the roughly 2.5sec duration between fax beeps to repeat, no matter when it last fell compared to the answer. We've not checked more into the three rings before answer, but there's been discussion (here? elsewhere?) that it's down to the wait for caller ID. Try turning that off. TBH, I *like* the three rings - as a caller, it psychologically gives you time to get your head in gear before the call's answered. Besides - If you're ringing from a mobile, it also gives you time to physically put the phone to your ear... -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ 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 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] AU caller ID with Sipura SPA-3000
On Sat, 2005-02-05 at 08:28, Eric Bishop wrote: Hi All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using PSTN Ring Thru Line 1 (on the PSTN Line tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes Display Name, User ID and any PSTN CID Number Prefix I have configured. I have adjusted PSTN Ring Thru Delay to 10 as I realise caller ID is not presented until the second ring in oz. I have also verified that caller ID is enabled on the line (with an analogue LCD handset). Has any aussie out there had success getting the SPA-3000 to pass caller ID to Asterisk? Not specifically the Sipura, but check that your circuit has caller id presentation enabled (it is off by default). Check the wiki about Australian Caller ID, I posted there the other day. The only settings I havn't played with yet is Caller ID Method (on the Regional tab). It is set to the default of Bellcore (N.Amer, China) which I beleive is correct. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Encrypted VOIP?
On Sat, 2005-02-05 at 12:33, Steven P. Donegan wrote: I have done that extensively (H.323 and SIP over IPSEC tunnels) I was more interested in the possibilities of 'native' support of some kind. But thank you very much for the response. Isn't there a fairly significant overhead with this, given the small size if the IAX2 datagrams? dean collins wrote: Just run point to point encryption over a vpn. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven P. Donegan Sent: Friday, February 04, 2005 8:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Encrypted VOIP? Is there any support in Asterisk for encryption of IAX and/or any other VOIP protocols? I haven't seen anything on this in the wiki or on the list. Just curious. ___ 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 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 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Incoming calls
On Thu, 2005-02-03 at 07:07, Martin Roy wrote:
OK I have 12 phone lines connected to 3 digium TDM04B cards on the same
server. I must do the following thing :
The first 10 lines will be use by one company and the 2 left by another
one. For outgoing calls it's quite easy I just create 2 different group
and let them dial on a different one. But for incoming calls how can I
setup asterisk to answer on the first 10 lines with one message and on
line 11 and 12 with another one?
If I put the s,1, Answer thing it will answer all 12 lines with the same
message...
I'm sure it's easy but I just don't know how to do it.
Do some string manipulation on the ${CHANNEL} to identify which channel
is ringing (look at the Cut and SetVar commands), then you can either
have an robot/recording answer appropriately, or use the A option in the
Dial command to announce to the callee which line is answered, or modify
the displayed caller id with SetCIDName and SetCIDNum.
I've done it, and it takes a bit of thinking about, but after this
amount of info the meter is running :)
HTH.
Thanks
Martin
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--
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LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.
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[Asterisk-Users] Soft phones that _actually_ work under Linux?
Surely there has to be one soft phone that works under Linux. I've tried: kphone - it sometimes complains about the need to release the sound device linphone - lowww iaxcomm - needs some strange widgets various others - either only supplied as binaries, or just plain don't work, or won't compile. Is there just one out there that is guaranteed to work with adequate performance with FC2 or FC3. I don't mind whether its SIP or IAX2 - I just need it to _work_. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Soft phones that _actually_ work under Linux?
