Re: [Asterisk-Users] delay problem in h323
yes, I agree with you. I verify with a sniffer and asterisk manages RTP flows. The problem is asterisk decode and then code again RTP flows. This function requires 5-7% CPU On my test-box (Linux rh 7.3 on P3 600 GHz). This solution don't scale without dedicated HW, I think! Another problem is codec supported: ok for G.711, G.729. I don't know for GSM BUT: what about video codec? what about proprietary codec or ciphered codec? Do you have any suggestion on how I can manage this with asterisk? I'm very interested into asterisk as sip-to-h323 translator. Thanks Andrea Quoting Steven Thomas <[EMAIL PROTECTED]>: > > > > > > The only way I was able to solve my delay issue with Chan_oh323 was to > switch to Chan_h323. > > Chan_oh323 caused a similar 3 -4 sec delay on one way of the conversation. > Checking the CPU stats on asterisk during the call - confirms that the RTP > stream was somehow routing through asterisk - not sure why! > > > > Regards, > > Steven Thomas > > > > > > > andrea <[EMAIL PROTECTED]> > > Sent by: To: > [EMAIL PROTECTED] > [EMAIL PROTECTED]cc: > > .digium.com Subject: Re: > [Asterisk-Users] delay problem in h323 > > > > > 10-09-03 12:45 AM > > Please respond to > > asterisk-users > > > > > > > Hi all, > > is it possible to disable RTP routing through asterisk? RTP routing is a > very nice feature but, I think itâs important also to disable it in some > cases (e. g. in a LAN). > Do you have any suggestion? > > Andrea > > Rattana BIV wrote: > > > Hi, > > > > I have a delay between two H323. > > > > Netmeeting1 - || > > | gnuGK | --- [asterisk-oh323] > > | Asterisk | > > Netmeeting2 --|| > > > > Netmmeting 1 call Netmeeting 2. When Netmmeting 1 speak Netmeeting 2 > > receive the voice without delay. But in the other way I have 3 secondes > > delay. > > In oh323.conf I set jittermin and jittermax to 20, the ipTos=lowdelay. > > I try to find where I can delete the delay. > > Does anyone have a tip ? > > > > > > Best Regards > > Rattana > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > Ë^®+$RÇ«²f¢)à+- Ë^®+$RÇ«²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«- z¸¬ë ® --- This mail sent through CSP Webmail System ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets
Hi all, I'm interested in using asterisk WITHOUT codec support: I work in a LAN, with no bandwidth, delay, ... problems; I use a Cisco GW as PSTN interface and when I use asterisk the overall delay is to high and the quality drops. In particular, I'm interested in using asterisk as h323 to sip translator (and viceversa). do you have any suggestion? thanks Andres Quoting Mike Ciholas <[EMAIL PROTECTED]>: > > On Tue, 9 Sep 2003, Eric Wieling wrote: > > > Transcoding would be required for access to ANY of the asterisk > > sound files, voicemail and PSTN via Zap interfaces. > > If you are using G711 ulaw from the SIP phones, and that is what > you are getting from the T1 PSTN link, would * have to transcode > that? Is there more to it than digital to digital copy? Perhaps > echo canceling? > > Can we also store sound files in ulaw? I know that takes more > space, but perhaps it is less CPU work to move the bits around > than to codec them. > > -- > Mike Ciholas(812) 476-2721 voice > CIHOLAS Enterprises (812) 476-2881 fax > 2626 Kotter Ave, Unit D [EMAIL PROTECTED] > Evansville, IN 47715http://www.ciholas.com > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > --- This mail sent through CSP Webmail System ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phone notification?
Hello everybody. Is there a way how to setup asterisk to notify caller's phone? Example: I have some numbers and names in asterisk database ( cidname, cidnum), and I want to display the name of person on my phone ( which has no addressbook, but can display chars ) which I am calling to be sure that I have dialed the right number. Thank you for any answer. Andrej ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Next step in extensions.conf after answer the phone in Queue
Thank you for your answer.
But the Dial command has a option 'g' which means that after succes will
proceed next priorities in the dialplan. Is there something also for
Queue() because according to manual there is no option for it. So I am
looking for some other solution.
Andy
Tony Mountifield napsal(a):
> In article <[EMAIL PROTECTED]>,
> <[EMAIL PROTECTED]> wrote:
>
>> Hello everybody.
>>
>> I was looking for the solution but nothing found. I have this in my
>> extensions.conf:
>>
>> exten => 233,1,SetAccount(queue1)
>> exten => 233,2,Queue(queue1|rn)
>> exten => 233,3,NoOp(${QUEUESTATUS})
>> exten => 233,4,NoOp(${DIALSTATUS})
>>
>>
>> But when the call is placed in the queue and somebody answer it, it will
>> throw an error:
>> == Spawn extension (default, 211, 4) exited non-zero on
>> 'Local/[EMAIL PROTECTED],2'
>>
>> And no other command in extensions is executed.
>> Any suggestions?
>>
>
> Queue() is like Dial(), in that if it succeeds in connecting to someone,
> it will not return to the next priority in the dialplan. However, if you
> define an 'h' extension, that will get executed when the call is complete:
>
> exten => 233,1,SetAccount(queue1)
> exten => 233,2,Queue(queue1|rn)
> exten => 233,3,NoOp(${QUEUESTATUS})
> exten => 233,4,NoOp(${DIALSTATUS})
>
> exten => h,1,NoOp(${QUEUESTATUS})
> exten => h,2,NoOp(${DIALSTATUS})
>
> Cheers
> Tony
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Next step in extensions.conf after answer the phone in Queue
I want to log in database some info ( total agents logged in, busy
agents, time ... ). I have some variables and checking them.
Let me explain it from beginning:
Somebody call the queue and everyone is busy, i need to play to caller
that everyone is busy and he should call later, and log this situation.
So I have to variables. Total agents available (TQC) to pick up and busy
agents (BQC). If everyone is busy than this to variables matches and the
caller has to wait. This I need to log with this two numbers.
If BQC != TQC I need just log the numbers nothing else.
Hopefully it is clear to you.
Maybe there is another solution how to do that.
Btw. I am putting this stats in MySQL database.
Andy
Al Baker napsal(a):
> Why would you want a "channel to continue" after the caller has hung up.
> I clearly am missing something here because I can't see what good that
> would be. What do people do with this "Continued Channel" ?
> What is is used for ? How Does having it help you ? ???
>
> Atis Lezdins wrote:
>
>> Queue will continue if called person hangs up (and there's no option).
>> If caller hangs up, call goes to h extension in same context. Just the
>> same way as Dial with 'g'. There's a change in 1.6 that allows called
>> channel to continue if caller hangs up, so probably something like
>> this could be applied also to Queue (or was that actually working with
>> using Local channels?).
>>
>> Regards,
>> Atis
>>
>> On Wed, Apr 23, 2008 at 7:13 PM, AnDY <[EMAIL PROTECTED]> wrote:
>>
>>
>>> Thank you for your answer.
>>> But the Dial command has a option 'g' which means that after succes will
>>> proceed next priorities in the dialplan. Is there something also for
>>> Queue() because according to manual there is no option for it. So I am
>>> looking for some other solution.
>>>
>>> Andy
>>>
>>> Tony Mountifield napsal(a):
>>>
>>>
>>>
>>>
>>>> In article <[EMAIL PROTECTED]>,
>>>>
>>>>
>>> > <[EMAIL PROTECTED]> wrote:
>>> >
>>> >> Hello everybody.
>>> >>
>>> >> I was looking for the solution but nothing found. I have this in my
>>> >> extensions.conf:
>>> >>
>>> >> exten => 233,1,SetAccount(queue1)
>>> >> exten => 233,2,Queue(queue1|rn)
>>> >> exten => 233,3,NoOp(${QUEUESTATUS})
>>> >> exten => 233,4,NoOp(${DIALSTATUS})
>>> >>
>>> >>
>>> >> But when the call is placed in the queue and somebody answer it, it will
>>> >> throw an error:
>>> >> == Spawn extension (default, 211, 4) exited non-zero on
>>> >> 'Local/[EMAIL PROTECTED],2'
>>> >>
>>> >> And no other command in extensions is executed.
>>> >> Any suggestions?
