Re: [Asterisk-Users] AGI say number but in french
it was the second one i needed - thank you. I only needed the numbers
in french on one b number and make it that the number could be
dialled from any extension (which is why option a was unsuitable.
thanks again.
On 28/06/05, Arvanitis Kostas [EMAIL PROTECTED] wrote:
On Monday 27 June 2005 23:04, David John Walsh wrote:
Hello,
does anyone know how to get the say number (say.c) agi application
to work in french [assuming that I have the French voice files]
I have looked in the code and about a 1/3 of the way thru there is :
} else if (!strcasecmp(language, fr) ) { /* French
syntax */
and then further on there is logic for french numbers.
does anyone know the syntax as looking on the code / google / wiki
gives me no ideas.
Have you tried setting the channel language to fr (language=fr in the
device configuration, or SetLanguage(fr) in the extensions.conf file)?
This seems to be all that is needed.
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[Asterisk-Users] AGI say number but in french
Hello,
does anyone know how to get the say number (say.c) agi application
to work in french [assuming that I have the French voice files]
I have looked in the code and about a 1/3 of the way thru there is :
/*--- ast_say_number_full: call language-specific functions */
/* Called from AGI */
int ast_say_number_full(struct ast_channel *chan, int num, char *ints,
char *language, char *options, int audiofd, int ctrlfd)
{
if (!strcasecmp(language,en) ) { /* English syntax */
return(ast_say_number_full_en(chan, num, ints, language,
audiofd, ctrlfd));
} else if (!strcasecmp(language, da) ) { /* Danish syntax */
return(ast_say_number_full_da(chan, num, ints, language,
options, audiofd, ctrlfd));
} else if (!strcasecmp(language, de) ) { /* German syntax */
return(ast_say_number_full_de(chan, num, ints, language,
options, audiofd, ctrlfd));
} else if (!strcasecmp(language, es) ||
!strcasecmp(language, mx)) {/* Spanish syntax */
return(ast_say_number_full_es(chan, num, ints, language,
options, audiofd, ctrlfd));
} else if (!strcasecmp(language, fr) ) { /* French syntax */
and then further on there is logic for french numbers.
does anyone know the syntax as looking on the code / google / wiki
gives me no ideas.
Thanks
David
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Re: [Asterisk-Users] PBXfreeware.org Open for business! / JavaScript module for Asterisk Unveiled!
Sorry for the newbie style posting, but i normally install my applications from an RPM or at least a make install etc How does one go from app_valetparking.c for example to a application one can use within asterisk? Thank you for your assistance. Anthony has uploaded a few of his own popular open source modules for Asterisk namely res_perl, res_sqlite and app_valetparking as well as a brand new module just ___ 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: app_valetparking.c for * STABLE (1.0.X)
Sorry for asking a seemingly new person question, but how do you get from the code below to a working app for asterisk? is it just a case of putting this into the source directory and doing make install or gcc somthing.c? then how do you get asterisk to understand that it is avaible to it? Thanks David On 20/06/05, Paul Zimm [EMAIL PROTECTED] wrote: Oops, I sent the wrong one. Here's one I modified to work with 1.0.X Try again Nope ! This is the one that tries to include PRE 1.0.X header file parking.h. It cannot compile on * 1.0.X (I have tried also to include features.h instead of parking.h (as far as I know features.h is successor to parking.h), but still without results). Thanks anyway. Nenad Try this /* * Asterisk -- A telephony toolkit for Linux. * * Routines implementing call valetparking ___ 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] Bill seconds
Americo 60+60 isn't a VoIP term directly but a generic one within the telephony industry if it were 60+30 it would mean the following You are billed for 60 second as soon as the call is answered, even if you only stay on the line for 7 seconds The +30 then referes to the onward billing cycle, so in this case you are billed in blocks of 30 seconds (ie if you call is 1 min 15 seconds you are billed for 1 min 30) You said that you are billed for a whole second minuite if you go over by even 1 second, so that would be a +60, and since its always a bigger or equal number first we are guessing that you are in 60+60 rate plan I think its more common in your part of the world for your carriers to bill 30+6. One in the replys suggested a very favorable rate of 6+6 The important thing to rember here is that you can't gaurentee enough return if you do a billing rate that is better than that of your carriers - it sounds to me like your offering your service on a 1+1 (ie true per second billing) rate - very honarable, but your carrier needs to offer the same. I hope that helps On 17/06/05, Americo Sanchez C. [EMAIL PROTECTED] wrote: From: Leon Sun [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Bill seconds Date: Thu, 16 Jun 2005 10:56:23 -0700 The easiest way is to change another vendor asap. Do you mean to change to another telecom? In my country there is a telephone monopoly :( Telefonica del Peru) It is ridiculous that your carrier still uses 60+60 now(30+6 is an asset). 2 seconds doesn't matter and billing unit does. Sorry I am not an expert in VoIP, What is the meaning of 60+60? Leon Sun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: June 15, 2005 10:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bill seconds I've done a little thinking on this one If you are using ASTCC, it would be fairly straightforward to edit it and have it make a 2 second adjustment. If your using another solution it probably would be fairly easy also... Darren Wiebe [EMAIL PROTECTED] Americo Sanchez C. wrote: Hi all, We've installed Asterisk on a rural development project and we're testing a prepaid phone service. As far as now we're having terrific service results but there's a problem with the calls billing at our local telecom. For instance, a farmer buys a 1 dollar phone card and use it to dial a USA number, the call should lasts for 60 seconds. Asterisk is doing a great job finishing the call exactly at 60 seconds. The problem is that the telecom company billing system adds a two second delay for each call, so the bill is not for 1 but 2 minutes (they round fractions up). We're loosing money and the local telecom doesn't seem to have a solution for this matter. Have you experienced something similar? Do you have any idea of how can we solve this? Is it possible to configure Asterisk so that the system thinks that a minute has 58 seconds instead of 60? _ MSN Amor: busca tu naranja http://latam.msn.com/amor/ ___ 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 _ Consigue aqu las mejores y mas recientes ofertas de trabajo en Amrica Latina y USA: http://latam.msn.com/empleos/ ___ 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
Re: [Asterisk-Users] Dell PowerEdge + TDM
Here we have PowerEdge 2850's doing the donky work with a Wildcard TE405P in each. I have seen no operational issues at all with the system or the cards. We are running CentOS 3 as the operating system and the stable version of asterisk The only niggle is that when the cards are modprobed on start up they sometimes 2 in a 100 give an NMI message, causing an error code on the servers little window, its not affected the stability at all, and its on my list of things to do to find out what causes it! The systems generally have around 400 - 500 SIP extensions comming off the back, running around a dual xeon 3Ghz and 3Gb of ram (no transcoding all G711.u) - we are very happy! David On 17/06/05, David Hajek [EMAIL PROTECTED] wrote: Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David ___ 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
Re: [Asterisk-Users] 2nd Dialtone after answer
it sounds like you need to investigate the application called DISA.. it might be what you are looking for if not, what dial tone is your client expecting? (internal, external - other??) David On 17/06/05, Oswaldo Arratia [EMAIL PROTECTED] wrote: Hi I am trying to achive this for a specific need of a customer. He has a DID pointed to an Asterisk server, I need to provide him dialtone when the calls hits the server. How can I achieve this? Let's say something like this: Exten = s,1,Answer Exten = s,2, Provide Dial tone Exten = s,3, Dial the number the person will enter after receiving the dial tone Exten = s,4,Hangup Any ideas? Thanks very much Oswaldo ___ 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
Re: [Asterisk-Users] Bill seconds
Another way I have seen this done is to sell units, not pounds and pence credit eg a 2 calling card has 160 units (ratio of 80 units to the pound). If you were to charge 8p per min you make that 8 units per min. This gives you a 20% increase which might help if your on per second billing to your upstream carrier. otherwise you need to make changes to your rating engine with a /60*58 to re-rate all calls back to a second ( /60) and move the minuite charge to be a 58 second minuit (*58) how that is achived needs you to give specific information on which calling card platform you are using. You may have a problem in defining the rates as per minuite if they are not a widely understood minuite legally - it depends on the laws of your country (in the UK the Trades Descriptions Act would apply and you'd be hit hard) David On 16/06/05, Race Vanderdecken [EMAIL PROTECTED] wrote: Your customers are not going to like this. You have to change the way you bill for calls. For $1 your customer gets 60 seconds worth of phone time. However you have to also charge, like the Bells used to, for setup and teardown time. Remember the operator used to say Deposit $1.85 for the first three minutes and then it would be 30 cents per minute after that. Buy a phone card from a competitor and look at the fine print on the card. You charge buy seconds they are connected to your system, not for the time they are actually talking to the remote party. Example: To set up the call you charge 10 seconds, and to stop the call you charge 5 seconds. So the customer only gets 45 seconds of call time. You get a 15 second cushion. Does not seem fair does it. But if they buy an hour 3600 seconds worth of calls the missing 15 seconds won't be noticed. You can go further. Say they buy a 3600 second card. When they call to check their time the first time on the card you tell them they have 60 minutes, but you charge them 30 seconds for asking. Set up the code so that every time they call you have too fields to track call time. The time they think they have and the time you know they have. You tell them they have 45 minutes, but the other field knows they only have 30 minutes. If they ask then your script says 45 minutes left but you cut them off when the use 30. Then you chip away each time the call. 10 seconds for making a call, and 5 seconds when they hang up. This way you are always in credit and can cut them off without loosing money. Some card vendors go even further. They sell 3600 seconds, but each time a call is made they whack a random percentage of the time. Worse yet their card system will randomly or systematically hang up on callers. This will cause the user to redial the call and get hit with connection charges that vary. Customers eventually figure out which cards do this type of chicanery and they stop buying them, but only if there is a competitor for the route they want to call. Such is the world of unregulated phone calls. Not pretty is it. Charging time for each call is part of the business. If you don't want to charge time to setup and teardown then you have to charge more per minute. Your customers get all the time the pay for down to the second, but you are going to have to charge more per minute or you will be in the boat you are in now. Race the tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Thursday, June 16, 2005 1:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bill seconds I've done a little thinking on this one If you are using ASTCC, it would be fairly straightforward to edit it and have it make a 2 second adjustment. If your using another solution it probably would be fairly easy also... Darren Wiebe [EMAIL PROTECTED] Americo Sanchez C. wrote: Hi all, We've installed Asterisk on a rural development project and we're testing a prepaid phone service. As far as now we're having terrific service results but there's a problem with the calls billing at our local telecom. For instance, a farmer buys a 1 dollar phone card and use it to dial a USA number, the call should lasts for 60 seconds. Asterisk is doing a great job finishing the call exactly at 60 seconds. The problem is that the telecom company billing system adds a two second delay for each call, so the bill is not for 1 but 2 minutes (they round fractions up). We're loosing money and the local telecom doesn't seem to have a solution for this matter. Have you experienced something similar? Do you have any idea of how can we solve this? Is it possible to configure Asterisk so that the system thinks that a minute has 58 seconds instead of 60? _ MSN Amor: busca tu naranja http://latam.msn.com/amor/
Re: [Asterisk-Users] AreskiCC Calling Problem
in one of the two defines configs (where you set the database up)
(sorry cant recall which one and im out of the office) there is a min
call value, its set by default around the 10 unit mark. if the cards
credit is below this it stops you going any further. I can only assume
this was to end the call quickly if there is no chance of it
completing and you user is dialing in on a 0800 or 0808 style number
where you as the operator pick up that part of the bill
That aside, if you change this value to 0 it take away that limit.