On Wed, 2005-02-02 at 07:41, Michael Van Donselaar wrote: On Wed, 02 Feb 2005 07:12:54 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Surely there has to be one soft phone that works under Linux. I've tried: kphone - it sometimes complains about the need to release the sound device linphone - lowww iaxcomm - needs some strange widgets What does it ask for that you can't find? This is the version: -rw-r--r-- 1 lannet lannet 1392640 Feb 1 06:37 iaxcomm-lin-1.0rc1.tar and this is the error: $ ./iaxcomm Error wxWindows Fatal Error : Couldn't Initialize IAX Client . various others - either only supplied as binaries, or just plain don't work, or won't compile. Is there just one out there that is guaranteed to work with adequate performance with FC2 or FC3. I don't mind whether its SIP or IAX2 - I just need it to _work_. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zap channels in AU hanging up on STD pips
On Mon, 2005-01-31 at 16:51, jurgen wrote: Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: Big T [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=5 Sometimes the busydetect hack hits a false positive and disconnects during a conversation, so I'm thinking of upping the busycount, but aside from that, calls through this are quite reliable. Mine's pretty similar: context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes ;pulsedial = yes pulsedial = no rxgain = +15% txgain = +5% immediate = no busydetect = yes busycount = 5 callprogress = yes musiconhold = default usecallerid = yes callerid = asreceived ;usedistinctiveringdetection = yes useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 4 Best, ...jurgen On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] A neat hot seating mplementation
On Tue, 2005-02-01 at 08:12, Eric Bishop wrote:
Has anyone implemented hot seating in any neat way? This where
people can log in to any phone in the company and have their
calls/voicemail come to that particular handset.
I think this usually called follow me and is a variation on call
forward immediate. I have successfully done cfim where the user, at
their home station, can redirect their calls to another station, but I
haven't yet got around to fm, where the user is at a foreign station and
directs * to do a cfim to his new station. It shouldn't be difficult
and is really only an extension of cfim. I used DBPut, DBGet and DBDel
to implement it.
Here's the part of the dial plan that goes in the internal call
context:
'_1[1-7]Z' = 1. NoOp(Chan:${CHANNEL} Cntxt:${CONTEXT} Exten:${EXTEN}
Prio:${PRIORITY} Time:${TIMESTAMP} Clid:${CALLERID}) [pbx_config]
2. SetVar(CFIM=0) [pbx_config]
3. Macro(cfim|${EXTEN}) [pbx_config]
4. GotoIf($[${CFIM} = 0]?10:cfim|${CFIM}|1) [pbx_config]
10. Macro(voice|SIP/${EXTEN}|${EXTN_TIMEOUT}|t|${EXTEN})
[pbx_config]
and then this is the macro-cfim:
[ Context 'macro-cfim' created by 'pbx_config' ]
's' =1. NoOp(Chan:${CHANNEL} Cntxt:${CONTEXT} Exten:${EXTEN}
Prio:${PRIORITY} Time:${TIMESTAMP} Clid:${CALLERID} Arg1:${ARG1}) [pbx_config]
2. DBGet(CFIM=CFIM/${ARG1}) [pbx_config]
3. Playback(call-forwarding) [pbx_config]
103. NoOp() [pbx_config]
This is where I allow them to set and clear their cfim details, in
another internal context:
'*84' = 1. NoOp(Chan:${CHANNEL} Cntxt:${CONTEXT} Exten:${EXTEN}
Prio:${PRIORITY} Time:${TIMESTAMP} Clid:${CALLERID}) [pbx_config]
2. Wait(1)[pbx_config]
3. Read(CFIMNUM|custom/callforward-number)[pbx_config]
4. DBPut(CFIM/${CALLERIDNUM}=${CFIMNUM}) [pbx_config]
5. Wait(1)[pbx_config]
6. Playback(your) [pbx_config]
7. Playback(call-forward) [pbx_config]
8. Playback(has-been-set-to) [pbx_config]
9. SayDigits(${CFIMNUM}) [pbx_config]
10. Hangup() [pbx_config]
'*85' = 1. NoOp(Chan:${CHANNEL} Cntxt:${CONTEXT} Exten:${EXTEN}
Prio:${PRIORITY} Time:${TIMESTAMP} Clid:${CALLERID}) [pbx_config]
2. DBDel(CFIM/${CALLERIDNUM}) [pbx_config]
3. Playback(call-fwd-cancelled) [pbx_config]
4. Hangup() [pbx_config]
HTH, but usual disclaimers apply.
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--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.