>>> >>
>>> >
>>> > Queue() is like Dial(), in that if it succeeds in connecting to someone,
>>> > it will not return to the next priority in the dialplan. However, if you
>>> > define an 'h' extension, that will get executed when the call is
>>> complete:
>>> >
>>> > exten => 233,1,SetAccount(queue1)
>>> > exten => 233,2,Queue(queue1|rn)
>>> > exten => 233,3,NoOp(${QUEUESTATUS})
>>> > exten => 233,4,NoOp(${DIALSTATUS})
>>> >
>>> > exten => h,1,NoOp(${QUEUESTATUS})
>>> > exten => h,2,NoOp(${DIALSTATUS})
>>> >
>>> > Cheers
>>> > Tony
>>> >
>>>
>>>
>>>
>>>
>>> ___
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>>
>>
>>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.17 and ExtensionStatus
Hello everybody. I have installed new Ubuntu 8.10 and distro package asterisk. Everything is ok but I need to catch events from manager api, but there is not sending any ExtensionStatus package. I have somewhere else Ubuntu 7.04 installed with asterisk version 1.4.10 and there is everything ok. I have setup call-limit, limitpeer... and everything what was in documentation but nothing helps. Can somebody help me? Thanks a lot! Andy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100p on VIA EPIA-V problems
DScale=0 PME- 00:11.0 ISA bridge: VIA Technologies, Inc. VT8231 [PCI-to-ISA Bridge] (rev 10) Subsystem: VIA Technologies, Inc.: Unknown device aa03 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping+ SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- 00:11.1 IDE interface: VIA Technologies, Inc. VT82C586/B/686A/B PIPC Bus Master IDE (rev 06) (prog-if 8a [Master SecP PriP]) Subsystem: VIA Technologies, Inc.: Unknown device aa03 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- SERR- 00:11.2 USB Controller: VIA Technologies, Inc. USB (rev 1e) (prog-if 00 [UHCI]) Subsystem: VIA Technologies, Inc.: Unknown device aa03 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- 00:11.3 USB Controller: VIA Technologies, Inc. USB (rev 1e) (prog-if 00 [UHCI]) Subsystem: VIA Technologies, Inc. (Wrong ID) USB Controller Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- 00:11.4 Bridge: VIA Technologies, Inc. VT8235 ACPI (rev 10) Subsystem: VIA Technologies, Inc. VT8235 ACPI Control: I/O- Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- SERR- 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT82C686 AC97 Audio Controller (rev 40) Subsystem: VIA Technologies, Inc.: Unknown device aa03 Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- 00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev 51) Subsystem: VIA Technologies, Inc. VT6102 [Rhine II] Embeded Ethernet Controller on VT8235 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- 00:14.0 Communication controller: Tiger Jet Network Inc. Intel 537 Subsystem: Unknown device 8085:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- 01:00.0 VGA compatible controller: Trident Microsystems CyberBlade/i1 (rev 6a) (prog-if 00 [VGA]) Subsystem: Trident Microsystems CyberBlade/i1 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- SERR- [disabled] [size=64K] Capabilities: [80] AGP version 2.0 Status: RQ=32 SBA+ 64bit- FW- Rate=x1,x2,x4 Command: RQ=0 SBA- AGP- 64bit- FW- Rate= Capabilities: [90] Power Management version 1 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Here's the contents of /proc/interrupts cat /proc/interrupts CPU0 0: 69499742 XT-PIC timer 1: 16980 XT-PIC i8042 2: 0 XT-PIC cascade 3: 0 XT-PIC uhci_hcd, uhci_hcd 5: 67957058 XT-PIC wcfxo 8: 1 XT-PIC rtc 10:1044947 XT-PIC via82cxxx 11: 562610 XT-PIC eth0 12: 161145 XT-PIC i8042 14: 190384 XT-PIC ide0 NMI: 0 ERR: 87 Thanks, Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: X100p on VIA EPIA-V problems
I paused myself when I saw this. The generic /etc/init.d/zaptel (that you get if you do make config) tries to load wct4xxp, wct1xxp, wcfxo, wcfxs, and wcusb Paring down the list to just wcfxo generates exactly the same problems. Cheers, Andy. Message: 12 Date: Sun, 19 Sep 2004 09:32:37 -0500 From: "Lyle Giese" <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] X100p on VIA EPIA-V problems To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Why is wsusb loading? The X101P uses the wcfxo module. Lyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need 25-50 Linksys boxes
Anyone care to quote on 25 Linksys PAP2-NA units unlocked can email me direct. Straight forward sale best price new equip etc etc... I am a buyer located in the U.S.Need someone with stock that can ship right away. Will want 25 more in less than a week. email [EMAIL PROTECTED]Andy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question on soft phones with SIP and *
Tielman, You can take a look at the quick and dirty guide I'm slowly putting together if you like... http://www.automated.it/guidetoasterisk.htm I'd appreciate any feedback you have on it.. and if it helped Andy *** REPLY SEPARATOR *** On 06/06/2003 at 14:17 Tielman Koekemoer wrote: >I am new to the Telephony world and am trying to get a basic idea of how >things work. > >Can I use Asterisk to connect two soft phones on a LAN to communicate >with one another without any additional hardware (besides the >sound-card)? > >If this is possible, I seem to be doing something wrong. I have >installed a version of X-Lite and can't seem to get it to ring on my Win >PC. > >Any pointers to docs that can give me an idea (except for the "handbook" >which I'm reading) would be greatly appreciated. > >TIA > >Tielman > > >/* Tielman Koekemoer > Unix and Network Administrator at Vista University > Tel: 012-352 4093 > Cel: 083-445 0019 >*/ > > >_ >Content and Virus scanned > by Inflex and Mcafee > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question on soft phones with SIP and *
On 06/06/2003 at 17:36 Patrick wrote: > >Excellent stuff Andy. It was quite a disappointment that the document >stopped before explaining ..errr everything :) Look forward to learn how >to setup one-way conference and music on hold. Thanks for the guide so >far. > >Regards, >Patrick Glad it was of use Patrick... I'll try to get some more done soon... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small office
Dante, IP is this instance does *not* mean IP as in TCP/IP ... Here are some I think it could mean.. Idiot Proof Iggy Pop Inky Plumbs Ingrid Peterson Impersonated Pope Iron Peanuts I [don't have any] Protocols [installed] Incomplete Pigmy Ok, so I'm just messing - but this is NOT a VoIP phone... it's just a standard jobbie... Andy On 06/06/2003 at 17:09 Dante Alzamora wrote: >What is the best cost effective solution for a small office: >I need 3 FXS & 2 FXO. > >Can I hookup a TDM400P and 2 X100P on the same computer? > >Also, I saw some IP phones for $25.99 >http://www.wosmile.com/cgi-bin/view_store_item.cgi?pid=4424&sid=5&category=1 >Can I use them with asterisk? will they be able to do the same as the >TDM400P? >I read that to run the conference app Meetme you needed a Zaptel driver. > >Thanks, > >Dante > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the going rate for the Snom 100 in the UK?
Nathan, Get in touch with www.provu.co.uk ask to speak to Tim, and tell him you heard from me (Andy Powell) that they had a deal running where you could get Snom 100's for 140 gbp... HTH Andy *** REPLY SEPARATOR *** On 29/05/2003 at 12:44 nathan wrote: >Hi All, > >What is the going rate for the Snom 100 in the UK? I've found >a couple of suppliers with prices around the £170 (exc vat) mark. > >Regards, >Nathan. > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI - BT Easicom 1000
Hi Folks, If anyone is interested I now have a BT Easicom 1000 working with *. Some initial problems but they are sorted. Everything seems to work ok with the exception of the vm script. At the moment I basically have to decline the download for it to work.. The really good news is that I guess these phones aren't locked. Nice big screen and there's a pullout keyboard too :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detect hangup on unanswered POTS call
You could always put a wait in before the answer, ie exten => s,1,Dial(SIP/analog1&SIP/analog2,20) exten => s,2,Wait(20) exten => s,3,Answer exten => s,4,Voicemail(u1234) exten => s,5,Hangup HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer Problem
Sorry, I might be being stupid, but I don't see what the problem is. Following your example, 1. Secretary calls someone for the Boss 2. Other caller answers, Secretary asks other end to wait. 3. Secretary presses the flash button (or recall or whatever it's called on the phone) 4. Secretary dial boss, tells boss that caller is on the line 5. Secretary hangs up, boss has caller. Andy *** REPLY SEPARATOR *** On 04/06/2003 at 16:11 Surajee Ratnayake wrote: >yes, u are quite right, you can find this feature in almost every pbx now. > >We are also wondering whether, presently some one is implementing this >feature or not, if no body is doing that, we can >start on that > >Surajee > > > - Original Message - > From: George Lin > To: [EMAIL PROTECTED] > Sent: Wednesday, June 04, 2003 3:36 AM > Subject: RE: [Asterisk-Users] Call Transfer Problem > > > so, What should the call initiator do if s/he wants to transfer the call >initiated by himself/herself, by using flash keypad or what else ? > > I can see such application can be used in some big office, where the >BOSS always asks the secretary to make the call, once the call is >connected, then the secretary can trasfer the call to the BOSS. in order >to let the BOSS talk on the phone. am I right ?? > > Please let me know once the feature is implemented. > > George Lin >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Surajee >Ratnayake >Sent: Monday, June 02, 2003 1:05 AM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Call Transfer Problem > > >U get the following output when u execute the "show application Dial" >command in the Asterisk prompt, > > > -= Info about application 'Dial' =- > >[Synopsis]: > Place an call and connect to the current channel > >[Description]: > >Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL]): >Requests one or more channels and places specified outgoing calls on >them. >As soon as a channel answers, the Dial app will answer the >originating >channel (if it needs to be answered) and will bridge a call with the >channel >which first answered. All other calls placed by the Dial app will be >hunp up >f a timeout is not specified, the Dial application will wait >indefinitely >until either one of the called channels answers, the user hangs up, >or all >channels return busy or error. In general, the dialler will return 0 >if it >was unable to place the call, or the timeout expired. However, >if all >channels were busy, and there exists an extension with priority n+101 >(where >n is the priority of the dialler instance), then it will be the > next >executed extension (this allows you to setup different behavior on >busy from >no-answer). > This application returns -1 if the originating channel hangs up, or >if the >call is bridged and either of the parties in the bridge terminate the >call. >The option string may contain zero or more of the following characters: > 't' -- allow the called user transfer the calling user > 'T' -- to allow the calling user to transfer the call. > 'r' -- indicate ringing to the calling party, pass no audio >until answered. > 'm' -- provide hold music to the calling party until answered. > 'd' -- data-quality (modem) call (minimum delay). > 'c' -- clear-channel data call (PRI-PRI only). > 'H' -- allow caller to hang up by hitting *. > 'C' -- reset call detail record for this call. > 'P[(x)]' -- privacy mode, using 'x' as database if provided. > In addition to transferring the call, a call may be parked and then >picked >up by another user. > The optionnal URL will be sent to the called party if the channel >supports >it. > > > >Surajee > > > - Original Message - > From: George Lin > To: [EMAIL PROTECTED] > Sent: Monday, June 02, 2003 1:11 PM > Subject: FW: [Asterisk-Users] Call Transfer Problem > > > Hi, > > > > Which document describes the Dial with T option ? Could you let >me know or email it to me. > > > > Thanks, > > > > George Lin > > > > -Original Message- > From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Surajee >Ratnayake >
Re: [Asterisk-Users] Getting netmeeting to work with Asterisk
>I've played with modifying the extensions.conf and h323.conf but don't >have things right. I keep getting a message on the >console: > >ERROR[376849]: File chan_h323.c, Line 974 (setup_incoming_call): Call from >user 'Simon' rejected due to no default context > >However I am unsure what this really means and how to configure the >extensions to allow incoming and outgoing calls to the netmeeting client. I'm only guessing here, since I don;t have h323 set up (tho seeing as it seems quite easy from your description I may well give it a go). In your h323.conf I assume much like a sip.conf entry you have to define a phone and give it a context e.g. in sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.11.190 context=sip note the last line context=sip. Then in my extenstions.conf file I have a section: [sip] which matches the context of the phone, this is where my extensions are defined. Do you have en entry for your netmeeting client in the h323.conf? and a corresponding entry in extensions.conf? Again this could all be guff since I don;t have h323 setup, but it strikes me as a fairly logical error if you have no matching context in your extensions.conf... HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on hold, Call Parking, etc
Ok, so are you pressing # then hearing the word 'transfer' and dialing the exten to transfer to? Andy *** REPLY SEPARATOR *** On 14/05/2003 at 11:34 Derek Beaumont wrote: >I am using a regular analog phone. > > > >Derek, > >What are you using to place the call? Snom Phone? Cisco or soft phone? > >Andy > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Back
On 09/06/2003 at 18:24 Alex Lopez wrote:
>
>I am at the point where it all works except I do not know the variables
>in extension.conf {$CALLERID} is the whole strings including name!! I
>want just the number.