David
On 11/06/05, Junaid Uppal [EMAIL PROTECTED] wrote:
Hello There,
I *think* i've setuped the AreskiCC2 Calling Card system right , but
i've yet to make any calls out of it , i added a rate card , trunk
and defined some rates , generated some users , added 10 dollars in
them , okay , now i call any number , it asks me to enter my pin , i
do , it tells me i have ten $ , right after that it says sorry you
dont have enough funds for this call and hangs up. i see this in cli
help me out please guys , thanks a lot!!
regards
~junjun
--
CLI LOG START
--
areskicc2.php: 'agi_callerid' = '1001'
areskicc2.php: 'agi_calleridname' = 'Junaid Uppal'
areskicc2.php: 'agi_callingpres' = '0'
areskicc2.php: 'agi_callingani2' = '0'
areskicc2.php: 'agi_callington' = '0'
areskicc2.php: 'agi_callingtns' = '0'
areskicc2.php: 'agi_dnid' = '011905'
areskicc2.php: 'agi_rdnis' = 'unknown'
areskicc2.php: 'agi_context' = 'default'
areskicc2.php: 'agi_extension' = '011905'
areskicc2.php: 'agi_priority' = '3'
areskicc2.php: 'agi_enhanced' = '0.0'
areskicc2.php: 'agi_accountcode' = ''
areskicc2.php:
areskicc2.php: ANSWER
areskicc2.php: string(48) 1001 ; SIP/1001-d6fb ; 1118521907.13 ; ;
011905n
areskicc2.php: string(26) Requesting DTMF :: Len-10n
areskicc2.php: GET DATA prepaid-enter-pin-number 1 10
-- Playing 'prepaid-enter-pin-number' (language 'en')
areskicc2.php: string(21) RES DTMF : 5882431851n
areskicc2.php: string(25) CARDNUMBER :: 5882431851n
areskicc2.php: string(94) SELECT credit, tariff, activated, inuse,
simultaccess FROM cc_card WHERE username='5882431851'n
areskicc2.php: array(1) {n [0]=n array(5) {n[0]=n
string(2) 10n[1]=nstring(1) 1n[2]=nstring(1)
tn[3]=nstring(1) 0n[4]=nstring(1) 0n }n}n
areskicc2.php: STREAM FILE prepaid-you-have #
areskicc2.php: SAY NUMBER 10 X
-- Playing 'digits/10' (language 'en')
areskicc2.php: STREAM FILE prepaid-dollars #
areskicc2.php: string(60) UPDATE cc_card SET inuse=inuse+1 WHERE
username='5882431851'n
areskicc2.php: CHANNEL STATUS SIP/1001-d6fb
areskicc2.php: result is 6
areskicc2.php: string(20) [CHANNEL STATUS : 6]n
areskicc2.php: STREAM FILE prepaid-no-enough-credit-stop #
areskicc2.php: string(60) UPDATE cc_card SET inuse=inuse-1 WHERE
username='5882431851'n
areskicc2.php: STREAM FILE prepaid-final #
-- AGI Script areskicc2.php completed, returning 0
-- Executing Wait(SIP/1001-d6fb, 2) in new stack
-- Executing Hangup(SIP/1001-d6fb, ) in new stack
== Spawn extension (default, 011905, 5) exited non-zero on 'SIP/1001-d6fb'
-
CLI LOG ENDS
here's the /tmp/areskicc-errors.log
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[TRY :
callingcard_ivr_authenticate]
[11/06/2005
16:08:51]:[CallerID:1001]:[CN:5882431851]:[callingcard_acct_start_inuse]
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[Start: UPDATE
cc_card SET inuse=inuse+1 WHERE username='5882431851']
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[CHANNEL STATUS : 6]
[11/06/2005 16:08:53]:[CallerID:1001]:[CN:5882431851]:[Start: UPDATE
cc_card SET inuse=inuse-1 WHERE username='5882431851']
[11/06/2005 16:08:53]:[CallerID:1001]:[CN:5882431851]:[exit]
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Re: [Asterisk-Users] Do I need a ring capacitor to use TDM400P cards in UK
Angus a BT socket with a capacitor in is commonly refered to as a Master socket, and are very cheap even without wholesale. It gets its name from being the socket that BT installed into the house for the line, all other sockets in the house will be slave or secondary (ie no capacitor) (and its against the law to play with the one BT installed - but thats off topic!) I'm not going to name names of where you can buy them, but they extreamly widely availible across the UK. Its all from historic reasons from when BT started providing wall sockets, and the phrase BABT aproved was common, quite a quirky thing to the UK these days. If you want exact suppliers, email me directly. On 08/06/05, Henry Coleman [EMAIL PROTECTED] wrote: Hi Angus, If you connect the phone directly to the outside line will it ring ? The ring from the C.O. provides a 90volt AC (30cps) and is capable of ringing a standard phone ( a real two tone gong bell) My guess is that the TDM400 card does not supply enough current to actually do this. Most modern phones have an electronic ringer which requires a fraction of the power and will work fine. I don't quite understand the reference to a capacitor unless your phone is as old as I am in which case the phone has 2 x pairs of wires going to the phone plug. The first pair of wires are the voice pair and the second pair are connected to the ringer if this is the case your phone will work normally but simply doesn't ring. The fix is to connect two capacitors approx *0.15 uf 250vw *from each wire of the voice pair to each wire of the ring pair (you can do this inside the phone jack) This should not cost much (about a dollar) and can be found in any electronics component shop (try Maplin electronics) . Concidering the time and effort you might want to buy a new phone. 0.1uf Wall Jack TDM400 | ---||---yellow---0 0---green--0---0 0---red-0--- 0 | ---||---black-0 0.1uf Hope this helps ...Henry * * Angus Comber wrote: Hello I have played about with a TDM400 card and plugged in some standard analog phones. I am using the card in FXS mode - for analog extensions. I did notice that one of my phones did not ring and I wondered why. I later read in Paul Mahler's book VoIP Telephony with Asterisk that in his section on the TDM400 on page 127 he says In the UK, you may need an adapter that provides a ring capacitor, or the phone may not ring. Can anyone confirm this. Also what is one of those and where would I find a good supplier? I am in the trade so wholesale would be OK. Angus Comber ___ 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
Re: [Asterisk-Users] AAH 1.1 - CRM Setup
Breifly - yes in the users extension - define it as SIP/3001 (if the users extension is 3001) in the contacts part - define it as you would dial it eg 020 0001 01234 David On 07/06/05, Wiley Siler [EMAIL PROTECTED] wrote: Hello All, Has anyone successfully gotten the Click to Dial to work in SugarCRM in the latest AAH? I keep getting 'Invalid Channel' but I cannot figure out why. Thanks! Wiley ___ 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
Re: [Asterisk-Users] DELL 2800 : PCI Parity error
Frank Did you ever resolve this? If so what was the issue? On 03/05/05, list [EMAIL PROTECTED] wrote: Hi, I am struggling to get rid of a conflict on DELL 2800 : PCI Parity error (EB113 on the display) I am learning linux and asterisk as I go along, there might be obvious things I should know, but bear with me. From demsg below my 2 digium cards installed are listed (no config or connections done to digium cards yet), the conflict is with the TDM400P card, without that card, in any slot, no alarm. Zapata Telephony Interface Registered on major 196 Registered Tormenta2 PCI Controller version: 24 FALC version: TE110P: Setting up global serial parameters for E1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 Freshmaker version: 71 Freshmaker passed register test Uhhuh. NMI received. Dazed and confused, but trying to continue You probably have a hardware problem with your RAM chips Module 0: Installed -- AUTO FXS/DPO Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) wcte1xxp: Setting yellow alarm usb.c: registered new driver wcusb Wildcard USB FXS Interface driver registered TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) ramchip problem is false, without the card all ok, ramtests on machine as well. lsmod shows wcusb driver on zaptel, I dont need that, can I remove it? is that a problem or not? # lsmod Module Size Used byNot tainted usbserial 23964 0 (autoclean) (unused) lp 9156 0 (autoclean) parport38848 0 (autoclean) [lp] autofs416984 0 (autoclean) (unused) wcusb 19552 0 (unused) wctdm 41088 0 (unused) wcte11xp 22048 0 (unused) zaptel182080 4 [wcusb wctdm wcte11xp] e1000 77884 1 (autoclean) floppy 57552 0 (autoclean) sg 37388 0 (autoclean) microcode 6912 0 (autoclean) ide-cd 34016 0 (autoclean) cdrom 32896 0 (autoclean) [ide-cd] keybdev 2976 0 (unused) mousedev5688 1 hid22308 0 (unused) input 6176 0 [keybdev mousedev hid] ehci-hcd 20776 0 (unused) usb-uhci 26860 0 (unused) usbcore81152 1 [usbserial wcusb hid ehci-hcd usb-uhci] ext3 89960 6 jbd55060 6 [ext3] megaraid2 38344 7 diskdumplib 5228 0 [megaraid2] sd_mod 13904 14 scsi_mod 115112 2 [sg megaraid2 sd_mod] finally my interrupts, bit confusing to me, looks like I have dual processor, can see the NMI but what else can be found here? # cat /proc/interrupts CPU0 CPU1 0:32983953303167IO-APIC-edge timer 1: 3300 2876IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 8: 0 1IO-APIC-edge rtc 12: 236637 237965IO-APIC-edge PS/2 Mouse 14: 261779 262965IO-APIC-edge ide0 16: 0 0 IO-APIC-level usb-uhci 18: 0 0 IO-APIC-level usb-uhci 19: 0 0 IO-APIC-level usb-uhci 23: 0 24 IO-APIC-level ehci-hcd 29: 33133540 32846566 IO-APIC-level t1xxp 38: 72500 83317 IO-APIC-level megaraid 58: 32838989 33150525 IO-APIC-level wctdm 72: 222855 12 IO-APIC-level eth0 NMI: 1 0 LOC:66014626601460 ERR: 0 MIS: 0 any suggestions from someone experienced something similar? regards Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] SNOM 360 extension lights
Sorry Ross I must have missed your first postings, but what are you trying to achive? David On 03/06/05, Ross Kevlin [EMAIL PROTECTED] wrote: I contacted SNOM and they told me to change a couple of options but still no lights, here is what they told me Line page SIP tab: o Long SIP-Contact (RFC3840) to off o Support broken Registrar to on Advanced page: o Filter Packets from Registrar to off And please ask the Asterisk community for help, I'm sure they solved that issue 100%, and we are not knowing so much about Asterisk. Your snom support Team has anyone gotten a 360 to work with the lights? what options and modifications to .conf files did you have to make? here are the subscribe and notifies. it seems it terminates the subscription as soon as its created. I don't think its a proxy authentication problem because it eventually sends the proxy authentication information Using latest SUBSCRIBE request as basis request Sending to 192.168.2.230 : 2051 (non-NAT) Found peer '83' Transmitting (no NAT) to 192.168.2.230:2051: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n From: sip:[EMAIL PROTECTED];tag=z6kvtd67bu To: sip:[EMAIL PROTECTED];user=phone;tag=as6c1cb2a5 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: MVC 001 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=16747f76 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms sip1*CLI -- SIP read from 192.168.2.230:2051: SUBSCRIBE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n;rport From: sip:[EMAIL PROTECTED];tag=z6kvtd67bu To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:2051;line=kcx1qlml Event: dialog Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0 --- (12 headers 0 lines)--- Ignoring this SUBSCRIBE request Found peer '83' Transmitting (no NAT) to 192.168.2.230:2051: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n From: sip:[EMAIL PROTECTED];tag=z6kvtd67bu To: sip:[EMAIL PROTECTED];user=phone;tag=as6c1cb2a5 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: MVC 001 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=16747f76 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms sip1*CLI -- SIP read from 192.168.2.230:2051: SUBSCRIBE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x;rport From: sip:[EMAIL PROTECTED];tag=z6kvtd67bu To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:2051;line=kcx1qlml Event: dialog Accept: application/dialog-info+xml Proxy-Authorization: Digest username=83,realm=asterisk,nonce=16747f76,uri= sip:[EMAIL PROTECTED];user=phone,response=15d72104244317e2c0afa3499220e4ab,a lgorithm=md5 Expires: 3600 Content-Length: 0 --- (13 headers 0 lines)--- Using latest SUBSCRIBE request as basis request Sending to 192.168.2.230 : 2051 (non-NAT) Found peer '83' Looking for 117 in localusers-C2021-1 Transmitting (no NAT) to 192.168.2.230:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x From: sip:[EMAIL PROTECTED];tag=z6kvtd67bu To: sip:[EMAIL PROTECTED];user=phone;tag=as77c7b911 Call-ID: [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: MVC 001 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 3600 Contact: sip:[EMAIL PROTECTED];expires=3600 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 361 ms Reliably Transmitting (no NAT) to 192.168.2.230:2051: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport From: sip:[EMAIL PROTECTED];user=phone;tag=as77c7b911 To: sip:[EMAIL PROTECTED];tag=z6kvtd67bu Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: MVC 001 Event: dialog Content-Type: application/dialog-info+xml Content-Length: 203 ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=0 state=full entity=sip:[EMAIL PROTECTED] dialog id=117 stateterminated/state /dialog /dialog-info --- sip1*CLI -- SIP read from 192.168.2.230:2051: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=as77c7b911 To: sip:[EMAIL PROTECTED];tag=z6kvtd67bu Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Content-Length: 0
Re: [Asterisk-Users] Sipura 3000 dialing noise
Eric
A completly off topic response (and not even a response in that I'm
asking you a question - sorry)
you say that you have several 3000 devices and you show your dial string as :
Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Is the sipura1 section referencing a single sipura or the group of
several. The only reason that I ask if it is the latter - how are
you grouping them.