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Re: [Asterisk-Users] Zap channels in AU hanging up on STD pips
On Tue, 2005-02-01 at 12:29, Stuart Elvish wrote: Dear Howard, Which version of Asterisk are you running? ext*CLI show version Asterisk CVS-HEAD-12/20/04-15:18:30 built by [EMAIL PROTECTED] on a i686 running Linux On the earlier versions we had problems with the call progress detect disconnecting calls (not specifically related to STD pips but it may be of help), however with the newer version of Asterisk we don't seem to encounter this problem as they have included the tone definitions for Australia. Kind Regards Stuart On Monday, Jan 31, 2005, at 13:34 Australia/Perth, Howard Lowndes wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 !DSPAM:41fdc41d213711706326924! ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zap channels in AU hanging up on STD pips
On Tue, 2005-02-01 at 14:27, Stuart Elvish wrote: Dear Howard, That is pretty much the latest version. In zapata.conf where you have callprogress=yes we have progzone=au. Ah ha, now that is a good point of which I was not aware. Many tks. We also have default and load zones set to au in zaptel.conf. Yes I have that set and similar in indications.conf. This should tell asterisk to look for Australian tones rather than the US ones which I assume it does by default. Hope this helps. Kind Regards Stuart On Tuesday, Feb 1, 2005, at 10:46 Australia/Perth, Howard Lowndes wrote: On Tue, 2005-02-01 at 12:29, Stuart Elvish wrote: Dear Howard, Which version of Asterisk are you running? ext*CLI show version Asterisk CVS-HEAD-12/20/04-15:18:30 built by [EMAIL PROTECTED] on a i686 running Linux On the earlier versions we had problems with the call progress detect disconnecting calls (not specifically related to STD pips but it may be of help), however with the newer version of Asterisk we don't seem to encounter this problem as they have included the tone definitions for Australia. Kind Regards Stuart On Monday, Jan 31, 2005, at 13:34 Australia/Perth, Howard Lowndes wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 !DSPAM:41feee08352724861317368! Stuart Elvish Business Development Manager TNet.com.au - Becoming Australia's Favourite Internet SERVICE Provider Mobile Telephone 0433 133 601 (+61 433 133 601) Email Address [EMAIL PROTECTED] Direct Telephone 08 9221 7874 (+61 8 9221 7874) Office Telephone 1300 661 NET (1300 661 638) Direct Facsimile 0433 133 598 (+61 433 133 598) Office Facsimile 08 9221 3864 (+61 8 9221 3864) This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zap channels in AU hanging up on STD pips
Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Caller ID in AU
On Fri, 2005-01-28 at 19:02, Simon Brown wrote: Insert a Wait(2) before Answer OK, I'll try that. I have also done the suggested mod to the chan_zap.c module to make the default rings 2. Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes Sent: Friday, 28 January 2005 17:30 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Caller ID in AU Is anyone in AU successfully getting Caller ID from the analogue PSTN service? If so, what settings? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Caller ID in AU
On Fri, 2005-01-28 at 19:21, PHP Mechanic wrote: Is anyone in AU successfully getting Caller ID from the analogue PSTN service? If so, what settings? -- Howard. http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID Done that one already. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Festival Jittery (bad udp checksum)
On Sat, 2005-01-29 at 05:50, Manjit Riat wrote: Just installed festival from source and the voice is very jittery and I get this a lot in the asterisk CLI (at least once on every call) NOTICE[3236]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum I get that also, but the stutters seem to have gone now that I have upgraded the CPU. One problem I do get is that it drops out when reading a long script, typically after 15-20 seconds, and then goes on the the next line in the dialplan. Maybe the packets are malformed so I get the jittery sound. __ ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Q: Can I over-ride the value of ${CALLERIDNAME} ?
On Fri, 2005-01-28 at 17:12, [EMAIL PROTECTED] wrote:
Folks,
I'd like to change the value of ${CALLERIDNAME} for incoming PSTN
calls from certain numbers, but haven't found a way that works. The goal is
to provide more informative names on my phones' caller ID displays--e.g., I
would prefer to display ROB CELL instead of CELLULAR CALL when I call
home from my cell phone.
This is what I tried in the context which handles inbound PSTN
calls:
...
exten = s/3125882300,1,SetVar(CALLERIDNAME=ROB CELL)
SetCIDName(ROB CELL)
exten = s/3125882300,2,Goto(100,1)
exten = 100,1,Macro(exten_vm,Zap/1)
...
The exten_vm macro context handles calls to extensions equipped with voice
mail.
When I make a call from my cell phone, the telephone caller ID unit
shows CELLULAR CALL instead of ROB CELL.