For just the number use
${CALLERIDNUM}
if you want the name use
${CALLERIDNAME}
Andy
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie : i try and test to use asterisk
Hi, You need to change your settings in X-lite: >Display name : roseau >user name : 1000 <--- this is wrong! >authorization user : >Password : >Domain/Realme : 192.168.0.2 >SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty to: user name : roseau (That should match the definition in sip.conf so on the other Pc you would use bambou) authorization user : Password : Domain/Realme : SIP Proxy : :5060 HTH Andy *** REPLY SEPARATOR *** On 11/06/2003 at 11:11 Hervé THIBAUD wrote: >I try to use X-lite with asterisk on intranet > >In sip.conf i have > >[general] >port = 5060 >bindaddr = 0.0.0.0 >context = default > >[roseau] >type=friend >host=dynamic >dtmfmode=inband >context=sip > >[bambou] >type=friend >host=dynamic >dtmfmode=inband >context=sip > >and in extensions.conf > >[sip] >exten => 1000,1,Dial,SIP/roseau >exten => 2000,1,Dial,SIP/bambou > >i use X-Lite on windows >in setup ; > >Display name : roseau >user name : 1000 >authorization user : >Password : >Domain/Realme : 192.168.0.2 >SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty > >i obtain in /var/log/messages when i try to call >[handle_request]: Registration from 'roseau '' failed >for '192.168.0.4' > >Is anybody help me to start please > >regards (and very sorry for my english) > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail notification
I'd like to use either the message waiting light or stutter tone but on searching the archives I found conflicting answers. Everyone seems to agree that you should add mainbox= but some people are saying that it should be added to zapata.conf and others are saying zaptel.conf Can someone who has it working clarify this? If it is zaptel.conf can somone supply a sample.. my zaptel.conf file only consists of fxsks=1 fxoks=2 fxoks=3 loadzone=uk defaultzone=uk and that's it... Thanks in advance Andy *** REPLY SEPARATOR *** On 11/06/2003 at 16:53 Steven Critchfield wrote: >On Wed, 2003-06-11 at 15:16, Derek Beaumont wrote: >> Besides email notification, is there another way to get asterisk notify >> the user that they have a message? >> >> Example: Some analog phones have a blinking light that lets the user >> know that they have a voicemail message. >> Is asterisk capable of doing this? > >Yes, and I know it works on Sip and Zap channels. Check archive for MWI, >for Message waiting indicator. >-- >Steven Critchfield <[EMAIL PROTECTED]> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor application
Hi, I've had a search through the archives and didn't find much. Is anyone using the Monitor application? I have it working but there is a really big drawback. The files are always called the same thing, which means if I make 2 calls one after the other the first recording is lost. I half expected Monitor to use something like ZAP-2-1--in/out.wav for it's filenames but it just uses the channel eg Zap-2-1-in.wav Zap-2-1-out.wav has anyone found a solution to this? Thanks Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor application
Ahh, wonderful thanks... Andy On 12/06/2003 at 13:35 Pertti Pikkarainen wrote: >Check >http://www.loligo.com/asterisk/current/extensions.conf > >and find macro called macro-record-on >There is at least one way described ( author is John Todd ). > > >--Pertti > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail notification
Ok, thanks for the clarification
Shame it still doesn;t work for me :( maybe it only works
with US phones... anyone in Europe got this working?
Andy
*** REPLY SEPARATOR ***
On 11/06/2003 at 21:55 Tilghman Lesher wrote:
>On Wednesday 11 June 2003 19:10, Andy Powell wrote:
>> I'd like to use either the message waiting light or stutter tone but
>> on searching the archives I found conflicting answers.
>>
>> Everyone seems to agree that you should add
>>
>> mainbox=
>>
>> but some people are saying that it should be added to zapata.conf and
>> others are saying zaptel.conf
>>
>> Can someone who has it working clarify this?
>
>zaptel.conf is used for the kernel module. zapata.conf is used for the
>Asterisk program. As MWI is an Asterisk feature, it must therefore be
>placed in zapata.conf.
>
>If you're still not convinced, you can do a grep in the various cvs
>repositories to confirm which is which:
>
>[EMAIL PROTECTED]:/cvs/asterisk# grep -r '"mailbox"' /cvs/zaptel
>[EMAIL PROTECTED]:/cvs/asterisk# grep -r '"mailbox"' channels/chan_zap.c
>} else if (!strcasecmp(v->name, "mailbox")) {
>} else if (!strcasecmp(v->name, "mailbox")) {
>
>-Tilghman
>
>___
>Asterisk-Users mailing list
>[EMAIL PROTECTED]
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk switch => statement
So is that one switch statement per installation or one per context. ie can i have multiple switch statements each one applicable to a different section in extensions.conf Andy On 13/06/2003 at 13:28 Martin Pycko wrote: >The idea of switch is for every box to know what it can reach locally. And >then to do the 'switch' to remote boxes if the called number can't be find >locally. > >Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID forward???
Derek,
exten => 400,1,SetCallerID(${CALLERIDNUM})
You can use
${CALLERID}
${CALLERIDNAME}
${CALLERIDNUM}
Andy
On 13/06/2003 at 16:18 Derek Beaumont wrote:
>I don't understand how or where I would use setcallerid.
>I have tried to do
>exten=>400,1,Setcallerid,asreceived
>but that doesn't seem to work
>
>
>What am I doing wrong?
>
>-Derek
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Whoooaaa!!! Feaky - but in a good way
Ok, this has really freaked me out, but in a good way - sort of.. I've made no changes at all to my system, save messing with ADSI. However this has nothing to do with ADSI. The thing is all of a sudden my DECT phones have started reporting caller id, and not just the number, the name too! They have never done this before in the couple of months that I've had * running. I'm pleased that they have decided to work, but I am confused and concerned as to how and why it suddenly started ... anyone got any ideas? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Whoooaaa!!! Feaky - but in a good way
On 16/06/2003 at 10:26 DUSTIN WILDES wrote: >If this is through your Telco, they may have turned on the Callerid-Name >field along with your number. >I had mine turn on the Callerid-Name field for us. No, not from my teleco, this is from * via the TDM card to the DECT phones that's why it spooks me... I don't have caller id on my pstn line, since it's a chargable option here in NL and I have no idea if KPN's callerid works with the Digium card. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Test System?
On 17/06/2003 at 10:23 Rushowr wrote: >Is it possible to set up Asterisk without any of the cards? I'm >interested in setting it up for the company I work for, but I would like >to set it up and see how difficult it will be before I start having the >company spend a chunk on equipment. Yes, you can set up asterisk without adding any cards. You could use SIP and download a client such as SJPhone - http://www.sjlabs.com to test with > >Additionally, what phones can be used with Asterisk? we currently use a >NEC Nitsuko phone system with phones, but I have been confused as how to >set up the entire thing I really would like more information on the >setup procedures, including the interface between the telco's lines and >the server system, and how to wire the phone equipment (they're not >going to use the computers to take calls) to the server. Unless you buy some cards the only phones you can use are softphones and hardphones (eg, snom or cisco). If the NEC phones are analog then you can either use a Channel bank (see http://www.digium.com/index.php?menu=developerskit_fxs ) or you could try the TDM400P cards with options of 2 X 4, 2 X 8, 3 X 8 ports.. ( http://www.digium.com/index.php?menu=wildcard_tdm400p ). Unless the company is very small shelling out for a developer kit isn't going to break the bank.. for simple testing you could go for the Asterisk Developer's Kit (TDM) but add another 3 ports on the TDM card, giving you a connection to the PSTN and connectivity for 4 analog phones. As an alternative you could connect your existing PBX to an * box for testing but we'd need to know what interfaces your pbx has... and now I'll stop acting like an advert for Digium... :) HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] festival error
You have looked at festival.conf right? What's your exten line line here's one of mine: exten => 1021,1,Festival(mary had a little lamb) Note the lack of quotes hth Andy *** REPLY SEPARATOR *** On 19/06/2003 at 13:57 Chad Sawyer wrote: >I followed the directions I found in the list to a tee >http://www.marko.net/asterisk/archives/0209/0389.html > >The server starts fine, but when I call the festival extension it gives me >an OID error variable tts_textasterisk > >I have RH7.3 >festival 1.4.2 >speech_tools 1.2.2 >patched it with the festival-1.4.2.diff located in the /usr/src/asterisk/ >folder . When I patched it, the patch was looking for festival to have >been extracted in festival-1.4.2 instead of just "festival". So I did it >that way and patched it. Same error. > >Any ideas, maybe a nudge? I know it has something to do with that patch >adding the variable to festival. The patch says it completed >successfully... > >Thanks > >Chad Sawyer, Manager, Network Administrator >Your Total Communications, LLC > > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [HS] results testing asterisk with ISDN BRI & look for help to understand configuring SIP with asterisk
Hi, >I don't understand what i have to make and set to communicate with >external telephons SIP (Sjphone, X-lite, MS messenger ...) >Must i have a SIP provider subscription, how to integrate this >subscription with asterisk Do you mean internally i.e. Sjphone, X-lite, MS messenger phones on your pc's or other people - out there on the net? You could take a look at my guide - it may help explain things (then again it may not) http://www.automated.it/guidetoasterisk.htm I recently had to move hosting co's, just noticed the one I moved was old!! I've updated it... >I am "swimming" with (english) documentation anglaise You're lucky, I'm English and I have trouble speaking it! HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager interface, again
Ok, is it me or do some of the commands just not work properly? I asked for mailboxstatus and got: Response: Success Message: Mailbox Status Mailbox: 1000 Waiting: 0 which is all well and good, except of course I have 2 messages waiting... which kinda means it only works, if you have 0 messages... (using voicemail not voicemail2) Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Manager interface, again
Ok, like a I'll answer my own message: If in your voicemail.conf you have * configured to the send message in an email you will NOT get a stutter dialtone or any MWI light you may have. I've just removed my email address from voicemail.conf.. much better like that... HTHSITF Andy *** REPLY SEPARATOR *** On 20/06/2003 at 18:17 Andy Powell wrote: >Ok, > >is it me or do some of the commands just not work properly? I asked for >mailboxstatus >and got: > >Response: Success >Message: Mailbox Status >Mailbox: 1000 >Waiting: 0 > >which is all well and good, except of course I have 2 messages waiting... >which kinda means >it only works, if you have 0 messages... (using voicemail not voicemail2) > >Andy > > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Manager interface, again