Thanks for you response if i figure the answer to your question I
will post back.
David
On 31/05/05, Eric Bishop [EMAIL PROTECTED] wrote:
Hi all,
We have several sipura 3000's working well for outbound calls, however
the issue we have is that when calls are sent to the Sipura with
Dial(SIP/${EXTEN:[EMAIL PROTECTED]) the Sipura does a SIP answer immediately
and then proceeds with the call in band therefore sending dialing
sounds back to the caller. Other SIP gateways we have notably the
Vegastream and others do not do a SIP answer until the call is
successfully connected to the called party.
Any ideas?
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Re: [Asterisk-Users] Automatic Codec change for different communication channels!?
This is off the top of my head - never tested For the end user device (ie polycom in your case) your sip settings would be something like [5000] username=5000 SNIP deny=all allow=ulaw allow=alaw allow=G729 which would give you both Then if you in the Trunk set the following [Trunkroute] username=asterisk2 SNIP deny=all allow=G729 Actually thinking some more, this might not work, as your asterisk box may transcode it, although it might not - but even if my logic is flawed here, it might inspire? David On 31/05/05, Kib Eki [EMAIL PROTECTED] wrote: could you please give more information concerning this setting? Pavel Jezek wrote: you can try use variable preffered_codec in dial command (if you now the prefixes/dial numbers, for which to use eg. g729)... PJ Kib Eki wrote: Hi, I am looking for a way to let * choose the voice codec relying to the used communication channel. Example I am using a Polycom 500 which supports G729 and G.711. When I am doing internal calls (with my LAN) or calls over the PSTN (ISDN) I want to use the G.711 codec because there is enough bandwith. When I am doing inter asterisk calls (over my WAN to another * server) I want to use G.729. Is there a way how i can achieve this? Kib ___ 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 ___ 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] Connecting a peer to a dynamic ip asterisk box ???
Yes, as long as the phone is happy to take something other than a strict IP address, then a dynamic DNS provider is a good way to achive this. On 31/05/05, Manjit Riat [EMAIL PROTECTED] wrote: Hi, I prevoiusly has asterisk on a public static ip and had a phone from a different location registering to the asterisk box. But now we have dropped the previous connection and the current connection has a dynamic ip. Is there any way for the phone to register to now-dynamic ip addressed asterisk box (using something like dyndns.org or something). Thanx ___ 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
Re: [Asterisk-Users] retail unit for cards
assuming you mean digium zap style cards, yes there are several. I don't want to directly quote you any as I have a relationship with a number of them, however googling for digium wildcard brings up several David On 5/19/05, Iqbal [EMAIL PROTECTED] wrote: Hi Does anyone know of a retail outlet in the UK where you maybe able to purchase cards for asterisk. Iqbal ___ 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
Re: [Asterisk-Users] User cannot dial
When you say she can't dial out, what error message is she reciving? (if your using the windows version, turn off the skin, then you get an info button, click on that and you get another box below the user side - it gives some debug but not a lot) does you asterisk box see any packets from her? As she is behind a firewall, and you can ring her, it means that your asterisk box has seen her register requests and has communicated with her, so its unlikely to be the SJphone, unless there are some wayward settings on it also what settings does she have on her asterisk profile? does it work with x-lite if (on asterisk cli) you do sip debug ip (her ip address and port) ; or sip debug peer peername and try to make a call - do you see anything comming in? David On 5/19/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote: I have a user connecting from behind a firewall. The location is remote and I have no access to the firewall to so any port forwarding. She is using SJPHONE as the client. I can dial the extension and she can answer, we can converse. However, she cannot dial out. Any ideas what causes this? Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 ___ 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
Re: [Asterisk-Users] Help with extensions - can't dial 700
Chris, Don't forget that a change in features.conf requires a restart of asterisk (or the modual features.c) - you can't get away with just a reload. On 5/17/05, Chris Mason [EMAIL PROTECTED] wrote: Thanks, I removed that and will test. I don't have an analog extension here, I am testing using SIP remotely, will have to go to the resort to test. Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ 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] zaptel.conf in /etc not /etc/asterisk - historical reason?
Hello all I am in the process of trying to create a more fault tolerent HW setup for my asterisk platform, its all going well and I intend to do a wiki about it once its seen to be working. One thing gets me, and hopefully someone here can confirm my suspision - why is zaptel.conf not with the other asterisk files (I assume it is because its responsable for bringing up the hardware, not strictly part of the asterisk application) Would someone care to confirm my suspision, and if I'm wrong advise me why. As a follow on to this - if i were to move it somewhere else, is it the somthing.c file that would need to be changed to reflect this move. Thanks David ___ 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: zaptel.conf in /etc not /etc/asterisk - historical reason?
Thanks for getting back to me, the only reason that I see to move it (and more importantly to move it to /etc/asterisk) is that I am intending to use DRDB to make the machines as identical as possible, and to ensure that the configs of the two machines are kept in-sync. My mount points for the 3 replicated drives were going to be /etc/asterisk /var and /home (or /users) I cant replicate /etc as things need to be different in some of its child directories (init.d and sysconf are two) (although I guess I could as I'm not intending to replicate /var/spool/ and thats below var) If zaptel.conf moves to /etc/asterisk, it keeps my replication simpler than adding lots of mount points nb - DRDB is a replication technology (laymans term I know) (commonly used with linux-ha) I agree it doesn't belong in /etc/asterisk, but its convient, especially since I know of no other application that interfaces with it :) David On 5/16/05, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], David John Walsh [EMAIL PROTECTED] wrote: One thing gets me, and hopefully someone here can confirm my suspision - why is zaptel.conf not with the other asterisk files (I assume it is because its responsable for bringing up the hardware, not strictly part of the asterisk application) Yes. Zaptel came before Asterisk and is independent of it. It is possible for other non-Asterisk software to make use of Zaptel, without Asterisk needing to be present at all. Would someone care to confirm my suspision, and if I'm wrong advise me why. As a follow on to this - if i were to move it somewhere else, is it the somthing.c file that would need to be changed to reflect this move. Don't know, but I have trouble understanding the need to move it. The only place that it would make sense to move it to would be /etc/zaptel/zaptel.conf, but since it is a single file, why bother? It certainly doesn't belong in the /etc/asterisk directory. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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
Re: [Asterisk-Users] Asterisk@home 1.0 + Sipgate UK/SIP Provider
-- Executing Dial(SIP/201-fcb3, SIP/sipgate/###) in new stack -- Called sipgate/## Paul I apreciate why you've the dialled digits out there, but would you be good enough to include the first few, as if your asterisk box is sending extra / unwanted / too few digits to sipgate its never going to work :) Other than that it seems someone else has posted config for your reference to check. David ___ 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] AreskiCC Install Problems
Those files I indicated to check : /var/lib/pgsql/data (on a redhat flavor) pg_hba.conf - This one needs lines similar to local all all password host all all0.0.0.0 0.0.0.0 password (not you probably want a more restrictive ip range / net mask here!!) postgresql.conf make sure it has a line tcpip_sockets=true Make sure you have the following packages rh-postgres-server php-pgsql or the files containted within Finally, if you haven't, make sure you restart both postgres and apache to ensure they have seen the changes to the config (apache needs to see the updates containted within php-pgsql as an after thought, it is required that php-globals=on, I have never had to set that and am not sure which file its in (I do belive however that it refers to an apache config file not an areski one) As a hope thought - I have sucsessfuly got both versions of areskicc working at some point, so its not flawed code. On 5/11/05, Julius Igugu [EMAIL PROTECTED] wrote: Make sure postgresql is running and the database username/passwords are correct. --- Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote: I have followed the Idiots' guide for installation, but still could not make it work. When I try to login at the web page coming from /var/www/html/areski , I get the following errors: Can some body give me some hints where and what to check for this error?. I am looking for info on the changes we have to make for 1) the database name 2) user name 3) password 4)connection name (server running postgresql) in all the files involved in the application, so that it works. Seshu --- Warning: pg_pconnect(): Unable to connect to PostgreSQL server: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? . in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 68 Database error: Link-ID == false, pconnect failed PostgreSQL Error: 0 () Warning: pg_pconnect(): Unable to connect to PostgreSQL server: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? . in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 68 Database error: Link-ID == false, pconnect failed PostgreSQL Error: 0 () Warning: pg_errormessage(): supplied argument is not a valid PostgreSQL link resource in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 101 Warning: Cannot modify header information - headers already sent by (output started at /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php:68) in /var/www/html/areskicc/lib/module.access.php on line 66 Warning: Cannot modify header information - headers already sent by (output started at /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php:68) in /var/www/html/areskicc/lib/module.access.php on line 67 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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 Julius Igugu SouthWork Co. Ltd. __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ 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
Re: [Asterisk-Users] Snom 360
Colin Similar to Gary's response in that I haven't seem many of these issues. One that is similar, is that of you saying you need to press voicemail key twice to get *97 (or eqivilent code) This as I understand it is not a fault of snom, but a feature of asterisk and the whole MWI protocol. When asterisk signals the phone to say it has voicemail (any phone) it sends in from an address of [EMAIL PROTECTED]. the message text is basically that which pops up on the bottom line of the display. When you press the voicemail key, or even the soft voicemail key it first tries to make contact with unknown as this helps ensure that the right line acesseses its voicemail without the user having to be aware of which line the voicemail is waiting for them on. You have two choices, a change the address of the MWI indicator to come from [EMAIL PROTECTED] on the asterisk box or add some lines in your message-centre context that is similar to exten = Unknown,1,Voicemail etc Either of these will bring asterisk up to the level of the snoms features. I have only one minor issue, and thats if I have several people ringing into the phone, when I am not already on a call (all calls are still in the setup phase) I can't choose by pressing the flashing lights, I have to dump them using the soft no thanks or the hard x key You almost sound like you have a earlier firmware issue. The latest one is 3.60f a direct link to the firmware is http://www.snom.com/download/share/ I tell a lie -the very latest firmware is 3.60h - as of the 4th May David On 5/12/05, Gary Stimson [EMAIL PROTECTED] wrote: Hi Colin I've been using a Snom 360 for 2 weeks and am generally pleased with it. On Wednesday 11 May 2005 22:12, Colin E. McDonald wrote: I am having major problems with the first run of Snom 360s that rolled out last month. Issues: Speakerphone/Hands Free volume spikes up and down during a call. Haven't seen that problem. You have to manually set the volume during every call. When you set the volume, press OK. Then it's stored for next time. This makes it totally unusable. The sound will cut out completely at the beginning of a call sporadically. Have you tried a different provider? Call comes through speaker phone after you pick up handset and then cuts to handset a couple of seconds later I don't have that issue. There is a mnaufacturing defect where the display cable is disconnected so you get what appears to be DOA desk sets. Nor that one. Maybe I was lucky! Have to press the retrieve message button twice pretty regularly to get it to dial vociemail (*97) in asterisk Haven't got the VM button configured yet, or tried to. Major problem with calls being dropped when you place callers on hold I haven't tried putting callers on hold yet. Have you updated to the latest firmware? Copy the firmware URL from snom.com into the relevant box on the phone's web interface, save and reboot the phone. Gary -- Gary Stimson Zedcore Systems ___ 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
Re: [Asterisk-Users] AMP and dialparties.agi
dialplan is the logic giver for Call Waiting, Call Forward Do Not Disturb and a couple of other things that escape me. is it required? thats a decsion you need to make, technically it is not required in either AMP or asterisk, but it is a good way of achiving these features. hope that helps David On 5/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, i dont know if this is the right place to ask for AMP questions, im using it in production and have noticed high cpu usage and even hangs with the dialparties.agi scripts, is this scripts really necessary?, why not use DIAL command directly? 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 ___ 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] What do you name yours
Gents Whilst I apreciate the sentiments regarding my question, if you are to look at my track record of helping people - across in the majority [EMAIL PROTECTED], AMP lists and to a lesser (but growing) extent asterisk-user and asterisk-biz, its not up there with the super gurus, but I am putting more back into the list. I have asked my fair share of questions as well, (and will continue to do so when needed). I noticed that you didn't make these comments when people have talked exclusivly about polycom or cisco hardware (as technically these are not asterisk either) There is a high technical aim to the list, there is also a community factor. I would however kindly ask that for the benifit of the list any further discussions regarding this are directed to me personally, the header information contains my email adress. Regards David ___ 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] What do you name yours
I quite like the idea that came about earlier with regards to Romand and Greek gods, I am thinking (if I ever get off the phone to google today) of findind the roman and greek gods of communication.. On 5/11/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On May 11, 2005 03:38 am, David John Walsh wrote: I would however kindly ask that for the benifit of the list any further discussions regarding this are directed to me personally, the header information contains my email adress. Nonsense; it's little sidetracks like this that make the list interesting. You've done absolutely nothing wrong, and I for one am enjoying the different naming schemes (some old, some new) that are coming up here. Don't let the odd social miscreant scare you off. -A. ___ 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
Re: [Asterisk-Users] What do you name yours
very nice touch! I like that - apart from parc asterisk (the theme park just outside Paris) I went there as child and went on the seven loop roller coaster, as we went around the loop, we saw something drop past (i thought it was someones glasses / wallet) it was a wheel. still went on it again an hour later once they put it back on!!! David (it was asterisk related, and I was a user of their service!) On 5/11/05, Steve Kennedy [EMAIL PROTECTED] wrote: On Wed, May 11, 2005 at 05:40:57PM +0200, Dave Cotton wrote: On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: Naming Conventions for Asterisk Hostnames, . For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. There's a whole french comic suited to an Asterisk naming convention. I'll leave it as an exercise for the reader ... Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 Skype/In callto://stevekennedyuk / UK callto://+442088167166 US callto://+13106518226mob 07775 755503 Personal Blog http://stevekennedy.blogspot.com Euro Tech News Blog http://eurotechnews.blogspot.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 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] outbound proxy field in sip.conf
I have been given the following settings for connecting to a voip provider. The names of the fields match my snom phone, and when configured, the phone both makes and recives phonecalls without issue. I am trying to put the same values in asterisk, but there seems to be one field that doesn't seem to exist in asterisk - that of outbound proxy all suggestions welcome SIP headings account= user password = secret registrar = host = registrar.provider.com outbound proxy = ?? = nat.provider.com:5065 If I put in an extra field of port=5065 it doesn't register (in sip show registry) with either of the above addresses in the host box. Putting registrar.provider.com in the register = string and nat.provider.com in the host makes outbound calls fail, putting registrar.provider.com in the host and nat.provider.com (with or without the :5065) in the registrar doesn't allow incoming or outgoing calls putting reistrar.provider.com in both host and the register = string allows outgoing but not incomming firstly, what does the outbound proxy do? secondly, can anyone advise on settings for this senario? does asterisk have this concept in its SIP client? Thanks David ___ 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: snom mass deployment - settings via DHCP
bootfile-name seems to come into play when you do a Reset values not a reboot i don't know why there is 2 types of behavior snom 360 with 3.60f David On 5/11/05, Stefan Tichy [EMAIL PROTECTED] wrote: I have to adjust my last statement. If Setting URL field of advanced.htm webinterface is empty the value of tftp-server-name is used. option tftp-server-name http://192.168.100.1;; On reboot the phone sends two requests to the specified IP: GET /download/snom190.htm GET /download/snom190-mac.htm bootfile-name does indeed seem to be ineffective. I tried using snom 190 with 3.60b firmware and a dhcp-3.0.1 server. -- Stefan Tichy [EMAIL PROTECTED] ___ 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] high availibilty (heartbeats) - a good way to ensure automatic redundency?