Does anyone have any ideas?
Cheers,
Rob
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--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.
___
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[Asterisk-Users] Caller ID in AU
Is anyone in AU successfully getting Caller ID from the analogue PSTN service? If so, what settings? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] TFTP Server Facing the Internet
On Thu, 2005-01-27 at 03:34, Michael Welter wrote: Since we're chatting about tftp servers... Let's say I have a new customer with Cisco 79xx phones, and he desires to SIP register on my Asterisk system. I would have to provide the SIPmac.cnf and SIPDefault.cnf files on my tftp server for his phones. These files would be world readable, which I don't want. Is the solution to put the tftp server behind the firewall and port redirect based on the customer's IP, or is there a better way of restricting access? TFTP on an open server is a definite no-no. Port redirection is better _if_ you have a static IP - but what are you going to do about dynamic IPs. Thanks, Mike ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Festival as background
Is it possible to run the Festival command in the same manner as the Background command so that it can be interrupted by caller key presses? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Dial command announcement
The Dial command can be made to make an announcement to the called party before channel is bridged. Is it possible to make that announcement a Festival command in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Festival
On Mon, 2005-01-24 at 14:45, Gary wrote: On Mon, 24 Jan 2005 14:57:06 +1300, Matt Riddell wrote: Howard Lowndes wrote: Is it possible to get the Festival command to read the text from a system file rather than having it input as a text string? Is this a case of having to use AGI, or is there a simpler way? Most people would use AGI for that (combined with the text2wave or whatever program). In fact there may even be an example on the wiki. I might also add that if you look in the wiki for cepstral as well some good examples. And cepstral voices sound much nicer than festival :-) Never heard of it. Tks for the lead. . ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zapata in Australia
On Mon, 2005-01-24 at 21:37, Emanuele Venditti wrote: Does anybody what the regional settings are to use an x100p (clone) card with Asterisk in Australia? I got mine installed and recognised by * but I get no sound and terrible hangup detection. Basically after each test call to the landine number (plugged into the x100p card) I need to unplug the cord and plug it back in to get a normal dialtone. When * answers the call (or diverts it to any internal IP phone) there is absolutely no sound. This works for me in AU. In /etc/zaptel.conf: fxsks=1 loadzone = au defaultzone=au In /etc/asterisk/zapata.conf: [channels] context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes pulsedial = yes rxgain = +15% txgain = +5% immediate = no busydetect = yes busycount = 3 callprogress = yes musiconhold = default usecallerid = yes callerid = asreceived useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 1 Note that I do not get callerid but I do get fax. many thanks manny __ ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zapata in Australia
On Tue, 2005-01-25 at 03:23, Andrew Yager wrote: As a general rule, the X100P should not be used in Australia as it is set to an incorrect impedence and can't be changed. The TDM series of cards with FXO/FXS modules can be set to work in AU. ... You should also be aware that the PSTN connect cards do not have Austel approval as yet, and so they shouldn't be connected the the public phone network. Another example of a situation where the sale and use of an article in Australia by an Australian business is legal, but the use of the article in Australia can be illegal. How do you spell telco cartel? Andrew On 25/01/2005, at 2:25 AM, Howard Lowndes wrote: On Mon, 2005-01-24 at 21:37, Emanuele Venditti wrote: Does anybody what the regional settings are to use an x100p (clone) card with Asterisk in Australia? I got mine installed and recognised by * but I get no sound and terrible hangup detection. Basically after each test call to the landine number (plugged into the x100p card) I need to unplug the cord and plug it back in to get a normal dialtone. When * answers the call (or diverts it to any internal IP phone) there is absolutely no sound. This works for me in AU. In /etc/zaptel.conf: fxsks=1 loadzone = au defaultzone=au In /etc/asterisk/zapata.conf: [channels] context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes pulsedial = yes rxgain = +15% txgain = +5% immediate = no busydetect = yes busycount = 3 callprogress = yes musiconhold = default usecallerid = yes callerid = asreceived useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 1 Note that I do not get callerid but I do get fax. many thanks manny __ ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] TDM400 in aging Dell Optiplex