>This is odd because I email all my users voicemail out and the ones that >don't clear the voicemail on the phone still get stutter tones. I had to >inform them of what to do, and then mass delete their voicemail to get >the stutter tone to stop. One user had almost 50 messages waiting. I had this initially, but it was due to a 'zombie' message (as pointed out to me by citats... an easy way to double check is to connect to the manager interface and look at the status of a mailbox.Iif it reports something like mailbox: 1000 Response: Success Message: Mailbox Status Mailbox: 1000 Waiting: 1 <--- that is NOT a msg count btw then the user would get a stutter tone.. if Waiting: 0 then they wont... I'd be interested to hear if this is/isn;t the case on your setup HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Manager interface, again
On 20/06/2003 at 14:45 Wade Weppler wrote: >Same here. E-mail and MWI/Stutter tone work fine together. > if that attaching the file or just sending a messages without a file attached..? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Manager interface, again
>> if that attaching the file or just sending a messages without a file >attached..? > >We attach the voicemail, thats how 2 of my users eneded up with 50 or so >messages waiting. They didn't feel the need to listen to or delete the >messages via the phone when they had listened to them via the email >interface. >-- >Steven Critchfield <[EMAIL PROTECTED]> > Mmmm, that's odd... as soon as I add the email address I stop getting the stutter tone and MWI... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asteisk, sip & NAT
>Andy, your update is >http://www.automated.it/guidetoasterisk.htm isn't it ? yes, same place, just added some extra notes in there (they should be obvious) HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference calls on Pingtel Phones
Has anyone been able to get conference calls to work on the Pingtel Phones? I assume this feature works with their implementation, but connected to my asterisk box it doesn't work. The Pingtel phone thinks it is making a second call, but asterisk never sees anything about a second call. Any help would be appreciated. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 100 and GSM codec
>Anybody have the latest word on Snom's development? Last I had heard, >they were still working on compatibility with GSM. Firmware version >1.15e, which is what my Snom 100 automatically updated itself to, does >not work with GSM. > >-Tilghman I'm using 1.15u and it's a little better. Snom are aware of the issue and said that the next firmware should fix it. TBH, I'm doubtful of a new release for the 100's, the rumour mill tells me that they are working on the 105 and that the 100 has been EOL'd Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hardphone
You can try Snom or Cisco... Or get a TDM card and use an analog phone... Andy *** REPLY SEPARATOR *** On 25/06/2003 at 15:44 Chris wrote: >I've got Asterisk up and running nicely using a couple of different >softphones. Audio quality is suffering a bit due to the hardware that I >am working with. So I tried to use a Polycom hardphone but the politics is >enough to give you a headache. Polycom seems to support SIP only if you >buy it thought their vendors. So I'm looking at a Cisco phone. Has >anyone successfully implemented Asterisk with a hardphone? Which one? >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk hardphone
I am using Pingtel phones right now and find that they work well and have some cool features. Got most everything working except conferencing and we're working on that. I love some of the ringtones (i.e. 007 theme, Beavis & Butthead, Homer Simpson Doh! etc) Sincerely, Andy Hester Consero -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of ChrisSent: Wednesday, June 25, 2003 2:45 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk hardphone I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented Asterisk with a hardphone? Which one?
Re: [Asterisk-Users] X100P and PSTN caller id
Mmm... I don;'t know what else to try, I've had callerid turned on here but it doesn't work at all... :( Andy *** REPLY SEPARATOR *** On 26/06/2003 at 13:02 Dan wrote: >There is nobody with an X100P in Europe having this issue related to the >PSTN Caller ID? >Please help! > >Thanks, >Dan > >> > On Wed, 25 Jun 2003 22:37:36 +0300, Dan wrote: >> > >> > >Hi all, >> > > >> > >I have an X100P card which works fine with Asterisk, but I have some >> > >problems with the caller id. >> > >When someone calls me from the PSTN, I see as a callerid my own PSTN >> number. >> > >There is a specific setting for this type of CallerID in order to be >used >> > >with X100P? >> > > >> > >Thank you, >> > >Dan >> > > >> > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - first impressions
I tell you what, just relpy to every message with the word remove rather than actually reading the instructions. *** REPLY SEPARATOR *** On 26/06/2003 at 13:30 cisb wrote: >REMOVE >- Original Message - >From: "Peter Zeltins" <[EMAIL PROTECTED]> >To: <[EMAIL PROTECTED]> >Sent: Thursday, June 26, 2003 3:16 AM >Subject: [Asterisk-Users] Asterisk - first impressions > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P and PSTN caller id
>Well, I can't get it to work with an MD110 PBX over here either (.nl). It >probably should work, but I never found how. Tried several options as >suggested by this list though.. > >-- >From what I can gather the caller id in NL is similar to Denmark, it's just a series of DTMF tones send down the line. The problem of course is that the only notification you get is the DTMF "A" sent to indicate the start of caller id... Arf, arf Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Important: PSTN access-number for Dutch gateway changed
Oliver, can you clarify how the gateways is supposed to be used, I've tried calling the number from a PSTN line, the call is answered and i get dialtone, I then try to dial my iaxtel number and just get told that it's an invalid extension.. the 'error' occurs after dialing 17001 of my iaxtel number... Thanks Andy *** REPLY SEPARATOR *** On 26/06/2003 at 21:55 The Traveller wrote: >Yo all, > >The PSTN access-number for the Dutch IAXTel <-> PSTN-gateway has changed. >The new number is: +31 20 3987 567. Calling from IAXTel to Dutch >toll-free PSTN-numbers is still done in the same way, by calling >"31800". > >Mark: Could you please update your web-sites to reflect this >change? The old number is mentioned on "http://www.gnophone.com/";, >not sure about other places. > > > >Grtz, > > Oliver >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web interface for Asterisk
I tend to disagree and agree on this one. Someone on irc has already asked if there is any sort of interface to allow the creation of voicemail boxes, primarily for use by the HR department to allow them to create them. For those little things you might hand over to other people in a company a gui of some sort would be useful. However as any sort of implementor you'd want to get to the real config files. In all honesty once you start to understand the commands it gets pretty easy All imho Andy *** REPLY SEPARATOR *** On 26/06/2003 at 20:55 Jeremy McNamara wrote: >So why not just open vi and bang out a conf file?How is a newbie >ever not gonna be a newbie with a GUI config? > > >Good luck, your gonna need it, > > >Jeremy McNamara > > > > >Gary wrote: > >>I tend to agree with Steven on this... >> >>If the web form makes it easier for the "newbies" why not, its just >>another option >> >>It could even be expanded to be a dialplan for dunnies (woops, i meant >>dummies:-) interface >> >>Considering all it is, is an interface to write out a .conf file >> >>On Thu, 26 Jun 2003 17:04:28 -0700, Steven P. Donegan wrote: >> >> >> >>>I disagree - for many tasks a GUI would be just fine, for others direct >coding would do the trick. They do not have to be mutually >>>exclusive. >>> >>>-Original Message- >>>From: [EMAIL PROTECTED] >>>[mailto:[EMAIL PROTECTED] Behalf Of Jeremy >>>McNamara >>>Sent: Thursday, June 26, 2003 4:42 PM >>>To: [EMAIL PROTECTED] >>>Subject: Re: [Asterisk-Users] Web interface for Asterisk >>> >>> >>>That GUI is going to dramaticly limit the flexibility of your config. >>>The only way you can make a GUI config work with Asterisk is if you have >>>a very very specific task you want to accomplish, but even then you >>>still will have issues as your requirements change with time. >>> >>>Stick with what the AstGod has bestowed upon us It will save you >>>many headaches. >>> >>> >>>Jeremy McNamara >>> >>> >>> >>> >>> >>>Dylan VanHerpen wrote: >>> >>> >>> >>>>Hi everybody, >>>> >>>>I've been tinkering with a web based interface for Asterisk. I tried >>>>to stick as closely to the current configuration format as possible. >>>>The web interface should help to do things a little easier (sort by >>>>extension, context, do bulk changes). >>>> >>>>www.packetbell.com/asterisk >>>> >>>>Feedback appreciated! >>>> >>>>Dylan. >>>> >>>>___ >>>>Asterisk-Users mailing list >>>>[EMAIL PROTECTED] >>>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>>> >>> >>>___ >>>Asterisk-Users mailing list >>>[EMAIL PROTECTED] >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>>___ >>>Asterisk-Users mailing list >>>[EMAIL PROTECTED] >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >> >>. >> >> >> >>___ >>Asterisk-Users mailing list >>[EMAIL PROTECTED] >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail issue
Dan, The first question is : is your voicemail in the default location or have you moved it to another disk? if you do this you need to update the vm system link in the /var/spool/asterisk directory eg: vm -> /home/asterisk/voicemail/default/ using ln -s vm also make sure * has the privs to write to those directories. HTH Andy On 27/06/2003 at 14:21 Dan wrote: >Hi, > >Nope. > >I have recorded my own busy and unavailable message from the '0' menu of my >voice mailbox. >When someone is redirected to the mailbix, it hears both of them... first >my >recorded message, second the default one. > >I check that on two separate Asterisk boxes. >I have the latest version from the CVS. > >What else can be done? > >Thanks, >Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTEL numbers and example
Hi, During my recent chat with Oliver with regard to the NL IAX<->PSTN gateway one thing has cropped up. Either my IAXTEL number is a bit dodgy or the example configs for IAXTEL numbers are wrong. My IAXTEL number starts 17001, but the examples clearly exclude these numbers: [iaxtel] exten => _17XXNXX,1,Dial(IAX/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1888NXX,1,Dial(IAX/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1877NXX,1,Dial(IAX/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1866NXX,1,Dial(IAX/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1800NXX,1,Dial(IAX/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) Note the first line, 17XX N XX ... and since N = digits 2-9 a 17001 number is un-dialable. Could someone clarify and if required make the appropriate changes to IAXTEL.COM and the sample configs in CVS... Many thanks Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I disable musiconhold for agents
You could create a simple moh class that played a silent mp3 as a very low rate, or even the occasional beepthen just use setmusiconhold, hth Andy On 27/06/2003 at 13:10 Derek Beaumont wrote: >I was playing with the agent application to see if I could get it to >work. >Everything works fine, except that Asterisk plays musiconhold while an >agent is logged in and is not taking a call. Is there a way to disable >the music in this situation? > >Imagine working tech support where you had to listen to hold music when >you >weren't taking a call. Now think of your company's choice of hold >music. >Unless you work for a "cool" company, chances are it's elevator music. > >Any help is appreciated, > >Thanks. > >-Derek > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] modprobe ? for TDM40B