being from a telecoms background, the thought of a single asterisk box solution (even in a low production environment of say 10 phones) worries me slightly! starting from say a base of [EMAIL PROTECTED], you would have several MySQL databases, in addition to numerous config files. I have looked at high availiblity solutions, and from a hardware monitoring point of view, its relitivly straight forward, you have 2 (identical?) boxes, each with 2 network interfaces. One of the network interface cards on each box has the same IP address, there is another cable that is sending a heartbeat message between the two boxes, heart beat fails the other box brings up automatically the interface. I have used HA on firewalls, and as the equipment is propriertary you talk to the master side, which pushes config and state to the slave, in the same way telephone exchanges run 1 micro-instruction behind the other. Obviously howver if you have a PRI on the box, it will lose its calls, IP could be more resilient. asterisk as it stands isn't geared up for this push of state, so leaving that to one side there are a few obvious questions, but firstly my assumptions. MySQL has some sort of master/slave database system built in, so that config is ok. AMP self generates the dynamic config, so a cron job to reload the slave every few minuites is possible to keep that part in sync to my questions (sorry its dragged slightly). How can the astdb (the one that you type show database at the asterisk cli) be kept in sync?? is this the right way to design a warm standby system or is there an already established method. The wiki suggests HA, but doesn't specify how. Googling doesn't seem to find anything (I have been trying on and off for a couple of days now) Thanks for any comments ___ 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] AreskiCC Install Problems
I haven't done this for ages, but those errors if i recall mean that you either haven't got postgres listening for the tcp connections, or its being restricted by host based authentication one of the files is something_hba.conf and the other is in the same directory, in the HBA one, you have to set a line up for 127.0.0.1 to do passwords. In the other there is a #value tcp_connect=false. You have to unhash the value then change it to true (the line above talks about ssh) I will look at the servers tommorrow, it really has been months since i did that. The other issues is do you have pg_connect installed (and again i cant for the life of me rember the package) If no firmer response has come your way (its 23:00 here now) by tommorow I will check the settings on the server in the office David On 5/11/05, Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote: I have followed the Idiots' guide for installation, but still could not make it work. When I try to login at the web page coming from /var/www/html/areski , I get the following errors: Can some body give me some hints where and what to check for this error?. I am looking for info on the changes we have to make for 1) the database name 2) user name 3) password 4)connection name (server running postgresql) in all the files involved in the application, so that it works. Seshu --- Warning: pg_pconnect(): Unable to connect to PostgreSQL server: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? . in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 68 Database error: Link-ID == false, pconnect failed PostgreSQL Error: 0 () Warning: pg_pconnect(): Unable to connect to PostgreSQL server: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? . in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 68 Database error: Link-ID == false, pconnect failed PostgreSQL Error: 0 () Warning: pg_errormessage(): supplied argument is not a valid PostgreSQL link resource in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 101 Warning: Cannot modify header information - headers already sent by (output started at /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php:68) in /var/www/html/areskicc/lib/module.access.php on line 66 Warning: Cannot modify header information - headers already sent by (output started at /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php:68) in /var/www/html/areskicc/lib/module.access.php on line 67 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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
Re: [Asterisk-Users] Sizing a machine
Jacob Noting that you haven't said what codec you aim to run, heres my user experience so far: I have a dual xeon 3.06 with 3Gb of RAM - it handles easily a single PRI, 15 IP trunks (15 voip lines if you will) and 170 different SIP registations, actually make that about 220 registations (several phones have 2 different lines) Call recording, queues and other funky stuff runs on this box as well I can't promise anything, but it sounds like you should be somewhere close David On 5/10/05, Jacob Cazzell [EMAIL PROTECTED] wrote: Hello all, I am trying to determine if a machine I currently have would be adequate for the volume I want to put on it. I will be utilizing a PRI and I have approximately 70 extensions. We do not have extremely high call volume in or out. I could see maybe having the PRI fill with inbound/outbound calls, but that would be a fairly rare occurrence. There would be some light calling from extension to extension in the office. I have a dual Xeon 2.4Ghz with 2GB of RAM available. Would this machine bear the burden of 70 SIP registrations with maybe 25 active at any one time? Thanks! Jacob ___ 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] What do you name yours
Hello list we are installing 2 new servers (to run asterisk) shortly, for a stand alone service. Ignoring our current naming convention, we'd like to name them something.. but we are not sure what. a consideration is that on the screens of the phones it shows [EMAIL PROTECTED] (eg [EMAIL PROTECTED]) (all extensions are numeric) so the users will see it everyday i'm not creative in this way, it doesn't need to be a silly reference (like jarjar and anikin etc) per se but im curious what would you name them? ___ 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: snom mass deployment - settings via DHCP
Stefan I have raised the same point (and requested a working example from snom) in the same way. The white paper does suggest this (As both you and Andrew Latham point out) but does indeed seem to be ineffective. David On 5/6/05, Stefan Tichy [EMAIL PROTECTED] wrote: Hello, On Fri, May 06, 2005 at 02:13:02PM +0200, Nils Ohlmeier wrote: Regarding the real topic: did you already read our white-papers about mass-deployment and setting up snom phones? http://www.snom.com/white_papers.html My Snom 190 gets several options via dhcp, but defining the setting server URL does not work (in my environment) .. can be set ... automatically via DHCP (options 66 and 67) ... /etc/dhcp/dhcpd.conf 66: tftp-server-name 67: bootfile-name Is this correct? If not, do you have a working example? -- Stefan Tichy [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Cisco Mass Deployment
Being cheeky - can the next one be Zultys :) On 5/9/05, Andrew Latham [EMAIL PROTECTED] wrote: Cisco Mass Deployment just added.. http://www.voip-info.org/tiki-index.php?page=cisco+mass+deployment -- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) ___ 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
Re: [Asterisk-Users] snom mass deployment (probably off topic)
Nils
Firstly to the flash issue, if you are on a DHCP server this seems to
work. If you are not then my particular handset does indeed lose its
config - or rather it invokes the start up wizard which does not
have the last settings to confirm, so to all intents it apears to have
lost its settings (although I now freely admit if you give it an IP
address and go into the web interface, all is not lost) - therefor my
appologies on that, however I would prefer to see the last settings in
the setup wizard.
With regards to my second issue, regarding creative dns records
locally. You state in your white paper, that you need to invoke
options 66 and 67 for the snom device to check an update server other
than snom's its self.
Prehaps again it is my error, however looking at ethereal packet
traces, the phone is supplied the dhcp options, but still attemps to
goto snom.com - would you be kind enough to supply sample config for a
dhcpd server, so I can be sure of the setup
Other than those two minor observations - I do indeed think that the
phone is one of the best in the market place, and I have bench tested
most user agents to date.
Thank you for your time on this matter.
David
On 5/6/05, Nils Ohlmeier [EMAIL PROTECTED] wrote:
Hello,
to prevent further rumores and wrong facts about our phones:
All our phones, and this includes the 360 as well, do store their settings on
the flash. After the settings are stored once, you can leave the phone as
long as you want without power, and it will come up with old settings
whenever you restart it. If this is not the case the person which is facing
the problem should contact the snom support to get this sorted out.
Regarding the real topic: did you already read our white-papers about
mass-deployment and setting up snom phones?
http://www.snom.com/white_papers.html
You do not have to mess around with faked DNS responses. If the phone is
getting a settings server via DHCP, it will never contact snom.com (expcept
you say so). The phone just falls back to snom.com as a default setting, in
case it cant find a setting server locally. But in case: you can even turn
off that the phone tries to load any settings from any server.
Best regards
Nils Ohlmeier
On Friday 06 May 2005 05:40, Daniel Bingham wrote:
Hi David,
First, thanks for the reply to my questions about the Snom 360. I may have
a few followup questions when I get a little more time.
As for the 360 getting the configuration directly from Snom's servers, I
find that very backwards. What if your phones have no gateway to the
internet? It sounds like they are working around not having any flash
memory, but it's a poor workaround. Your idea of using DNS to fool it into
going to your servers is a good one. I assume you'll just put in a mapping
for provisioning.snom.com or just snom.com in your DNS server to the IP of
your web server.
If I understand correctly, you will be able to create an
/snom360/snom360.php script on your web server, which you would then like
to redirect to the static html files in the /snom directory. Assuming the
Snom supports redirects, the PHP code is as simple as:
?php header(Location: /snom/snom360-$_GET[mac].html) ?
If the phone doesn't support redirects, it gets a little complex, in that
the script will need to open the file from the filesystem and return it
directly.
If I misunderstood or I didn't make sense, I'll be happy to try again.
Thanks,
Daniel Bingham
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] ] On Behalf Of David John
Walsh Sent: Thursday, May 05, 2005 8:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] snom mass deployment (probably off topic)
Hello
Although not stictly a asterisk issue, any help would be apreciated.
Firstly a few notes on the snom 360, which I have had on a test bed for the
last week. Its a great phone, with a good user interface, both physically
and its web based one.
At its lastest firmware it does have a few quirks, with regards to the way
it handles usernames and passwords on the physical interface. These have
been passed back, and hopefully will be addressed.
Its worst feature as I see it is twofold, with regards to its power fail
features. If it loses power for more than a few minuites it loses its
settings - not the best thing in a world where routers and firewalls can be
given power back days later and be fine.
It has an interesting configuration mode, it tries to contact snom, who
then (if told about it) goes to their national distrubtor who then either
has your config or passes it on again
The settings file is well documented, and you can pull them direct from
phone in a ready to go way.