On Tue, 2005-01-25 at 11:13, Ronan Mullally wrote: I've got an old Dell Optiplex (Pentium-II, 1998 Vintage) which is successfully running an X100P card. I'm hoping to upgrade to a TDM400. Has anybody tried running these cards in old Optiplex machines? I'm not particularly worried about horsepower - more about the motherboard having a PCI bus that's able to power up the card... Make sure that your PCI bus is 2.2 spec (issued Jan '99). I had this problem the other day with an X101P working OK and a TDM400P not being found by the BIOS. -Ronan ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Festival
Is it possible to get the Festival command to read the text from a system file rather than having it input as a text string? I suppose I could put the text string into an Asterisk variable and reference that in the Festival command, but then, how do I get the contents of the file into the Asterisk variable? Is this a case of having to use AGI, or is there a simpler way? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zap randomly hanging up
On Sat, 2005-01-22 at 02:47, C F wrote: Any T extensions set? Yes there are, but it not going down that path because they all do things - like voicemail. Maybe autofallthrough=yes and absolutetimeout Where would the first be set, and the second is not set anywhere. On Fri, 21 Jan 2005 17:02:44 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: I have a zap line on a X101P which will occasionally just hang up the call for no apparent reason. Is there any good way of trying to diagnose what might be causing this? Monitoring the asterisk output in verbose mode does not provide any indications. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Ring an incoming call in multiple extensions
On Fri, 2005-01-21 at 14:32, [EMAIL PROTECTED] wrote: Hi asterisk users! Heres my issue, Ive deleted the s extension cause I dont want any action to be taken on incoming calls as my pbx is for home use, but I would like to ring all my VoIP extensions at the same time the PSTN line rings and to be able to pick up the call in any extension, honestly I dont know if this is possible, some ideas ??? You still need your s exten, but when you do the Dial app you just do it to multiple extensions exten = s,n,Dial(SIP/111SIP/122SIP/133) They all ring and the first one that answers gets the call. Thanks in advance! __ ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zap randomly hanging up
I have a zap line on a X101P which will occasionally just hang up the call for no apparent reason. Is there any good way of trying to diagnose what might be causing this? Monitoring the asterisk output in verbose mode does not provide any indications. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] TDM400P card PCI problems
I've just replaced a X101P card with a brand new TDM400P card (specifically TDM421B). I do have the molex plug attached. kudzu removed the config for the X101P OK, but didn't find the TDM400P lspci does not show the card ?? Bung card ?? How susceptible are these cards to XRays, as it has been thru AU customs and might have been thru the scanner. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Directory() Command
On Tue, 2005-01-18 at 07:44, kurt x wrote: I am trying to use the Directory() but am having difficulty using it. According to Wiki page that I found you need to pass it your voicemail.conf context. My vm-context is [local]. So when I setup the cmd (Directory(local)) I can search on the three letters of the last name find that user. But once I press one to except the name and dial the extension I get the following message form the * CLI. Jan 17 15:22:07 WARNING[-1285669968]: app_directory.c:182 play_mailbox_owner: Can't find extension '' in context 'local'. Did you pass the wrong context to Directory? Reading the above error message I see that I need to pass it my outbound context. So I setup the command to look as follows: Directory(local outbound). Directory (vm-context | dial-context) ^ required I reload * and try again but this time it does not even pick up the name I search for. I used the same name in the first example. Any ideas on where I want wrong would be greatly appreciated. Kurt ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Wait(n) -v- Background(silence/n) ?
Will Wait(n) still listen for DTMF input from the caller after there has been a Background(some-message) prompt, or do I need to use Background(silence/n) to still listen for DTMF? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Wait(n) -v- Background(silence/n) ?