The X100P is modprobe wcfxo The TDM40B is modprobe wcfxs Andy *** REPLY SEPARATOR *** On 27/06/2003 at 16:07 Steven P. Donegan wrote: >What is the module name for the TDM40B - I received my X100P and TDM40B >today (thanks Digium). >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help! Problems talking to upstream switch
Hi, Please let me know if you have any ideas - I am taking wild guesses now Here is the situation: I put in Asterisk for a local customer. I have Fractional T-1 with 12 Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it a chanel bank although it only has 2 outputs) in a CAC unit. The unit also has a router card that runs the data side. My extensions are all SIP phones save a few fax machines. The customer has 7 digit unverified account codes on the trunks for billing purposes. The Problem: As I watch the console, I see calls coming in for exten "73" or "708" or "08" or "730" although most come in correctly (ie "7308"). My carrier has verified numerous times that they are sending 4 digits. I have 40 DID numbers that need to be routed and they are all in the 73xx range. I need to know anything that would cause my box randomly not to hear all 4 digits on occasion. Also, I have had trouble with people who dial out getting a congestion signal mainly on Long Distance numbers. The person would dial the number 4 or 5 times and get congestion then it might go through. Both of these conditions seem to be happening only about 10-20% of the time. What I have done: Moved a T100P card to its own IRQ to prevent problems with interrupts - Did not solve either issue. On the second try, got the carrier to change the way their switch chooses channels for incoming calls to prevent "glare" - This MAY have fixed the outgoing long distance issue as it seems to have gone away( although it doesn't seem logical to me that this would affect only LD) but did not fix incoming calls. Has anyone else had problems getting all of the digits that the Telco sends? Thanks, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help! Problems talking to upstream switch
Thanks for the info... I've answered your questions below. I am not experienced with telecom at this level (yet), but this sounds like really good info to quiz XO's switch tech over. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Sunday, June 29, 2003 9:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help! Problems talking to upstream switch Whats the sound quality like on the calls especially when multiple calls are going? No problems with sound quality save the slight echo on calls over the TDM circuit. On my home system, I had a problem that DTMF was sometimes not correctly recognized and would either dial incorrectly or not at all. It was evedent when dial tone was played that it crackled. Also when multiple calls where running each would start to sound like crap. This was later tracked down to a timing problem. Since you mention using a CAC with 2 ports on it and a router, I'm going to assume you have a ADIT 600. Make sure the Adit is set to take timing from the telco, and then make sure you are set to take your timing from it. Check to make sure each of the T1 ports a:1 and a:2 are set to take timing from a:1 if it is your telco port. This will keep you from slipping and causing potential problems. I haven't looked around inside the CAC yet since it is XO's, not sure if it is an ADIT 600 but it sounds like the same unit. I am set to "0" for timing and "0" for LBO in Zaptel.conf I assume this is correct for * and that I need to verify the a:1/a:2 timing settings in the CAC unit? Come to think of it, there is a way to test this without bringing the T1s down. The Adit 600 has a show performance command, I may be wrong, but I'm sure it was performance, anyways it allows you to see slips and bipolar violations and a couple other stats. This was beneficial for me as the T100P didn't report problems but the Adit did. Can you give me a brief idea of what slips and bipolar violations are? Home this helps. I am glad to find that someone knows more than my little knowledge of the subject! To here the techs talk you'd think that they'd never run into anything like this before. On Sun, 2003-06-29 at 20:59, Andy Hester wrote: > Hi, > Please let me know if you have any ideas - I am taking wild guesses now > Here is the situation: > > I put in Asterisk for a local customer. I have Fractional T-1 with 12 > Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it a > chanel bank although it only has 2 outputs) in a CAC unit. The unit also > has a router card that runs the data side. My extensions are all SIP phones > save a few fax machines. The customer has 7 digit unverified account codes > on the trunks for billing purposes. > > The Problem: > > As I watch the console, I see calls coming in for exten "73" or "708" or > "08" or "730" although most come in correctly (ie "7308"). My carrier has > verified numerous times that they are sending 4 digits. I have 40 DID > numbers that need to be routed and they are all in the 73xx range. I need > to know anything that would cause my box randomly not to hear all 4 digits > on occasion. Also, I have had trouble with people who dial out getting a > congestion signal mainly on Long Distance numbers. The person would dial > the number 4 or 5 times and get congestion then it might go through. Both > of these conditions seem to be happening only about 10-20% of the time. > > What I have done: > > Moved a T100P card to its own IRQ to prevent problems with interrupts - Did > not solve either issue. > > On the second try, got the carrier to change the way their switch chooses > channels for incoming calls to prevent "glare" - This MAY have fixed the > outgoing long distance issue as it seems to have gone away( although it > doesn't seem logical to me that this would affect only LD) but did not fix > incoming calls. > > Has anyone else had problems getting all of the digits that the Telco sends? > > Thanks, > Andy Hester > Consero > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help! Problems talking to upstream switch
Title: [Asterisk-Users] Help! Problems talking to upstream switch This helps alot! I believe that it is an ADIT 600, and I definitely want to adjust those gain settings. I'll ask Martin about the timeout on DNIS, although it would seem that the fact that I have observed the loss of only the 2nd tone, for example, leads me to believe that it is not the timing out. Thanks, Andy Hester Consero -Original Message-From: Tim McQueen [mailto:[EMAIL PROTECTED]On Behalf Of [EMAIL PROTECTED]Sent: Sunday, June 29, 2003 10:11 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Help! Problems talking to upstream switch I'm new to *, but I've dealt with this issue on other switches. It sounds like either you are timing out getting DNIS information from your CO, or you are having trouble hearing the DTMF tones that are pulsed to you during the call setup process. Someone else on the list may know: is it possible to 1) configure the timeout for waiting on DNIS and 2) is it possible to change the Rx gain on the TDM cards? It's my understanding that the circuit is going throught the channel bank, which is acting like a drop-and-insert CSU by forwarding the 12 channels with voice to your * box. You mentioned that this was a Carrier Access channel bank, is it an ADIT 600? There are send and recieve gain settings on the channel bank unit that you might want to play with. HTH, HAND -Tim -Original Message- From: Andy Hester Sent: Sun 6/29/2003 8:59 PM To: [EMAIL PROTECTED] Cc: Subject: [Asterisk-Users] Help! Problems talking to upstream switch Hi, Please let me know if you have any ideas - I am taking wild guesses nowHere is the situation: I put in Asterisk for a local customer. I have Fractional T-1 with 12Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it achanel bank although it only has 2 outputs) in a CAC unit. The unit alsohas a router card that runs the data side. My extensions are all SIP phonessave a few fax machines. The customer has 7 digit unverified account codeson the trunks for billing purposes.The Problem: As I watch the console, I see calls coming in for exten "73" or "708" or"08" or "730" although most come in correctly (ie "7308"). My carrier hasverified numerous times that they are sending 4 digits. I have 40 DIDnumbers that need to be routed and they are all in the 73xx range. I needto know anything that would cause my box randomly not to hear all 4 digitson occasion. Also, I have had trouble with people who dial out getting acongestion signal mainly on Long Distance numbers. The person would dialthe number 4 or 5 times and get congestion then it might go through. Bothof these conditions seem to be happening only about 10-20% of the time.What I have done:Moved a T100P card to its own IRQ to prevent problems with interrupts - Didnot solve either issue.On the second try, got the carrier to change the way their switch chooseschannels for incoming calls to prevent "glare" - This MAY have fixed theoutgoing long distance issue as it seems to have gone away( although itdoesn't seem logical to me that this would affect only LD) but did not fixincoming calls.Has anyone else had problems getting all of the digits that the Telco sends?Thanks,Andy HesterConsero___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users <>
RE: [Asterisk-Users] Newbie, Loaded Asterisk can't figure out manual
John, I agree that the manual in its current state is not particularly comprehensive or cohesive shall we say. Here are my suggestions: 1. Configure Zaptel.conf in /etc 2. Configure Zapata.conf in /etc/asterisk to match your definitions in zaptel 3. Configure your end user devices in their config file as appropriate. 4. Write extensions.conf and voicemail.conf Should be ready to test. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Your Name Sent: Sunday, June 29, 2003 7:56 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie, Loaded Asterisk can't figure out manual Just loaded it yesterday running on TDM400P and X100P. I have also loaded the sample setttings. 1. What's the first thing you guys do? Change .conf files or do it from CLI? 2. Just trying to get it up and running to see if everything works. Do I setup Extensions first? 3. Can I just remove some of the ; within the .conf files to get a simple PBX working? I would appreciate your thoughts and knowledge! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help! Problems talking to upstream switch