---
I now have my configs in the file name format of snom360-{mac}.htm (where
{mac} is the MAC address
Re: [Asterisk-Users] ISDN transfer, handoff to masterswitch
Peter For PRI's is ECT / CD the default behavior of asterisk, or is there code changes (and what are they) to make these features work. In about 4 weeks, we are getting a test PRI, the quad-span digium wildcard and a test server. The behavior we want is not to tie 2 circuits up in the event of transfer, but to keep asterisk out of the loop and pass it back to the upstream switch. Thank you for any comments. David On 5/4/05, Peter Svensson [EMAIL PROTECTED] wrote: On Wed, 4 May 2005, Alex Mack wrote: So I'm already doing ECT by using the bristuff'ed version of *? I have no idea. We use PRI only, not BRI. Hopefully it is in the documentation for bristuff. Peter ___ 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] snom mass deployment (probably off topic)
Hello
Although not stictly a asterisk issue, any help would be apreciated.
Firstly a few notes on the snom 360, which I have had on a test bed
for the last week. Its a great phone, with a good user interface,
both physically and its web based one.
At its lastest firmware it does have a few quirks, with regards to the
way it handles usernames and passwords on the physical interface.
These have been passed back, and hopefully will be addressed.
Its worst feature as I see it is twofold, with regards to its power
fail features. If it loses power for more than a few minuites it
loses its settings - not the best thing in a world where routers and
firewalls can be given power back days later and be fine.
It has an interesting configuration mode, it tries to contact snom,
who then (if told about it) goes to their national distrubtor who then
either has your config or passes it on again
The settings file is well documented, and you can pull them direct
from phone in a ready to go way.
---
I now have my configs in the file name format of snom360-{mac}.htm
(where {mac} is the MAC address of the phone in question)
The phone initally tries to goto
provisioning.snom.com/snom360/snom360.html this sends it onto
http://snom.com/snom360/snom360.php?mac={mac}
Assuming that I perform some creative dns records on my dns server,
would someone be kind enough to write some sample php code to take the
url
http://snom.com/snom360/snom360.php?mac={mac}
and provide the url http://asterisk-demo/snom/snom360-{mac}.html
The code the url needs to go in is as follows:
# Redirect all phones to the php script
setting_server: http://asterisk-demo/snom/snom360-{mac}.html
I'm useless with php and most launguages, so thank you to any help
this request generates
David
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Re: [Asterisk-Users] Opinions on Cisco 7960G, Polycom IP-600, and Snom 360
Dan I've had a snom 360 on the bench for about 10 days now. I too have become dispondent with Cisco's licencing structure The snom 360 provides the majority of the high end Cisco features you would use 99% percent of the time, it doesn't have the flexibity of the on-demand buttons around the screen, but neither does it have a (IMHO) over complex settings file that the ciscos and polycoms seem to suffer. As it stands today, there are a few minor bugs in the firmware (sadly noted in the username / password login wizard) but nothing show stopping, which I feel is impressive given the phone is only a few weeks old (to the public). I have just posted another post asking for help regarding its provisioning (i'm useless at php), as it doesn't keep its settings for more than a few minuites without power At its price point, it seems a bargain for its power and flexibilty. I have been sent a pre-release of its manual (in its very final draft) which should be freely availible within the month. If it had the abilty to save its settings unless you did a reset, I would recomend it beyond doubt, as it stands its going to take a lot to beat it The softphone (if you have a windows pc) gives you a feel, but doesn't do it justice. If you have a few php skills, you help woluld be greatly apreicated - please see my other post :) Regards David On 5/6/05, Daniel Bingham [EMAIL PROTECTED] wrote: Apologies for asking more questions so quickly after my last one. A few more questions about the Polycom phones: Searching the list I found a few references like this: I would also like to figure out how to make the phone *ring* when you're already on another line, but haven't had a chance to seriously explore it yet. Is this still a problem in the latest firmware? This could sink my hopes of going with a Polycom phone if there isn't a way to have them give an audible alert that another line is ringing while you're already on the phone. The Wiki says the IP-500 requires an additional chip to support power over ethernet. Is this true of the IP-600 as well? If anyone can answer any of these questions, I would really appreciate it. Thanks, Daniel Bingham [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, May 05, 2005 11:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Opinions on Cisco 7960G, Polycom IP-600,and Snom 360 Those are all three great phones and the choice gets really preferential... I love my Polycoms and I recommend them all the time. I give props to the Cisco stuff but like you, I can't stand paying extra even if it is just a few bucks here and there. Polycoms can have a curve for figuring out the config files but once you do it is a breeze. The speakerphones are excellent and the features work with * with no real headaches. The IP500 (or even an IP300) is sufficient for most users so save some bucks if you don't really need the mini-browser and extra display lines of the 600. An IP 500 can take plenty of concurrent calls and the features are excellent. I will let the others speak about SNOM and Cisco though I can say they are well respected. My preference is just Polycom. If you get the Polys let me know if you have trouble and I will assist you with config off list. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Bingham Sent: Thursday, May 05, 2005 8:36 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Opinions on Cisco 7960G, Polycom IP-600,and Snom 360 Hello, We are planning to replace our current PBX with an Asterisk / SIP solution, and are now trying to decide which phones to get. My first thought was the Cisco 7960G, but the Cisco licensing scheme irritates me enough that I'll probably end up going with either the Polycom IP-600 or Snom 360. If anyone has any opinions of these phones, especially in comparison to each other, I would really appreciate hearing them. Is there a reason you would recommend one of these phones over the others, or any reason why you would steer people away from a particular model? This is for a small office, with only 8-10 phones. A receptionist and a couple of office staff will be responsible for watching the office line(s), and three or four support reps will be watching a technical support queue. Our environment dictates that we move around a lot, and not necessarily be tied to our workstations, so being able to take calls from any given phone is an important consideration. In the same vein, knowing the status of other staff (i.e. if they are on a call or idle) would be very useful, and is something we are used to with our current setup. Thanks, Daniel Bingham [EMAIL PROTECTED] ___ Asterisk-Users
Re: [Asterisk-Users] Newer Dell Servers + TDM card
I know I may be asking the obvious, but shouldn't we get tother to build a matrix (prehaps on the wiki) showing boards / servers that work against cards. Unless it is already in place, in which case would one of you be kind enough to point me in the right direction? Thanks David On 5/5/05, Nathan C. Smith [EMAIL PROTECTED] wrote: There is a thread in dev or biz about this too. A guy got referred some motherboards by Digium he can't get easily in Australia. I'll second that the DL380 G2 seems to work, I'll know more in a few months. The common thread seems to be either a serverworks chipset or more specifically, a chipset optimized for PCI bandwidth - two or three separate PCI buses with a bus dedicated to the Digium card. I asked the following to Digium about the new Asterisk Business Edition Will there be a recommended hardware platform or reference system? Yes, two servers are being used for the initial certification, a Dell PowerEdge and a HP/Compaq. They recognize people like or are restricted to Dell and HP/Compaq. -Nate -Original Message- From: Charlie Watts [mailto:[EMAIL PROTECTED] Sent: Thursday, May 05, 2005 2:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Newer Dell Servers + TDM card David Brodbeck wrote: I find this game kind of infuriating. If you have problems, they tell you to buy a different motherboard. But they don't supply a list of approved ones that they'll support. One fellow at Digium suggested to me that the HP/Compaq D380 works well. And it comes with a motherboard ... ___ 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
Re: [Asterisk-Users] Newer Dell Servers + TDM card
Ok that raises the simple question How do we know what firmwares will work, which won't Also how recent is the upgrade, will it have filtered thru to international distributors etc regards David On 5/4/05, Mark Phillips [EMAIL PROTECTED] wrote: Folks, This is a firmware bug in the TDMxxx and TExxx cards that Digium has recently fixed. I did an advanced replacement for mine which involved me buying another one and them refunding me when they got my old one back. Get onto their tech support. Mark Matt Schulte wrote: Is this with the TDM400P card right? -Original Message- From: David Brodbeck [mailto:[EMAIL PROTECTED] Sent: Monday, May 02, 2005 2:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Newer Dell Servers + TDM card -Original Message- From: Matt Schulte [mailto:[EMAIL PROTECTED] Really, how long does it take to recover? Mine just totally locks. No time at all. The only reason I know an NMI occurs is the front panel light, and the Dazed and confused, but trying to continue message from the kernel. I'm using a Dell PowerEdge 800. ___ 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 ___ 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] Taking asterisk out of the media path - SIP - how is it achieved
Hello How do you make asterisk stay out of the media stream? i.e once I set a call up between two parties, even if asterisk fell over the call would continue (in the same way a HLR on a mobile network works) I understand that many features will be lost if I do this, but all that I need seems to be supported by the end user hardware. incidentally I have tried canreinvite=yes, doesn't seem to work. I have also tried removing any flags in the dial() command Thank you for any information. David ___ 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: RE : [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?
what sort of level of PC is required for 300 concurrent calls? Regards David On 5/1/05, Hakem Taourchi [EMAIL PROTECTED] wrote: Can this Dell run 90 calls simultaneously ? Or need a higher Dell machine? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ariel Batista Envoyé : samedi 23 avril 2005 1:27 À : 'Ben Hencke'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : RE: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P? I just setup a SC420 with two TDMO4b cards in it and it works just fine. No problems what so ever with it so far. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Hencke Sent: Friday, April 22, 2005 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P? I have head that the SC prefixed Dells are not good to use with Digium hardware. Considering how picky my TE405P cards were in other low end Dell servers, I would suggest using an 1850 instead. OTOH, if it does work, please let me know :-) If you go to small biz, you can get the SC1425 trimmed down with dual 2.8hgz for under $1k - Ben On 4/22/05, Greg Boehnlein [EMAIL PROTECTED] wrote: Hello, I've been asked to build a couple of Gateway servers for a client w/ TE405P hardware, and have been looking around at various 1U options. I've been looking at SuperMicro and Tyan barbones boxes as possible platforms, but then was directed to Dell's SC1425 by a friend. Short story, is that you can purchase a 2x3.0Ghz/1GigDDR400/1xSATA box in a 1U form factor for $1,498.00. This seems almost too good to be true, so I'm asking if anyone has had any experience with this box? I'm not up on my PCI terminology, but as I understand it, the TE405P can only be used in a 32 bit 33Mhz slot at 5.0 Volts. This SC1425 lists a 1x 64-bit/1xxMHz PCI-X slot under it's expandability information. I'd venture to guess this is probably NOT going to work with a TE405P. That being said, if it works, great. If not, what 1U boxes are people using IN PRODUCTION w/ TE405P cards? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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
Re: [Asterisk-Users] how to use dialparties.agi
Christian As I understand it After a user dials an extension number, Asterisk calls dialparties.agi dialparties.agi checks the asterisk database (show database [from cli]) for data matching items like Call Wating (CW) Call Forward (CF) etc. If one is present (in a defiend order) then rather than dialing, dialparties invokes that option. If none of the options are set, dialparties returns control back to a near regular dial string, and Dial takes over and places the call as the A party was expecting. Using defined etensions (by default in AMP they are the regular American ones), the B party (callee) can activate these features. What basically happens here is a database put command is used to put the value in the asterisk database and then play a recorded anouncement to the user before hanging the call up. for CF its a little more complicated as you might have to specify the B number and the C number, but essentially it puts the data in the database and confirms it Now the only thing that is missing is a web / gui provsioning system - so that admins can take the features off again, else its a databse del command at the terminal --- the best way to see this in action is to set some things like CW (*73 i think) and then do a show database at the CLI - you will also get back other things like the SIP registery David On 4/26/05, Christian Wengel [EMAIL PROTECTED] wrote: Hi! I looking for an example how to use the dialparties.agi from Asterisk Management Portal 1.10.007a. I tried to understand it by reading the extensions.conf of AMP, but without success. Is anybody out there, who can give me a more easy example or an explanation. Thanks, Christian ___ 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
Re: SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?
Ronald,
I am more than happy to give you the 3 suggestions, when you
appologise to the list. Yes getting things to work can be
frustrating, and sometimes the answers are not as helpful as you'd
like, but I do refuse to help people who get irate on a public list
Especially when the outburst is to those who spend hours creating
programs that help many many people, those people who have talent
beyond my wildest dreams.
Please remember all advice on here is of a volentary nature, a lot
from people who could earn their crust providing this advice for a
charge, they don't, they spend hours helping and most of the time we
get it working - together
Now, take a deep breath, do the gentlemanly thing and lets see if we
can fix your issue.