On Tue, 2005-01-18 at 13:18, Eric Wieling wrote: Howard Lowndes wrote: Will Wait(n) still listen for DTMF input from the caller after there has been a Background(some-message) prompt, or do I need to use Background(silence/n) to still listen for DTMF? The WaitExten and Read applications won't work for you? Duh! Ta! ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Is it the 15th or the 16th :)
Have a close listen to digits/h-15 and digits/h-16. To my ears the latter could be mistaken for the former ... or perhaps I am more deaf than I think. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Softphone for Linux recommendation
Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Softphone for Linux recommendation
On Sat, 2005-01-15 at 07:09, Adam Fineberg wrote: Howard Lowndes wrote: Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. How about iaxcomm? http://iaxclient.sourceforge.net/iaxcomm/ I should have added SIP reqd. I assume this only does IAX2 but I will look at it. I have almost got sflphone compiled only I have hit a missing file in one of the library compiles along the way. Adam ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] iaxComm 0.99pre11 binaries posted to Sourceforge
On Sat, 2005-01-15 at 12:27, Michael Van Donselaar wrote: iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol. It is distributed as part of Steve Kann's iaxclient library. I've just posted new Windows, Linux and Mac OSX binaries to sourceforge. The Windows binary was compiled on WinXP. The Linux binary was compiled on RedHat 9. ...and when I try to run this on FC2 it complains: # ./iaxcomm Error wxWindows Fatal Error : Couldn't Initialize IAX Client . WTF is wxWindows? The OSX binary was compiled by Andreas Wrede on 10.3 and was tested on 10.4 (Tiger) beta. These builds are from iaxclient CVS of 8 JAN 2005. http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-0.99pre11.zip http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-mac-0.99pre11.zip http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-lin-0.99pre11.tar -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Remote Voicemail Retrieval...
On Sat, 2005-01-15 at 15:03, Philippe Daoust wrote: Hello list, I want to listen to voicemails on my * box from a phone that is not local to my pbx. I.e., from my cellphone or my PSTN work line etc. I'm aware that I can forward VM to email or use a web interface but that is not always practical. Other than doing an IVR type arrangement or a phone number dedicated to VM access is there a way to do this? On my old POTS line I used to be able to call my line and simply punch * during unavailable message playback to go to the equivalent of voicemailmain(). Is there a way to do this in *? Set up voicemailmain in an extension that is part of the context used by the dial in line and use a Background message so that you can capture the DTMF for the extension. Thanks! ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Echo Training - how long
I have echo training set on in my zapata.conf file for a X101P card: echocancel = yes echocancelwhenbridged = yes echotraining = yes Now, I know that echo cancellation is a black art, but I am finding that at the beginning of a call bridged between a SIP channel and a Zap channel the voice quality is poor to abysmal for the first few seconds, but as the call progresses, esp after about 30 seconds, the call quality becomes very acceptable. Should echo training take that long? Is it, in fact, echo training or some thing else? Has any one got any guidance on ET other than what is in the wiki, which I find to be very hard to follow? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Problem patching asterisk CVS with SpanDSP
On Fri, 2005-01-14 at 06:20, Keith LeClaire Jr wrote: I'm trying to patch the current asterisk CVS with spandsp-0.0.1k.tar.gz. Everything compiles fine but when I go to patch the asterisk/apps/Makefile it fails: asterisk:/usr/src/spandsp2# patch Makefile.patch can't find file to patch at input line 3 Perhaps you should have used the -p or --strip option? The text leading up to this was: -- |--- Makefile.orig 2004-03-16 22:36:31.0 +0800 |+++ Makefile 2004-03-16 22:14:09.0 +0800 -- File to patch: /usr/src/asterisk/apps/Makefile patching file /usr/src/asterisk/apps/Makefile Hunk #1 FAILED at 35. Hunk #2 FAILED at 68. 2 out of 2 hunks FAILED -- saving rejects to file /usr/src/asterisk/apps/Makefile.rej Have a look in this file and you will see the lines that didn't make it to Makefile - there are ususally only a couple. asterisk:/usr/src/spandsp2#akefile in asterisk/apps I get: Am I using the wrong versions together? I've also patched this asterisk source for ast_data mysql support. -Keith ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Voice Mail Notification