Steven, I thought that "1" would mean that my T100P card would set the timing for the line. Is this incorrect? If I am reading this wrong then please set me straight. My carrier has their end set to be the sync source. If I set the timing to "1", won't that conflict? My line is set up esf/b8zs, so does that mean I can ignore all bipolar violations, or just that a certain number are to be expected? Also, shouldn't the switch tech from my carrier be knowledgeable about these things and trying to help me match up to their settings? I appreciate you indulging me this evening. I'm in somewhat of a bad spot with my customer for a variety of reasons, most beyond my control and I am trying to get their problems resolved asap. Thanks again. Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven > Critchfield > Sent: Monday, June 30, 2003 12:04 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Help! Problems talking to upstream switch > > > Use 1 for timing. You actually do have a timing source to sync to. > > As for slips and bipolar violations... > T1s are just high speed serial lines. A sleep is when you loose sync > with the far side and when you see a 1 come across the line, you may not > know which bit it was for. This would be a slip. Bipolar violations are > a part of the signaling, but can also be errors. A T1 alternates the > polarity of the 1 pulse to allow the line to run farther on lower > voltage. Just doing alternating polarity is AMI or Alternate Mark > Inversion. A bipolar violation is when a bit is received as the same > polarity as the last bit received. On an AMI line a bipolar violation is > an error. On a B8ZS, bipolar violations are intentionally inserted into > the line to keep the line from transmitting too many 0's in a row and > contributing to a slip. When set for B8ZS the "error" is somewhat > expected and ignored. > > On Sun, 2003-06-29 at 23:19, Andy Hester wrote: > > Thanks for the info... I've answered your questions below. I am not > > experienced with telecom at this level (yet), but this sounds > like really > > good info to quiz XO's switch tech over. > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of Steven > > Critchfield > > Sent: Sunday, June 29, 2003 9:40 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Help! Problems talking to upstream switch > > > > > > Whats the sound quality like on the calls especially when multiple calls > > are going? > > > > No problems with sound quality save the slight echo on calls > over the TDM > > circuit. > > > > > > On my home system, I had a problem that DTMF was sometimes not correctly > > recognized and would either dial incorrectly or not at all. It was > > evedent when dial tone was played that it crackled. Also when multiple > > calls where running each would start to sound like crap. This was later > > tracked down to a timing problem. > > > > Since you mention using a CAC with 2 ports on it and a router, I'm going > > to assume you have a ADIT 600. Make sure the Adit is set to take timing > > from the telco, and then make sure you are set to take your timing from > > it. Check to make sure each of the T1 ports a:1 and a:2 are set to take > > timing from a:1 if it is your telco port. This will keep you from > > slipping and causing potential problems. > > > > I haven't looked around inside the CAC yet since it is XO's, > not sure if it > > is an > > ADIT 600 but it sounds like the same unit. I am set to "0" for > timing and > > "0" for > > LBO in Zaptel.conf I assume this is correct for * and that I > need to verify > > the a:1/a:2 timing settings in the CAC unit? > > > > Come to think of it, there is a way to test this without bringing the > > T1s down. The Adit 600 has a show performance command, I may be wrong, > > but I'm sure it was performance, anyways it allows you to see slips and > > bipolar violations and a couple other stats. This was beneficial for me > > as the T100P didn't report problems but the Adit did. > > > > Can you give me a brief idea of what slips and bipolar violations are? > > > > Home this helps. > > > > I am glad to find that someone knows more than my little > knowledge of the > > subject! > > To here the techs talk you'd think that they'd never run into > anything like > > this before. >
RE: [Asterisk-Users] Minimum budget question ...
Tim, a good comprehensive answer to the question...certainly gave me a few things to think about. I do have a few questions though, since I'm in Europe. Has anyone in Europe set up something equivalent to what Tim suggested? What sort of prices did it work out at? How did you solve the channel bank 'issue' in Europe? I keep reading that E1 lines are coax terminated, is this correct or do you usually get a choice from your teleco? Were there any other issues to contend with? I'd certainly be interested in the experiences of anyone in Europe... Thanks Andy On 30/06/2003 at 10:55 [EMAIL PROTECTED] wrote: >If this is for commercial use, especially if you are going to be selling >this solution, I would suggest that you don't even offer the choice of >analog lines except in the smallest of offices. Unless you like to >spend a lot of unbillable time supporting them :) > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Minimum budget question ...
Hi Tan, Thanks for the reply. I'll end up asking a load more questions now... What sort of prices are we talking about for the 24 port VoIP gateway? I assume that each port is individually addresable by *? As I recall the 24 port gateways tend to be terminated at the FXS side as some 'wierd' connector (wierd in that it's not rj45/11) do you just wire this to a patch panel? What codec is in use to get all 24 ports 'running' at the same time..G729? Does this cause problems since iirc * needs to run in console mode for the G729 codec to work properly Thanks for the info... interesting site too :D Andy *** REPLY SEPARATOR *** On 30/06/2003 at 19:21 Tan Aks wrote: >Hi, > >We provide asterisk-based solutions to customers based in the uk. One of >our >customers (9 users) is trialling our low-end solution which comprises of a >box with 2 x X100P (analogue line) cards installed, and a voip carrier for >outgoing calls. This customer intends to have 13 extensions in his "live" >scenario. The way to use multiple analogue phones is: > >1) get a T100P card and use a T1 channel bank sourced from the US >2) use a couple of TDM400P cards to give 8 extensions, and use IP >phones for the other extensions >3) use a voip gateway to provide up to 24 x analogue extensions per >IP address. VoIP gateways are commonly available and convert analogue lines >into a SIP/H323 VoIP stream. > >You can get an E1 terminated with an RJ45. If you have a coax termination >then you can use a balun to get rj45 connectivity. > >Hope that helps. >Tan (telappliant.com) > > > > >- Original Message - >From: "Andy Powell" <[EMAIL PROTECTED]> >To: <[EMAIL PROTECTED]> >Sent: Monday, June 30, 2003 5:26 PM >Subject: RE: [Asterisk-Users] Minimum budget question ... > > >Tim, > >a good comprehensive answer to the question...certainly gave me a few >things >to think about. I do have a few questions though, since I'm in Europe. > >Has anyone in Europe set up something equivalent to what Tim suggested? > >What sort of prices did it work out at? > >How did you solve the channel bank 'issue' in Europe? > >I keep reading that E1 lines are coax terminated, is this correct or do you >usually get a choice from your teleco? > >Were there any other issues to contend with? > >I'd certainly be interested in the experiences of anyone in Europe... > >Thanks > >Andy > > > > >On 30/06/2003 at 10:55 [EMAIL PROTECTED] wrote: > >>If this is for commercial use, especially if you are going to be selling >>this solution, I would suggest that you don't even offer the choice of >>analog lines except in the smallest of offices. Unless you like to >>spend a lot of unbillable time supporting them :) >> > > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] That is not a valid conference number meesage
I've just started trying to use this functionality and I get the invalid conference number message. Any ideas? I started out with: exten => 7315,1,Meetme,1234 and confno = 1234 and then tried: exten => 7315,1,Meetme and confno = 1234 and enter 1234 at prompt. All give the same message. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Coding Time/Date announce on voicemail
I have a customer asking to have the voicemail give the time and date a message is received when the message is played. Anyone have an idea of how big of a project it will be to code this into the voicemail app? Any suggestions would be greatly appreciated. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Sacrifice?
Hi, You just have to be a little patient... try http://www.automated.it/guidetoasterisk.htm as a start, it might at least get you going with sip based stuff. I don't like to particularly push the guide specifically because it's mine. I'd rather you got it by recommendation... but hey, Andy *** REPLY SEPARATOR *** On 04/07/2003 at 21:23 BK [address only for mailing lists] wrote: >Hi > >is there any ritual sacrifice a newbie has to perform to be welcome on >this list? > >I am new to this whole PBX thing in general and Asterisk in particular. >I had hoped that the community on this list would welcome a newbie like >myself and help me with some answers to my stupid questions, but somehow >it seems to me that nobody likes to respond to somebody who appears to >be a complete beginner -- too much bother and a risk to have to explain >everything from scratch -- better not answer at all and all that. > >Well, it may appear that way, but I am not a complete idiot. I know a >lot about mobile switching centres, HLRs, VLRs, IN service nodes, >mediation devices, billing and settlement systems etc -- I just don't >know much about PSTN and PBXes. I would appreciate it if somebody could >help me out with a few hints on how to set up my Asterisk box, in >particular in respect of VoIP as per my last posting. > >thank you very much in advance >kind regards >bk > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] switch => priority in the dialplan.. (probably an issue for Mark)
WipeOut, IIRC the qualify=yes directive in your iax.conf definition for the switch causes * to check to see if the host is alive. Andy >Is there a way to get asterisk to verify that the remote host is in fact >available before attempting the "switch" so that if it is unavailable the >local dialplan will process as if the "switch" statement wasn't there at >all?? > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel Alarms/Manager interface
Hi, I've had a search through the archives and can't find a mention of it at all. Is there any way that I can retrieve the alarm status of a line (like in zttool) from the manager interface. I'm building a small piece of hardware that connects to the manager interface so that you can see if a) asterisk is running ok b) a mailbox has mail c) an alarm state on a line The idea is that this device can be plugged into the network and could monitor an astersik box on either the local network or across the internet. The first two are done but it would be really useful to be able to see if the a line had gone down at the remote end... thanks Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and VMWare
Hi Dan, For a totally unrelated reason I did this today. * runs fine here under VMware, athough I haven't stressed it at all. Andy *** REPLY SEPARATOR *** On 07/07/2003 at 19:07 Dan wrote: >Hi, > >There is any experience using Asterisk with VMWare? >I think about installing a virtual linux box over VMWare and then Asterisk >over it. > >Thanks, >Dan > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank configuration