David
On 4/26/05, Ronald Wiplinger [EMAIL PROTECTED] wrote:
Robert Goodyear wrote:
On Apr 25, 2005, at 10:55 PM, Ronald Wiplinger wrote:
Hi Ronald,
What happens in your Asterisk box when you press the Speed Dial
number in
IPS?
Can we make it so that you FIRST answer below questions, please?
| | Let's try it together:
Ronald: wow. Take a breath before you torch a generous developer. IPS
works like a charm for me in every way.
Seriously,
/rg
For me it doesn't!!!
And IF it works so good for you, why you are not willing / able to fill
the simple three fields out for me, telling me what you think would work.
I ask this question now several times, and get only answers, like it
works, it works for me too
FOR ME IT DOES NOT WORK, ...
So how did yo make it that it works for you???
Questions below to be kindly filled out:
| | Let's try it together:
| 1. Open IPswitch
| 2. Open Extensions tab on top
| 3. Switch to the tab Speed Dials on the bottom
| 4. Fill in:
| Name: [EMAIL PROTECTED]
| Caller Id: Peter
| Visible on Panel: (ticket)
| Exentension Group: Speed Dial Numbers
CLI answers:
| | | Congratualtions, you have successfully installed the Asterisk Open
| Source .
| tgj wrote:
|
|
|
| Hi Ronald,
|
| I must admit I am getting confused now.
|
| I understand that you have a problem getting Speed Dial Buttons to
| work.
| The problem as I understand it is that the calls are placed in the
| wrong
| context.
|
| To solve that problem I have asked you to make sure that you have
typed
| a
| valid context on the configuration page. Have you tried that?
|
| I think thats all you need to do, how do I post an example of that?
| It's a
| fairly easy thing to do.
|
| Thorben
|
|
|
|
|
| What is the right syntax to do that?
| Context for dialing a trunk line is trunkint
| Peter has the phone number 011-234-5678
| How to set it up as a speed dial number? Below are all info you may
| need:
|
| The phone 601 (= Monitor extension) is a Sip phone,
|
| [general]
| context=default; Default context for incoming calls
|
| [601]
| type=friend
| username=601
| secret=dont+tell+you
| canreinvite=no
| host=dynamic
| dtmfmode=rfc2833
| [EMAIL PROTECTED]
| nat=yes
| callgroup=1
| pickupgroup=1
| callerid=Ronald Hotline,601
| qualify=1000
|
|
| extensions.conf
| [default]
| ...
| include = trunkint
| ...
|
| [trunkint]
| ;
| ; International long distance through trunk
| ; . other lines deleted
| exten = _9011Z.,107,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
| exten = _9011Z.,108,hangup
|
|
|
|
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Re: [Asterisk-Users] SIP behind IPTables/NAT
First off Isn't RTP a TCP protocol? or am I over tierd again? Secondly - unless several conditions are met (canreinvite=yes being one of them) it (asterisk) will still proxy the connection. - Check your dial statement for T's ie T and t - the wiki has a full list. David On 4/26/05, Ian Pattison [EMAIL PROTECTED] wrote: Hi All, Can anyone help me out here? I'm having some issues configuring my IPTables firewall to properly NAT SIP and RTP packets to my asterisk server hiding behind it. Here are my current rules: #Inbound SIP to HERMES $IPTABLES -A PREROUTING -t nat -i $EXTIF -p udp --dport 5060 -j DNAT --to 192.168.123.4:5060 $IPTABLES -A FORWARD -i $EXTIF -p udp -d 192.168.123.4 --dport 5060 -j ACCEPT #Inbound RTP to HERMES $IPTABLES -A PREROUTING -t nat -i $EXTIF -p udp --dport 1:2 -j DNAT --to 192.168.123.4:1:2 $IPTABLES -A FORWARD -i $EXTIF -p udp -d 192.168.123.4 --dport 1:2 -j ACCEPT When I dial out via my SIP provider I appear to get a partial connection (the phone rings... that's a good sign) but no audio. Inbound I just get a busy and asterisk sees nothing. SIP SHOW REGISTRY shows me as registered with the remote host. Something else that worries me is that I'm seeing the good old Attempting native bridge... message when the destination picks up which, to my understanding, shouldn't happen since I have canreinvite=no set for both my SIP phone and SIP provider. Make sense to anyone? Ian ___ 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
Re: [Asterisk-Users] Extensions / Contexts
In your channel config (eg sip.conf) you need to call them things like [companya-200] [companyb-200] Then in extensions.conf [companya] exten = 200,1,Dial(sip/companya-200) [companyb] exten = 200,1,Dial(sip/companyb-200) Hope that helps On 4/26/05, Sebastian Silva [EMAIL PROTECTED] wrote: Hi everybody, I am writing here because I can't find the solution to my problem (my asterisk configuration). I hope somebody can give me a hand with it: I need to provide a PBX service to several companies (extensions with softphones and Digium hardware to manage the analog lines), my problem is that I don't know how to configure the contexts to have, for instance, the following scenario: Company A ext 2000 ext 2001 ext 2002 Company B ext 2000 ext 2001 ext 2002 Company A must not to see extensions of company B and viceversa. I know this is possible with extensions, but I don't know how to distinguish when (for example) a sip phone is connecting to the extension 2000 from company A or company B. It is possible to configure my sip.conf (or iax.conf) like this? If it is, how do I need to configure the softphone? Does Asterisk realizes which context to use depending on the username? Does asterisk allows two extension sections with the same number?: [2000] username=companyA_2000 context=contextCompanyA [2000] username=companyB_2000 context=contextCompanyB Any help will be appreciated. Sebas -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ 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] Warm standby boxes - keeping config syncronised?
Ok probably not strictly an asterisk question. I have an asterisk box, which is running some non-critical telephones in our organisation, and if it fails it fails. However comming from a telecoms background I always want to make things recoverable quickly. Since I have little budget, and down time isn't an issue my thoughts are as folllows 2 servers with 2 NIC's each, one nic for managment, one for traffic. 1 NIC on each machine has the same IP address, but only one is plugged into the network at any one time. 2 PRI's that are plugged into the machine that is live to traffic. Apart from the managment NIC having different IP addresses they are configured identically If I make a change to the in-service server, how do I automagically get the other server to take a copy of it? I'm not a linux man by trade, so if you say set up master / slave would you be kind enough to suggest an aplication and how it would be implimented. Thanks for any ideas ___ 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] UK (english) sound files
Alex I too am on the hunt for the same. I am hoping that my good friend with the recording studio and his lovely wife will be able to perform this. My only issue at the moment is getting the scripts that was worked to, failing that, next weekend I am spending hours writing down what alison says :) David On 4/25/05, Alex Barnes [EMAIL PROTECTED] wrote: Ooops dan Outlook to Hades. Forgot to format in plain text. If you have been offended by this please feel free to ignore this thread. If not then I have left the original message below (this isnt a top post I swear) Thanks again alex -Original Message- From: Alex Barnes Sent: 25 April 2005 11:25 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] UK (english) sound files Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real English asterisk prompts. The only one I have found is here http://www.g7ltt.com/VoIP/vmfiles.html And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc) These prompts are actually a lot better than the standard prompts, according to my customer. But unfortunately they arent perfect. For example all of the queue prompts are missing as well as a number of other prompts. Personally I like Allisons sultry tones telling me that shes doing her utmost to connect my call :-) Couple of questions: 1) Does anyone else have english prompts they can share / point me to? 2) Does Mark (the kind guy that made the above) post on this list and is there any possiblity of adding some of the most needed prompts? Failing that I will give him an email and see what the chances are. 3) Failing everything else would anyone be interested in sharing the cost and getting some professional (female?) recordings done for all of the standard asterisk prompts? Currently I'm facing the possiblity of having three different people talking to the caller before they are put through. Company recording warning, UK transfer message and then American queue announcements. :-S So this has suddenly become a fairly urgent matter. thanks in advance for any help / advice on this Alex P.S. Sorry for the double post Asterisk-UK listers but as soon as I sent I thought it would be best to post here as well since this is pretty urgent for me. Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in a writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. 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 ___ 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] What small PC can take 8 FXS + 8 FXO cards
If i'm understanding this correctly, you shouldn't need 16 ports. If you buy 2 TDM400P cards, and load them up with 8 FXS (4 on each card) then buy 2 TDM400P cards, and load them up with 8 FXO (4 on each card) This should reduce your PCI count down to a more manageable 4 cards In total your shopping list would be 4 TDM400P PCI cards 8 FXS Daughter cards 8 FXO daughter cards hope that helps David On 4/25/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The problem now comes in the PCI ports. Is there any PC that can handle 16 ports? What is most optimal solution? ___ 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
Re: [Asterisk-Users] UK (english) sound files
Ian you do realise that alison is actually canadian :) (well as far as I know she is) On 4/25/05, Ian Pattison [EMAIL PROTECTED] wrote: Interestingly enough I'm looking to do the same for a Canadian English version... does anyone to collaborate on this one? Ian Pattison, Senior Analyst Technology Associates Inc. Tel: 905-459-2100 ext. 204 Mobile: 416-568-6548 E-mail: [EMAIL PROTECTED] WWW: http://www.technologyassociates.ca [EMAIL PROTECTED] 25/04/2005 06:24 Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real English asterisk prompts. The only one I have found is here http://www.g7ltt.com/VoIP/vmfiles.html http://www.g7ltt.com/VoIP/vmfiles.html And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc) These prompts are actually a lot better than the standard prompts, according to my customer. But unfortunately they arent perfect. For example all of the queue prompts are missing as well as a number of other prompts. Personally I like Allisons sultry tones telling me that shes doing her utmost to connect my call :-) Couple of questions: 1) Does anyone else have english prompts they can share / point me to? 2) Does Mark (the kind guy that made the above) post on this list and is there any possiblity of adding some of the most needed prompts? Failing that I will give him an email and see what the chances are. 3) Failing everything else would anyone be interested in sharing the cost and getting some professional (female?) recordings done for all of the standard asterisk prompts? Currently I'm facing the possiblity of having three different people talking to the caller before they are put through. Company recording warning, UK transfer message and then American queue announcements. :-S So this has suddenly become a fairly urgent matter. thanks in advance for any help / advice on this Alex P.S. Sorry for the double post Asterisk-UK listers but as soon as I sent I thought it would be best to post here as well since this is pretty urgent for me. Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in a writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. 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 ___ 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: astrecipes v2.0
your queue recipie, does that monitor record from when the agent answers or the music on hold prior to taking the call? thanks On 4/25/05, lenz [EMAIL PROTECTED] wrote: In data Mon, 25 Apr 2005 16:12:08 + (UTC), Tony Mountifield [EMAIL PROTECTED] ha scritto: In article [EMAIL PROTECTED], lenz [EMAIL PROTECTED] wrote: Hello, if anyone is interested, there is a new wiki about Asterisk recipes, i.e. step-by-step descriptions on how to perform something with your * box. This is quite different from most * sites around, that are either questions-and-answers forums or are dedicated to documenting a feature. The point of AstRecipes is how to implement something. See http://www.oinko.net/astrecipes All content is licenced as creative commons, so if you got a recipe to spere, feel free to post it. I've just looked at your Asterisk-OH323 recipe, and wanted to point out that with Asterisk 1.0.x the correct version of asterisk-oh323 is 0.6.5. Version 0.7.1 is only for use with CVS HEAD. Cheers Tony Thanks, I fixed it. See http://www.oinko.net/astrecipes/index.php?n=40 If you notice other bugs or problems, please let me know. l. -- Assum est, versa et manduca. ___ 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] using goto to do selective dialing
If I set up a context, which has lots of extensions eg
[ext-local]
exten = 3000,1,Dial(SIP/3000)
exten = 3001,1,Dial(SIP/3001)
.
exten = 3999,1,Dial(SIP/3999)
(I know the syntax is wrong, and it probably is not the best way to achive it)
then in another context, I use a goto like so
[selected-3000-numbers]
exten = _32XX,1,Goto(ext-local,${EXTEN},1)
exten = _34XX,1,Goto(ext-local,${EXTEN},1)
Will this allow me to only dial (from a phone in
selected-3000-numbers) the numbers starting 32 and 34??
Also is my goto syntax right?