On Fri, 2005-01-14 at 14:14, Mike Boger Jr wrote: Hi, Here's the deal. When someone leaves me a voicemail message I want Asterisk to call me on my cellphone by dialing my cellphone number and tell me I have a message. Is this possible? Can anyone cite examples? Most commercial voicemail systems produced in the last 10 years can do this. Any help would be much appreciated. There used to be a nice little program called sms-client that did this, basically by using the email notification function out of voicemail. Have a google for it. But isn't there an sms function in voicemail advanced functions anyway. Regards, Mike __ ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Updated kphone 4.0.5, asterisk v1.0.3
On Fri, 2005-01-14 at 15:09, Andrew McRory wrote: I have uploaded kphone and asterisk CVS stable. These packages are built for Fedora Core 1 and this asterisk release should fix the non-root permissions problem I worte about... ftp://ftp.linuxsys.com/pub/releases/FC1/ I have just upgraded to kphone 4.0.5 from 4.0.3 - tks for it I used to get a message occasionally about closing other programs that are using the sound card. I did get get it once with this update. It appears to be related to kphone sending SIP SUBSCRIBE packets. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Updated kphone 4.0.5, asterisk v1.0.3
On Fri, 2005-01-14 at 15:09, Andrew McRory wrote: I have uploaded kphone and asterisk CVS stable. These packages are built for Fedora Core 1 and this asterisk release should fix the non-root permissions problem I worte about... ftp://ftp.linuxsys.com/pub/releases/FC1/ OK, there are a number of issues I have detected. The error message about closing other applications using the sound card is definitly repated to the SIP SUBSCRIBE packets. When I run it from an xterm, on hangup it seg faults. This does not happen when I run it from a KDE panel button. The DTMF tones generated from the on-screen keypad appear not to be recognised by *. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] New SIP Phone Config
On Thu, 2005-01-13 at 10:40, John Dunham wrote: Just checking if anyone has experence with Integrated Networks IN1002 phone. You might like to try aredfox.com and see if there is anything there that might suit. I have HOP1002 phones and I am using the 1002 as a clue here. We just got 100 of them in and no manual or passowrd to program the phone. Also need some direction on the * sip.conf if anyone has experence with these phones. Thanks, John Dunham ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Setting channel display in SIP
On Thu, 2005-01-13 at 11:11, Adam Goryachev wrote: On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote: I have a situation where I need to know which Zap channel an incoming call is on, so that the call can be answered appropriately when a SIP phone displays the channel. These Zap calls are coming in over PSTN and don't have caller ID. As far as I can make out my SIP phones (WuChuan HOP-1002) display the user part from the SIP From: header as the second line on the display. If the call comes from another SIP phone then this shows as the phone's number, but when the call comes in over the Zap channels then it gets generated as asterisk. AFAIK, this is the default callerid asterisk uses when it doesn't receive callerid. Try adding setcallerid in your dialplan, I tried setcidname in the dialplan without success, so I will try this suggestion. or callerid in your zapata.conf for each channel. Tried that - no dice. Regards, Adam ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Setting channel display in SIP
On Thu, 2005-01-13 at 12:38, Adam Goryachev wrote:
On Thu, 2005-01-13 at 11:24 +1100, Howard Lowndes wrote:
On Thu, 2005-01-13 at 11:11, Adam Goryachev wrote:
On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote:
I have a situation where I need to know which Zap channel an incoming
call is on, so that the call can be answered appropriately when a SIP
phone displays the channel. These Zap calls are coming in over PSTN and
don't have caller ID.
As far as I can make out my SIP phones (WuChuan HOP-1002) display the
user part from the SIP From: header as the second line on the
display. If the call comes from another SIP phone then this shows as
the phone's number, but when the call comes in over the Zap channels
then it gets generated as asterisk.
AFAIK, this is the default callerid asterisk uses when it doesn't
receive callerid.
Try adding setcallerid in your dialplan,
I tried setcidname in the dialplan without success, so I will try this
suggestion.
Play with combinations of setcallerid and setcidnum and setcidname ...
see the wiki to correctly format your examples.
I have actually got a bit more cunning that this by using sipgetheader()
and sipaddheader().
The default user name is asterisk, hard coded in chan_sip.c, so what I
did was to use sipgetheader() to get the From: header, then I cut() it
at the : character and the @ character and checked the string
between these two characters. If the string was asterisk then I did
sipaddheader(From: ${PIECE_BEFORE}:[EMAIL PROTECTED]).
OK, so it adds a second From: header, but as it gets added after the
original it doesn't seem to matter because it works and
replacement-string is what gets displayed on the phone, which is what
I want. I also don't see that the tag= in the header makes any
difference either.