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Marty Mastera > Sent: Thursday, July 10, 2003 6:34 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Channel Bank configuration > > > Hello, > > I don't have any experience with channel banks and would appreciate any > feedback on my theory outlined below: > > We have a single T1 entering the building with channels 1-12 being voice > lines and 13-24 being a 768k internet connection. This T1 terminates to > an Adit 600 (T1-1). > > Here's what I know. Channels 11-12 go out the Adit 600's 25-pair > connector to a wiring block (and eventually to 2 fax machines - I assume > this is to have the fax machines bypass the currently installed phone > switch). The data comes out the Router card on the Adit and into > our network. > > The currently installed phone system is an NEC NEAX2000 IVS box which is > connected by CAT5 to the Adit 600's T1-2 port. (I am assuming that voice > channels 1-10 are mapped to the Adit T1-2 and getting to the NEC this > wayThere is also a 25 pair cable leaving the NEC and terminating on a > wiring block for the desk phones. > > So my assumption is that channels 1-10 are mapped as FXO onto the Adit's > 2nd T1 port, channels 11-12 to the Adit's 25-pair connector and 13-24 to > the router card for data. > > This leaves the NEC box to handle the FXS (and hence why it is directly > connected to the phones). So far so good... > > When I replace the NEC box with an * box/T100P, I'm thinking that I will > have to map Channels 1-12 to the T1-2 port and map the 8 Adit FXS > channels on the T1-2 port to the Adit's 25-pair for the Adit's FXS > capabilitythen run that T1 into the T100P and configure * to route > between the FXO and FXS channels appropriately. yes - map channels 1-12 to T1-2 on the Adit, but what are the "8 Adit FXS channels on the T1-2 port"? Also, what type of desk sets are you planning to use with *? > > Does this sound right? I'm trying to understand if the channel bank uses > the T1 from the Adit for both FXO and FXS channels! > > Thanks, > > Marty > > > > > -- > Jumping through hoops to get E-mail on the road? > You've got two choices: Join the circus, or use Molly Mail. > > Molly Mail -- http://www.mollymail.com > -- > Having trouble sending email from different locations ? > Need a single outgoing mail server that will work from anywhere ? > > Set it to smtp.com and never have to change it again ! > > http://www.smtp.com > -- > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank configuration
Marty, You will not be able to use these phones with *. As far as I know, there are no digital desksets that work with asterisk. This leaves you with either analog sets or VoIP sets. If you want to use analog sets, the config that you mentioned would probably work, but you would be dependant on your phone co. to make any changes or help with troubleshooting. This doesn't sound too good to me but then again, it might be less expensive. If you want to go VoIP, you could get the VoIP desk sets and plugem into an ethernet switch along with your * box. Whether or not you split out the analog lines at the ADIT or at the * box really just depends on whether you want to do any management on those calls. If you do want to run those through asterisk, you could get one of their new tdm cards instead of sending those channels back to the ADIT. Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Marty Mastera > Sent: Thursday, July 10, 2003 11:25 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Channel Bank configuration > > > Hello Andy, thank you very much for your response... > > The "8 Adit FXS channels on T1-2" that I was referring to is the 8 port > FXS card that is installed in the Adit. I was trying to describe our > voice channels 1-12 entering the * box via T1-2 and having * route up to > 6 extensions back via T1-2 to the Adit's 8 port FXS card and ultimately > to the phones via the 25-pair connector. > > Is this legitimate? > > I may have a problem with the phonesThe current phones are an NEC > model (DtermE) which say "for NEC pbx only" on the backdoes anyone > know if they will work? Otherwise can a decent business phone be > recommended (with transfer, hold, callerid, etc...capabilties)? > > Thanks again Andy, > > Marty > > On Thu, 10 Jul 2003 23:02:22 -0500 "Andy Hester" wrote: > > > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] Behalf Of > Marty Mastera > > > Sent: Thursday, July 10, 2003 6:34 PM > > > To: [EMAIL PROTECTED] > > > Subject: [Asterisk-Users] Channel Bank configuration > > > > > > > > > Hello, > > > > > > I don't have any experience with channel banks and would > appreciate any > > > feedback on my theory outlined below: > > > > > > We have a single T1 entering the building with channels 1-12 > being voice > > > lines and 13-24 being a 768k internet connection. This T1 > terminates to > > > an Adit 600 (T1-1). > > > > > > Here's what I know. Channels 11-12 go out the Adit 600's 25-pair > > > connector to a wiring block (and eventually to 2 fax machines > - I assume > > > this is to have the fax machines bypass the currently installed phone > > > switch). The data comes out the Router card on the Adit and into > > > our network. > > > > > > The currently installed phone system is an NEC NEAX2000 IVS > box which is > > > connected by CAT5 to the Adit 600's T1-2 port. (I am assuming that > > voice > > > channels 1-10 are mapped to the Adit T1-2 and getting to the NEC this > > > wayThere is also a 25 pair cable leaving the NEC and > > terminating on a > > > wiring block for the desk phones. > > > > > > So my assumption is that channels 1-10 are mapped as FXO onto > the Adit's > > > 2nd T1 port, channels 11-12 to the Adit's 25-pair connector > and 13-24 to > > > the router card for data. > > > > > > This leaves the NEC box to handle the FXS (and hence why it > is directly > > > connected to the phones). > > > > So far so good... > > > > > > > > When I replace the NEC box with an * box/T100P, I'm thinking > that I will > > > have to map Channels 1-12 to the T1-2 port and map the 8 Adit FXS > > > channels on the T1-2 port to the Adit's 25-pair for the Adit's FXS > > > capabilitythen run that T1 into the T100P and configure * to route > > > between the FXO and FXS channels appropriately. > > > > yes - map channels 1-12 to T1-2 on the Adit, but what are the > "8 Adit FXS > > channels on the T1-2 port"? > > > > Also, what type of desk sets are you planning to use with *? > > > > > > > > > > Does this sound right? I'm trying to understand if the c
RE: [Asterisk-Users] New Member
I am using Redhat 9.0 and have no problems...Redhat is a good choice if you want to get going fast. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Erik Anderson > Sent: Saturday, July 12, 2003 10:31 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] New Member > > > Redhat 8 > > We are having problems with RH 9. It is too bleeding edge. > > Erik > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of jltaylor > > Sent: Saturday, July 12, 2003 10:00 AM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] New Member > > > > > > Greetings, > > I'm ready to start and setup Asterisk. > > > > Any preference on which Linux to use? > > Windows & FreeBSD in use here. > > > > I'd like to get up and running as quickly and easily as possible. > > > > Thanks > > > > -- > > James Taylor > > [EMAIL PROTECTED] > > 903-793-1953 > > > > -- > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time and date stamp in voicemail
I have searched and not located this patch...is there a specific place that I need to look, or a specific file name? Andy > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of John Todd > Sent: Thursday, July 24, 2003 6:14 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] time and date stamp in voicemail > > > >Hi, > > > >I see that there's been some very light discussion on having a > standard time > >and date stamp in VM. How can I implement it today? (About to offer a > >system to a customer but they need the stamp to tell when people called.) > > > >Thanks, > >-- > > > >Steve > >__ > >This sig is pending approval > > Tilghman Lesher had a well-written patch he posted to the list a few > weeks ago for Voicemail, which I've been using without difficulty. > He has said he's going to work on Voicemail2, so I am hoping to see > that soon, and then integration into the main CVS tree after some > testing. > > Not only does the patch handle generic time and date stamps, but it > allows customizable timezones, announcement strings, and uses > standard UNIX-ish time code macros. Very slick, and really necessary > for voicemail systems that happen to have users in multiple timezones. > > If you are looking for words to match Tilghman's patch, see the > phrases I have submitted (and donated by a generous grant by > VoicePulse) as public domain, recorded by Allison Smith: > http://www.loligo.com/asterisk/sounds/ > > JT > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time and date stamp in voicemail
Dan, the page is actually http://asterisk.drunkcoder.com/patches/ . However, I didn't see the patch there. Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Dan > Sent: Friday, July 25, 2003 2:52 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] time and date stamp in voicemail > > > Hi, > > This page does not exist... > > Thanks, > Dan > > - Original Message - > From: "Steven Critchfield" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Friday, July 25, 2003 8:38 AM > Subject: RE: [Asterisk-Users] time and date stamp in voicemail > > > > Try looking drunkencoder.com/asterisk > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time and date stamp in voicemail
Tilghman, Thanks alot for posting that. I'll check it out.... Andy > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Tilghman > Lesher > Sent: Friday, July 25, 2003 10:48 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] time and date stamp in voicemail > > > On Friday 25 July 2003 14:12, Andy Hester wrote: > > Dan, > > the page is actually http://asterisk.drunkcoder.com/patches/ . > > However, I didn't see the patch there. > > I just added it. It's available there now. Note that there are three > files: a patch, sounds, and some instructions. > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time and date stamp in voicemail
Tilghman, I applied your voicemail_prompts patch and it works like a charm. Thanks for donating the code and thanks to those that donated the voice prompts! Another win for Asterisk Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Tilghman > Lesher > Sent: Friday, July 25, 2003 10:48 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] time and date stamp in voicemail > > > On Friday 25 July 2003 14:12, Andy Hester wrote: > > Dan, > > the page is actually http://asterisk.drunkcoder.com/patches/ . > > However, I didn't see the patch there. > > I just added it. It's available there now. Note that there are three > files: a patch, sounds, and some instructions. > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time and date stamp in voicemail
Tilghman, I'm not sure how to use this logic. Would this be for something like, for example, deleting of forwarding a message that a certain age? Andy > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Tilghman > Lesher > Sent: Sunday, July 27, 2003 11:24 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] time and date stamp in voicemail > > > On Saturday 26 July 2003 21:06, Andy Hester wrote: > > Tilghman, > > I applied your voicemail_prompts patch and it works like a charm. > > Thanks for donating the code and thanks to those that donated the > > voice prompts! Another win for Asterisk > > Is anybody at all using the variable substitution and/or the expression > logic at all? I'd like to know if and how well it works for you. > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time and date stamp in voicemail
Tilghman,
I will implement this... I think its a fairly important feature. Let me
know what you find in your testing.
Sincerely,
Andy Hester
Consero
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Tilghman
> Lesher
> Sent: Monday, July 28, 2003 1:24 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] time and date stamp in voicemail
>
>
> On Sunday 27 July 2003 23:57, Andy Hester wrote:
> > Tilghman,
> > I'm not sure how to use this logic. Would this be for something
> > like, for example, deleting of forwarding a message that a certain
> > age?
>
> No, this would be used in something like:
>
> central=US/Central|'vm-received' $[${DIFF_DAY} < 7]?A:BdY
>
> where it would read only the day of the week if the message were less
> than 7 days old and otherwise read the month, day, and year.
>
> Actually, now that I'm looking at the code, I don't think this works,
> so I'm going to do a bit more work and test it myself.