Thanks for your input
David
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Re: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
Taking this idea a little further. (I apreciate there may be legal issues with this request) Would it be possible for extensions to be tagged, so that if they make and / or recive a call the call is automatically recorded each and every time, at the end of the call the file is closed I would imagine, that its either set in the context menu of the extention (ie right click, select always record on active) or in the extensions list. A supervise (either on demand or always) would be a great help as well. On 4/23/05, tgj [EMAIL PROTECTED] wrote: Hi, As mentioned before, how about being able to search and replay recordings from the switchboard. With call records now searchable hopefully it wouldn't take too much more work to enable. For example, being able to search on extension by date and time or by cli would be very handy. Best regards, Steve. Hi Steve, I will implement that too, but in a later release. thorben ___ 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] Recording Queue agents
Hi all We have a psudo-switchboard in the offices, with receptionists who login into the switchboard queue, all is well. I have been asked to look into the possibilty of recording all the agents conversations. Legal issues aside: Can anyone give me any pointers as to how this might be achived. The basic rules are: We only want the conversation, otherwise i'd just monitor all thru the wait period :) when the agent transfers it off, the recording needs to stop. - Am I opening a can of worms here, or is it as simple as setting a flag in one of the commands? All the agents phones are SIP, the queue is an internal extention (so that all can use the switchboard) so just monitoring the trunks is out as well (sadly) Any help (even if its just ideas or sample code) would be greatly apreciated. We run [EMAIL PROTECTED] - to keep things simple (0.6 in production) (0.9 on my test machine) thanks David ___ 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] Billing
To breifly recap Your main asterisk box runs linux, asterisk, ASTCC and MySQL Another box runs linux, mysql, apache The two sql servers are joined, updating each other? or have I missed something? ___ 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 question about queues
it sounds like the default behaivor of an [EMAIL PROTECTED] setup. (B (Bnot that I am knocking [EMAIL PROTECTED] in anyway - its a great way to test (Bnew features. (B (BOn 4/20/05, Henry Devito [EMAIL PROTECTED] wrote: (B Can you post your config's? What version of * are you using? This doesn't (B happen on any of my queues. I have queues set up on several customers (B systems. If there are agents/members available the caller rings them (B directly, no announcements played. (B - Original Message - (B From: "Brett, Gary" [EMAIL PROTECTED] (B To: "Asterisk Users Mailing List - Non-Commercial Discussion" (B asterisk-users@lists.digium.com (B Sent: Wednesday, April 20, 2005 7:21 AM (B Subject: [Asterisk-Users] A question about queues (B (B Hi there, quick question about queues (B (B When calling a queue (which contains eg 4 extensions) it tells me what (B position I am in the queue and then plays some music$B!D(Bthat is (B fine$B!D(B (B however, If there is no-one in the queue , it tells me that im first in (B line (B and then plays hold music while the phones ring. This is annoying my (B callers (B quite a bit . How do I get it so that if I ring the queue, it just puts me (B straight through to one of the available 4 phones, and only if all 4 (B phones (B are busy (ie on calls) then announce a position in the queue and play (B music? (B (B For example (B (B User 1 dials 7272 $B"*(B goes through to agent 1 (B User 2 dials 7272 $B"*(B goes through to agent 2 (B User 3 dials 7272 $B"*(B goes through to agent 3 (B User 4 dials 7272 $B"*(B goes through to agent 4 (B User 5 dials 7272 $B"*(B announces message that you are first in line (B User 6 dials 7272 $B"*(B announces message that you are second in line (B (B (B Any help on this would be greatly appreciated (B (B (B ___ (B Asterisk-Users mailing list (B Asterisk-Users@lists.digium.com (B http://lists.digium.com/mailman/listinfo/asterisk-users (B To UNSUBSCRIBE or update options visit: (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (B (B ___ (B Asterisk-Users mailing list (B Asterisk-Users@lists.digium.com (B http://lists.digium.com/mailman/listinfo/asterisk-users (B To UNSUBSCRIBE or update options visit: (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (B (B___ (BAsterisk-Users mailing list (BAsterisk-Users@lists.digium.com (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home 0.9 released
Would it be possible in a future version to have the extensions have the beginings of class of service, in a similar way to incoming calls. e.g if a user is not logged on define what happens (route to operator etc) define if a user can access outgoing lines (and possibly by time of day) I understand that this may be more of an AMP issue and not relivent to a lot of offices, but in our charity, we offer a counter service, and having phones in the public area needs that kind of lock down At the moment I fudge it with changes to the context, but I am hoping that one day it becomes more user managable so that others can keep the system running in my absence. Thank you for the great product David On 4/13/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: More bug fixes. *69 works now. Cisco stuff works. Lots of other fixes. A wakeup call feature was added on *62 http://asteriskathome.sourceforge.net/ Discussion Forums http://sourceforge.net/forum/?group_id=123387 __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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] Supervisor monitor / barge in - automatically on call setup?
I'm aware of the legal issues surrounding my request, but any help technically would be greatly apreciated On site we have a fully staffed hospital and fire service (its a temporary event for a childrens charity) and an onsite 911 number. If a user dials the number, they goto the emergency crew, and the use of monitor helps to record the call - thats the easy bit I'm in the UK, and its an offence not to pass a 999 (our 911) call out to a 999 centre but with the sheer numbers involved, we have a few choices, only one of which is suitable. If a user inadvertantly dials 999 I would like to pass it to the true 999 and at the same time dial either a special phone, or all the phones in the emergency centre. Upon the centre answering it, it silently monitors the call between the user and the 999 centre. If for whatever reason the centre needs to barge in they can, prehaps even silencing the origninal user. We have a 2 min response time to anywhere on site, the offical user services have about 22, but we know and expect that in a moments panic someone will dial the number automatically Any assistance as to how this can be performed will be greatly apreciated. ___ 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] Version 0.80 of IPS released
On a slightly different note: Is there a setting to force IPS not to minimise every time an action is performed? It gets very annoying after a few minuites and with our reception being very very busy it could get quiet sickly On Apr 12, 2005 7:40 PM, Ivan Meic (Vox Mundi) [EMAIL PROTECTED] wrote: The versions are coming fairly fast I admit :-) I looked at the transfer problem with multiple calls, and I do not have a solution as yet, however it's not forgotten. Ok, thanks. Sorry for being so anxious. :) Ivan . ___ 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] 499 Error on X-lite / asterisk setup
I have a fairly simple asterisk setup - [EMAIL PROTECTED] 0.8 in SIP.conf: Extentions 200 - 204 - username, password, callerid all same as extension Extensions.conf - default build from [EMAIL PROTECTED] 0.8 In x-lite all spaces are either the IP address of the asterisk box or the extension number. On loading of x-lite, asterisk pipes up that the extension is seen dialling anything from x-lite gets to asterisk (seen with sip debug) however nothing comes up in the console (verbose 4) and 2 seconds later x-lite returns an error of 499 Not Acceptable Here In the console I get : SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 172.16.0.32:5060;branch=z9hG4bKAD3D66F2AAAF11D9A5AF000A95D3F194 From: 200 sip:[EMAIL PROTECTED];tag=1211608254 To: sip:[EMAIL PROTECTED];tag=as2a712e55 Call-ID: [EMAIL PROTECTED] CSeq: 16528 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 The AAH was a clean install from the ISO on known to be good hardware, and its nothing I haven't done before Have I missed something? David ___ 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] 499 Error on X-lite / asterisk setup
I recived 2 files from Ben. Two things I forgot to mention : performing a SIP show peer xxx (where xxx is the etension number) shows the UA as being xlite and the right IP address, so i'm fairly sure its registered properly xlite is on a mac, but on previous setups i've never needed to change more than the config. incidentally, without wishing to offend, the names of the files don't seem to be right. David On Apr 11, 2005 6:41 PM, Ben Bush [EMAIL PROTECTED] wrote: Here you go. Ben From: [EMAIL PROTECTED] on behalf of David John Walsh Sent: Mon 4/11/2005 11:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 499 Error on X-lite / asterisk setup I have a fairly simple asterisk setup - [EMAIL PROTECTED] 0.8 in SIP.conf: Extentions 200 - 204 - username, password, callerid all same as extension Extensions.conf - default build from [EMAIL PROTECTED] 0.8 In x-lite all spaces are either the IP address of the asterisk box or the extension number. On loading of x-lite, asterisk pipes up that the extension is seen dialling anything from x-lite gets to asterisk (seen with sip debug) however nothing comes up in the console (verbose 4) and 2 seconds later x-lite returns an error of 499 Not Acceptable Here In the console I get : SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 172.16.0.32:5060;branch=z9hG4bKAD3D66F2AAAF11D9A5AF000A95D3F194 From: 200 sip:[EMAIL PROTECTED];tag=1211608254 To: sip:[EMAIL PROTECTED];tag=as2a712e55 Call-ID: [EMAIL PROTECTED] CSeq: 16528 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 The AAH was a clean install from the ISO on known to be good hardware, and its nothing I haven't done before Have I missed something? David ___ 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
Re: [Asterisk-Users] 499 Error on X-lite / asterisk setup
Ah ha! Thats is Robert, you are a genius!!! Actually that gives a larger issue - it doesn't take a lot for a user to click the codecs off and then its a call to the help desk. I am not intending to use xlite in production, but it does beg the question can it be forced?? On Apr 11, 2005 7:21 PM, Robert Keller [EMAIL PROTECTED] wrote: David, do you have all the codec's enabled: I had that problem until I highlighted all of them. I doubt all are needed, but that helped me. Robert. From: David John Walsh [EMAIL PROTECTED] Reply-To: David John Walsh [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 11 Apr 2005 18:34:46 +0100 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 499 Error on X-lite / asterisk setup I have a fairly simple asterisk setup - [EMAIL PROTECTED] 0.8 in SIP.conf: Extentions 200 - 204 - username, password, callerid all same as extension Extensions.conf - default build from [EMAIL PROTECTED] 0.8 In x-lite all spaces are either the IP address of the asterisk box or the extension number. On loading of x-lite, asterisk pipes up that the extension is seen dialling anything from x-lite gets to asterisk (seen with sip debug) however nothing comes up in the console (verbose 4) and 2 seconds later x-lite returns an error of 499 Not Acceptable Here In the console I get : SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 172.16.0.32:5060;branch=z9hG4bKAD3D66F2AAAF11D9A5AF000A95D3F194 From: 200 sip:[EMAIL PROTECTED];tag=1211608254 To: sip:[EMAIL PROTECTED];tag=as2a712e55 Call-ID: [EMAIL PROTECTED] CSeq: 16528 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 The AAH was a clean install from the ISO on known to be good hardware, and its nothing I haven't done before Have I missed something? David ___ 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
Re: [Asterisk-Users] Low cost box for hosting Asterisk and at leastoneTDM400p
what sort of processing power etc should I be aiming for to support 60 SIP extensions and 60 SIP based lines? Thanks David ___ 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] ASTCC - IVR prompts
Hello Is there a set of ivr speech prompts availible for the ASTCC card system? I can't find them in the CVS or any reference in the WIKI? Thanks David ___ 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] Hardware dimesioning issues
Hello I am in the process of putting together a short term calling card solution that is rapidly deployable for charity events, and would apreciate some guidence on hardware dimensioning for the solution I have a test system running on an old P3 laptop, so in principle the solution works : It is configured as follows: Latest CVS of asterisk (well as of about 3 weeks ago) AreskiCC as the card solution Latest RPM of PostgreSQL Latest RPM of apache Latest RPM of php / pgphp 4 SIP accounts for the phones 1 SIP account with 4 concurrent calls for the lines Sipura 1001's as the ATA, DTMF phones on the end. It has a simple extension.conf User dials - runs DeadAGI(Areskicc.php) User goes on to enter PIN, phone number and then is connected (subject to credit and b-number being availible) The only difference between this test system and the production system is the number of lines. I need it to be able to run 80 extensions and therefor 80 lines (presented by SIP) How large should the processor, memory etc be - could anyone suggest a Dell / similar system that would be good for our needs. I don't need any zaptel hardware, as the places this is going to (its intended to be movable - not mobile per-se but movable) will only have outside internet connections, a local SIP provider is helping us which is why its SIP both sides. Thank you for your time on this matter David ___ 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] Hardware dimesioning issues
I sent this earlier today. I didn't see my copy of the mail arrive back. Does anyone know if I am supposed to get back any of my posts or is there a setting I need to change. If it has been reflected properly this morning, please accept my applogies for the re-send. David -- Hello I am in the process of putting together a short term calling card solution that is rapidly deployable for charity events, and would apreciate some guidence on hardware dimensioning for the solution I have a test system running on an old P3 laptop, so in principle the solution works : It is configured as follows: Latest CVS of asterisk (well as of about 3 weeks ago) AreskiCC as the card solution Latest RPM of PostgreSQL Latest RPM of apache Latest RPM of php / pgphp 4 SIP accounts for the phones 1 SIP account with 4 concurrent calls for the lines Sipura 1001's as the ATA, DTMF phones on the end. It has a simple extension.conf User dials - runs DeadAGI(Areskicc.php) User goes on to enter PIN, phone number and then is connected (subject to credit and b-number being availible) The only difference between this test system and the production system is the number of lines. I need it to be able to run 80 extensions and therefor 80 lines (presented by SIP) How large should the processor, memory etc be - could anyone suggest a Dell / similar system that would be good for our needs. I don't need any zaptel hardware, as the places this is going to (its intended to be movable - not mobile per-se but movable) will only have outside internet connections, a local SIP provider is helping us which is why its SIP both sides. Thank you for your time on this matter David ___ 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] Multiplexing (or what ever the term is) FXO ports into a Trunk