Can anyone see any probs I am likely to encounter using this?
or callerid in your zapata.conf
for each channel.
Tried that - no dice.
Send your zapata.conf file so we can see what you tried. AFAICT,
asterisk is sometimes picky with the formatting of the callerid info.
Regards,
Adam
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.
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Re: [Asterisk-Users] rxfax troubles..
On Wed, 2005-01-12 at 11:01, Matthew Boehm wrote: what is g723? ive never seen that before... It's a codec. and it look like you have some form of codec translation problem. -- Executing Answer(Zap/1-1, ) in new stack -- Accepting call from '2819870065' to '2815692780' on channel 0/1, span 1 -- Executing Wait(Zap/1-1, 1) in new stack -- Executing SetVar(Zap/1-1, FAXFILE=/var/spool/asterisk/fax/1105486770.492.tif) in new stack -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/fax/1105486770.492.tif) in new stack asterisk*CLI show channels Channel (Context Extension Pri ) State Appl. Data Zap/1-1 (fax-handle f 4 ) Up RxFAX /var/spool/asterisk/fax/1105486770.492.tif 1 active channel(s) Jan 11 17:40:24 NOTICE[26308]: channel.c:1691 ast_set_write_format: Unable to find a path from g723 to unknown Jan 11 17:40:24 WARNING[26308]: app_rxfax.c:302 rxfax_exec: Unable to restore write format on 'Zap/1-1' == Spawn extension (fax-handle, f, 4) exited non-zero on 'Zap/1-1' From debug: DEBUG[26308]: == DEBUG[26308]: Pages transferred: 1 DEBUG[26308]: Image size: 1728 x 112 DEBUG[26308]: Image resolution 7700 x 3850 DEBUG[26308]: Transfer Rate: 9600 DEBUG[26308]: Bad rows 24 DEBUG[26308]: Longest bad row run 14 DEBUG[26308]: Compression type 2 DEBUG[26308]: Image size (bytes) 0 DEBUG[26308]: == DEBUG[26308]: == DEBUG[26308]: Fax successfully received. DEBUG[26308]: Remote station id: DEBUG[26308]: Local station id: DEBUG[26308]: Pages transferred: 1 DEBUG[26308]: Image resolution: 7700 x 3850 DEBUG[26308]: Transfer Rate: 9600 DEBUG[26308]: == Additional Information Faxing the same page from a Sharp UX-510, a Brother intelliFax 1150, and a HP OfficeJet. All 3 fax machines reported OK fax sent. But the resulting TIFF always says something about Invalid Compression Type Using libtiff 3.7.0 and spandsp-0.0.2 bug report and the recieved page are here: http://www.opencall.org/mantis/bug_view_page.php?bug_id=019 any ideas? -matthew ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Installing * on fedora 3
On Wed, 2005-01-12 at 12:40, Matt Riddell wrote: Ferguson, Michael wrote: G'Day All, rpm -q kernel-source returns Package kernel-source is not installed Where can I find it and install it. Asterisk evidently needs it for a successful install. You can do: yum install kernel-source (although I thought you didn't need it in 2.6) Purely guessing, but isn't it yum install kernel-headers -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] very loud scratchy noise!
On Tue, 2005-01-11 at 00:29, Rich Adamson wrote: I am new to asterisk but learn a lot about it to this mailing list and wiki currently i am facing problem about sip phone i have PA 1688 chipset ip-phone and i have iptel.org sip account i registered locally and through iptel.org comfortably my problem is that when i called from my sip phone to analog or any number after connection my sip phone generates very load scartchy noise , i tried several settings of DTMF but all in vein i enabled/disabled DTMF settings but not workin any info any hints any suggestions please .. You didn't mention what type of analog interface you're using, but if it is an x100p or tdm digium card, you are probably seeing the same problem that lots of us have seen. If you are using either of those two cards, stop asterisk, reload the drivers, and restart asterisk. Or, simply reboot the complete system. I'm getting it with KPhone to a SIP phone even, also to a mobile (cell) phone through a X101P card. I'm putting it down to the sound card which I might get around to changing if I want to persist with KPhone. If you're not using one of those two cards, then help us understand your system. What OS? What * version? What analog interface? etc. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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