Hi all, For an event we are doing, we have been donated several analogue PSTN lines and an 8 port FXO bridge. On the bridge, we have set up each of the ports to work on the SIP protocol, and have referenced them, line1, line2, line3 etc for their username / password. I have placed the config in sip.conf, and they all work fine, inbound and out - for testing anyway! How do I get asterisk, to treat these 8 lines as one 8 call limit trunk? From a users perspective, all he/she needs to dial is 9 (where x's the number) to get any of the 8 outside lines? Sure I could hardcode somthing in each part of the extensions.conf, but if this trial is sucsessful, the number of lines may increase, and it would be nice to define the array once as it were. (I am aware that most of my troubles would go away if I used a more intelligent termination such as ISDN, but for several issues, its not possible) Thank you for your time on this matter. David ___ 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] Multiplexing (or what ever the term is) FXO ports into a Trunk
Hi all, For an event we are doing, we have been donated several analogue PSTN lines and an 8 port FXO bridge. On the bridge, we have set up each of the ports to work on the SIP protocol, and have referenced them, line1, line2, line3 etc for their username / password. I have placed the config in sip.conf, and they all work fine, inbound and out - for testing anyway! How do I get asterisk, to treat these 8 lines as one 8 call limit trunk? From a users perspective, all he/she needs to dial is 9 (where x's the number) to get any of the 8 outside lines? Sure I could hardcode somthing in each part of the extensions.conf, but if this trial is sucsessful, the number of lines may increase, and it would be nice to define the array once as it were. (I am aware that most of my troubles would go away if I used a more intelligent termination such as ISDN, but for several issues, its not possible) Thank you for your time on this matter. David ___ 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] Multiplexing (or what ever the term is) FXOports into a Trunk
Steve, I take it this also works for SIP? Regards David My appologies to the list, I did not realise that the first attempt earlier today hit the list. On Apr 5, 2005 7:38 PM, Steve Mann [EMAIL PROTECTED] wrote: In the zapata.conf where you define your channels, you would also define them as part of a group. Then in your dial plan, when you execute the dial command, you would pass it the ZAP/group_name This will tell the dial command to use the first available channel within the group you have defined. see: http://www.voip-info.org/tiki-index.php?page=Channels%20and%20Groups for more info. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David John Walsh Sent: Tuesday, April 05, 2005 1:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Multiplexing (or what ever the term is) FXOports into a Trunk Hi all, For an event we are doing, we have been donated several analogue PSTN lines and an 8 port FXO bridge. On the bridge, we have set up each of the ports to work on the SIP protocol, and have referenced them, line1, line2, line3 etc for their username / password. I have placed the config in sip.conf, and they all work fine, inbound and out - for testing anyway! How do I get asterisk, to treat these 8 lines as one 8 call limit trunk? From a users perspective, all he/she needs to dial is 9 (where x's the number) to get any of the 8 outside lines? Sure I could hardcode somthing in each part of the extensions.conf, but if this trial is sucsessful, the number of lines may increase, and it would be nice to define the array once as it were. (I am aware that most of my troubles would go away if I used a more intelligent termination such as ISDN, but for several issues, its not possible) Thank you for your time on this matter. David ___ 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
Re: [Asterisk-Users] asterisk sounds
Dov, If anyone responds to your request privately, I'd apreciate it if you were to forward it to me, as I need to translate them into several european launguages. Regards David On Apr 5, 2005 6:24 PM, Dov Bigio [EMAIL PROTECTED] wrote: Hello all, I am looking for a list of all available sound files for asterisk and a transcription of their content, so that I can have someone translate them into portuguese. Does anybody have a list of these files? Thank you Dov ___ 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
Re: [Asterisk-Users] Detecting when a called mobile is not reachable?
I guess I should have added that this is based on the European, and specifically UK model, but I would have expected it to have been deemed best practice by most operators. On Apr 4, 2005 4:04 AM, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Rod Bacon wrote: This is quite interesting. I tested calls to 2 mobiles that I knew were off, and not diverted to voicemail. 1 with Telstra, the other with vodafone (I'm in Australia). Via ISDN, both calls were shown as unanswered by asterisk. When the calls went to voicemail, the call was deemed to be answered. Via analogue circuits, the call is shown as answered, no matter what. That's what I would expect. ___ 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
Re: [Asterisk-Users] Authenticating username
Nabeel, Could you expand on your comments, or provide a link / paste in a sample extensions.conf to show how this would be set up? David On Apr 4, 2005 12:57 AM, Nabeel Jafferali [EMAIL PROTECTED] wrote: Dial(SIP/904)calls whoever logged on as john. You could define a variable in extensions.conf. Nabeel ___ 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
Re: [Asterisk-Users] Detecting when a called mobile is not reachable?
This is traditional accross the mobile / cell providers, and there is no real way around it. Background : The only way to ensure that a mobile is truly there is to page the mobile, normally based on the Mobile Switching Centre (MSC) coverage area, and thats after looking up on the subscirbers HLR, its a lot of signalling for a call not to connect, and a cost to the operator. With the rate that mobile operators charge the A party for the call, they get a percentage of the call from the originating operator, so they get cash as soon as it connects, and therefor its in their interest to connect that call, even if its to an announcement shelf. Its one of the reasons they invented voicemail If there is a way around it, don't shout it too loudly David On Apr 3, 2005 8:56 PM, Ian Hailey [EMAIL PROTECTED] wrote: Hello all, I was hoping to be able to call a mobile and if it is un-reachable for whatever reason (e.g. switched off) then I was expecting an unobtainable response that would be detected in Asterisk. It seems that the operator (Virgin in UK) imedately completes the call and plays an automated message before clearing the call. Does anyone know if there a way of avoiding the call completion for mobiles? I have noticed that Sipgate charge for a calls to an unavailable mobile regardless. Thanks. Ian. ___ 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
Re: [Asterisk-Users] How to number extensions - Which way is best?
Here's a thought. A user is dialing a local number (local for them) but accidently dials it 1+.. Is it LD? No, quite simply each of the prefix digits in the local exchange, would route back to the same routing case, which would be setup with the same charging record, effectively nulling the additional (non required) prefix it would be really nice, if asterisk where possible kept the same naming conventions that have been used in traditional telephony since its inception. [For clarifies sake this previous sentence was totally tongue in cheek, with a tad of truth with no offense intended] regards david ___ 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 number extensions - Which way is best?
On 8 Feb 2005, at 22:54, Mike Dent wrote: Phone numbers beginning with a '1'? Surely not, they should all start with a 0 :) Mike It depends on the country! At the end of the day, as long as the string is decipherable within the data transcript within the switch, ending in an exgress route the dialed digits are valid. david ___ 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] Sipua SPA-2000 and liong delay after dialling number
The delay is a time out. The SPA does not know how many numbers it is expecting before it has a complete number for your system. The invite message is sent as a single message to asterisk containing the whole number string, as apposed to each number individually. In simple terms you have 2 options at your disposal : a) encorage users to adopt pressing gate / pound / hash (the noughts and crosses board above 9 on the keypad - i cant belive this keyboard doesn't have the symbol ;) at the end of the last digit - this in the sipura (like 99% of telephony devices) is treated as a send / termination / enter instruction and sends the instruction (invite message) to asterisk immediatly Note this only applies if your using a touch-tone / dtmf (dual-tone multi-frequency) enabled hand set. b) edit the dial plan of the sipura, to instruct the device of your dial plan, so that it understands how your system is configured. It is sensitve enough to understand that numbers like 999 / 112 / 911 are only 3 digits when national dialing is a greater length. For assitance with that google for spa-2000 user guide, which contains examples or contact me with further information of your set up Hope this helps. david On 28 Jan 2005, at 11:14, Remco Barende wrote: When I use an analog phone connected to a Sipura SPA-2000 it takes about 3-4 seconds before the number is actually dialled. Very annoying especially if you are connecting an intercom to it. Can I change this behaviour and do I need to look at * config or the config of the SPA-2000? 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 ___ 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: Interesting bellster issue
Surely no other route would be tried in this instance, for as far as all devices are concerned the A party and B party were connected correctly, albeit in this instance to an announcement shelf device. I agree that the A party has a right to be annoyed at the loss of credit, but this has been tradition within telco's for as long as i can remember, as a call channel costs significantly more bandwidth than signaling The only time you don't lose credit (or get billed in traditional terms) is when the announcement shelf is contained within the same network as the A party. Why do you think that providers tend to offer free voicemail, to ensure every call is connected and further more get the call in the other direction It is however an interesting way of accruing free credits on the network. Food for thought David On 26 Jan 2005, at 08:30, Samuel Tardieu wrote: dhh == dhickman [EMAIL PROTECTED] writes: dhh When I make a call, bellster anounces that I have no credits and dhh says goodbye, but it still routes the call. I just noticed another interesting problem: I checked that using Congestion I can appropriately reject an incoming bellster call and that another route is used (on extension +331, France, Paris). However, the second route tried by bellster ended up with This is 9:25 local time, calls are only permitted from ... to It means that the remote asterisk accepted the call to play the message, instead of using Congestion to use another route or fail. I lost one credit without having the call placed, but what is more important is that no other route has been tried, and that my PBX thinks that the call succeedeed and will not try an alternative route such a Zap line. The problematic route is 179. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ 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
Re: [Asterisk-Users] Mediatrix voip gateway 1124 and 1204 in UK setting
Peter One thing to consider if you only have 3 PSTN lines is the Sipura SPA-3000 (you would need 3 of them, one for each line) We have 2 PSTN lines at our scout campsite, and they work very well, as well as providing a simple power outage solution. They retail about £80 + the VAT I can supply more information once you have looked at the devices. Regards David On 24 Jan 2005, at 11:20, Peter Hoppe wrote: Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Only very rarely does our call volume exceed three simultaneous connections (inside to inside plus inside to outside). We have looked into the issue of connecting the phones and the outside lines to the system. For the fxo connectivity we want to stick with the three PSTN lines, because they worked for us and we don't see a need to upgrade to ISDN. The asterisk system will be also connected to the internet anyway so we can perform VOIP calls. For the fxs connectivity we want to re-use the old telephone wiring and provide standard two-wire telephones. Putting in IP phones would mean a massive installation effort, as we would have to put an entire new computer network in place - plus many IP phones constantly connected to mains, plus admin headaches, plus security issues and so on. The two wire solution seems the best solution for our setting. We have looked into using a channel bank for the analog conectivity, and we are currently in contact with Carrier Access to purchase a new Adit 600 unit with space for 48 extensions. We cannot provide fxo connectivity via the channel bank because the fxo card from CA seems not to be EU approved. One downside of the channel bank is that we need a special T1 card for it to operate with the asterisk pbx. Also, channel banks seems to be a particular US concept, so we would have difficulties to get replacement parts, if something breaks. Recently I heard of the alternative solution of a voip gateway, and the particular units I have seen are the Mediatrix 1124 for fxs connection and the Mediatrix 1204 for the fxo connection. Both units support the SIP protocol, so it should be possible to connect them to the asterisk PC via standard network connection. Mediatrix seems to have resellers in Europe as well, so it might be possible that their devices are Europe approved as well. Question: * Does anyone have any experience with these units in a UK setting? * For the 1124: Does it work with standard UK two wire phones? Are there impedance problems (especially concerning echo problems)? Is the audio quality sufficient? Are they transparent to the asterisk system, i.e. does each fxs port look like a separate IP phone to the asterisk system? * For the 1204: Would it be approved for connection into the UK PSTN (The prospectus from Mediatrix didn't say anything about regulatory approvals)? Can they initiate outside calls / receive incoming calls or are there problems (signalling compatible with UK PSTN)? Are they transparent to the asterisk system, i.e.does each fxo port look like a separate IP phone to the asterisk system? I do realize that these questions are quite broad, but do appreciate any info. Thank you very much for your consideration. -- dyslexics of the world - untie ! ___ 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