>
> -Tilghman
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
I have been planning to suggest this as well, but I would recommend setting up a zope site... If you set it up in zope you can have alot of collaboration very easily. You could, for instance, designate certain people who have expertise in a certain config/technology as project coordinator for that area of documentation. As well you could, as with any other system, create areas for specific technologies etc for the purpose of being easy to navigate. And of course the main benefit is the ability to have as many people collaborate as you wish, and still maintain organization of data with very low administration. Have a look...I think it would make all of our lives much easier. www.zope.org example site http://www.openparadigms.com allows you to create an account and post pages etc... Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Mark Spencer > Sent: Monday, July 28, 2003 4:09 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Offering an Asterisk Documentation and FAQ > Portal > > > Agreed. We're more than happy to host it. The problem is the writing of > it :) > > Mark > > On Mon, 28 Jul 2003, Scott Stingel wrote: > > > Just a suggestion, but wouldn't it be more appropriate for > Digium to host > > the documentation? > > > > I think the missing link here is someone who will write (and > illustrate) the > > documentation. All of this open source software is great > because it's free > > - but commercial users and others would certainly appreciate > the time saved > > by referring to a nice doc set. I for one would have been > willing to pay > > for a reasonable documentation set, especially at the outset. > > > > Maybe this is a commercial opportunity for a good tech writer - maybe > > working in collaboration with Digium. > > > > ...just my 2 cents! > > > > Scott > > > > > > > > Scott M. Stingel > > Emerging Voice Technology Inc. > > Palo Alto, California and London, England > > > > Email: [EMAIL PROTECTED] > > URL:www.evtmedia.com <http://www.evtmedia.com/> > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Damian Flynn > > Sent: Monday, July 28, 2003 8:17 PM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] Offering an Asterisk Documentation > and FAQ Portal > > > > > > Hi, > > > > I have resources available to host a portal specifically for > the Asterisk > > system, to help correlate documentation, FAQ's and How To > > > > I am new to Asterisk, and my hardest work is in locating information on > > using or configuring the software. > > > > Would Mark, John or any of you feel this would be of benefit to host? > > > > I am offering a PHP-NUKE portal for this, (Unless you know of a better > > solution!) > > > > Regards > > Damian > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE Pingtel Phones
Hello, Is anybody else out there using pingtel phones? If so, I like to hear your experiences... Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Foong Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below Channel: SIP/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: mysipcontext2 Extension: 2000 Priority: 1 This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. All you need is a script to lookup in the database and generate the script file for you and it's done. HTH Andy *** REPLY SEPARATOR *** On 30/07/2003 at 16:30 Chee Foong wrote: >Hello Dan, > >Thanks for you reply. > >Base on you recomendation using the 'T' argument. I manage to do call >transfer an it works really well. > >My problem comes when my boss comes out with a superb idea where the >transfering process is automated without involving a human :( > >Say asterisk get 2 numbers (from database, text file, etc), one belongs >party A and the other belongs to party B. Asterisk will calls both parties >and do the tranfer automatically. In another words, asterisk is resposible >to 'press' the '#' to do the transfer. I don't this can be achieve in the >extension.conf not matter how you structure you dial plan. > >Perhaps, the only way is to write a apps and plug it into asterisk like all >the asterisk modules such as Meetme. > >Any ideas? > > >Foong > >- Original Message - >From: "Dan" <[EMAIL PROTECTED]> >To: <[EMAIL PROTECTED]> >Sent: Wednesday, July 30, 2003 3:42 PM >Subject: Re: [Asterisk-Users] Call Transfer > > >> Hi, >> >> It works if you put the 'T' switch in the dial line. >> >> You can then transfer the call from the caller. >> I have tested it in the folllowing configuration and it works: >> Call from a Cisco 7960 to an ATA 186. >> Select 'Transfer" on 7960 >> Call another extension (X-Lite) >> Select again transfer on 7960. >> The call remain between ATA and X-Lite. >> >> This is what you need? >> >> BR, >> Dan >> >> - Original Message - >> From: "Chee Foong" <[EMAIL PROTECTED]> >> To: <[EMAIL PROTECTED]> >> Sent: Wednesday, July 30, 2003 7:08 AM >> Subject: [Asterisk-Users] Call Transfer >> >> >> Hello all, >> >> I am in a situation where I need to use asterisk to call someone say >Party >> A. After the call to Party A got through, asterisk will put Party A on >hold, >> then asterisk will call Party B. If call to Party B got through, asterisk >> will transfer Party A to Party B. >> >> I wonder if this features is implemented into asterisk. I have found a >post >> in asterisk mailing list: >> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html >> >> but that doesn't help much. >> >> If this features is not implemented, can anyone give me some point on how >to >> implement this in asterisk? Do I need to write an app like the Dial apps >for >> asterisk to load at start up? >> >> >> thanks >> >> Foong >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best Analog sets for use w/*
Hi All, I am considering testing out some analog sets with * for a customer and thought I would ask what analog phones are in use? The customer would require the usual business functionality ie hold, conference calling, and preferably a soft key to vm and line apearances(correct terminology?) in order for secretary to see if their boss is on the phone before transferring a call, etc. I would appreciate haering any of your experiences. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Analog sets for use w/*
TC, Have you used these phones? They seem to be pretty nifty... I'm wading through the documentation to see how well they would integrate. Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of TC > Sent: Thursday, July 31, 2003 8:11 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Best Analog sets for use w/* > you might want to take a look at http://smartalk.ca/nrgover.htm > these are analog phones designed for pc-pbx's, that have a rj-45 cable > pins 1-8 are looped as a system bus to allow you to intercom, & see which > sets are in use... the reception phone SE310 has more soft keys & larger > display > they all have the ability to program the soft keys > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with the Internet LineJACK ISA card...
You also appear to have a big problem with your clock... unless you are from the future.. in which case how are Glaxo stocks doing? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] unsubscribe
Steve I have to say that some listserv's do allow this .. at least he didn't reply to 20 messages with REMOVE in them Andy On 07/08/2003 at 10:10 Steve Meyers wrote: >On Thu, 2003-08-07 at 10:01, Justin Carlson wrote: >> unsubscribe > >Has anyone ever been on a mailing list where you could unsubscribe >simply by sending a message with "unsubscribe" in it to the mailing >list? I swear, every list I've been on, people try to do that, but it >doesn't work on any of them. > >Steve > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ansering machine/voicemail detect?
Garry, yes this is possible although it would end up being quite convoluted. Essentially you could have a cron job that monitors your voicemail directory, or use the perl manager interface to check the status. Once it has been established that you have message(s) submit a .call file to dial you office number, and start an AGI script. In the AGI you ask for your pin number and time out (and hang up) if there is no response. If there is a correct response then you could playback the files. You'd need to move them to the OLD directory afterwards. Alternatively, someone has written app_hasvoicemail which checks a vm box for messages and acts on that... Andy On 07/08/2003 at 13:17 Garry Adkins wrote: >I've been playing with an asterisk box for about 6 months, (bought an FXO >card, etc.)... > >I was thinking about having the system "deliver" my voicemail from the >asterisk machine to me at work... I haven't found anything in the >documentation to help. > > >It would work something like: > > >Voicemail comes into the asterisk machine, >* Calls me at work >Plays message for me to enter PIN for voicemail >Retrieve Voicemail >Hangup. > > >However, if it got my voicemail at work (due to being on the phone or out >of the office), I'd like it to do something like: >Voicemail in * >* Calls me at work >Notices that it's voicemail (Possibly due to no pause at the beginning, >just continuous talking?) >Just plays a message that I have voicemail at home. >Hangs up. > >Possible? How could it tell that it got an answering device? > >Thanks! >-G >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID, DECT phones and ATA
Hi Dan, when you get voicemail the dial tone changes... not only that but on my dect phones i get a little specaking face icon that flashes... Andy >One more question about TDM20B. How can you know when you have a new >voicemail in your mailbox? >On ATA, the dialtone is modulated for a couple of seconds when you pick-up >the phone, to know that you have a voice message waiting. > >BR, >Dan > > > >- Original Message - >From: "Andy Powell" <[EMAIL PROTECTED]> >To: <[EMAIL PROTECTED]> >Sent: Friday, August 08, 2003 2:34 PM >Subject: Re: [Asterisk-Users] CallerID, DECT phones and ATA > > >> Hi Dan, >> >> I use panasonic DECT phones, when plugged into a TDM20B (2 port FXS card >from Digium) I get caller id passed through (name AND number) although i >can't get callerid via the pstn at the moment (located in nl) i do get it >for VoIP calls. Plus when a pstn call comes in and there is no clid (ie all >the time!) I force asterisk to set the caller id to "PSTN Call" number> >> >> The panasonic was purchsed in Currys in the UK ... personally I think >that the panasonic's understand all eu caller id standards (I can't prove >this) and just sell the same model throughout europe... Certainly the >manual >seems to cover all european languages... >> >> It may not be what you want to hear but I'd seriously consider getting an >fxs card.. >> >> Andy >> >> *** REPLY SEPARATOR *** >> >> On 08/08/2003 at 13:45 Dan wrote: >> >> >Hi, >> > >> >I have two DECT cordless phones (one Philips Onis 6311 and one Philips >> >Onis2 >> >Memo-6511). >> >The first one is for the french standard and the second one for the >british >> >standard. >> >Both of them have callerid functionality. >> >The british one does not show anything when connected to my PSTN line. >> >I have not yet tested the french one. >> >What I want to do is to connect both of them to the same ATA box. >> >There is any possibility to program callerid functionality in ATA in >order >> >to have different callerid standard on each port? >> >It would be nice to use this to get callerid functionality on both >phones, >> >as the X100P card takes it from the PSTN line. >> >Any hint or experience is wellcome. >> > >> >Thanks, >> >Dan >> > >> >___ >> >Asterisk-Users mailing list >> >[EMAIL PROTECTED] >> >http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Running Asterisk behind NAT?
Is this not just a case of a new entry in sip.conf EXTERNIP = with the code for the contact header modified to use it (if present). Then the external firewall is set to forward the rtp and 5060 to * .. I know many people either have sip aware firewalls (as i do) or their * box has a real IP, but the number of people requesting this feature seems to be growing by the hour. I'm trying to get this working for quite a number of FWD users, at the moment I'm trying to fudge it with partysip... it's not very pretty and requires a linux iptables based firewall it's not big, it's not clever and it's certainly not funny Andy *** REPLY SEPARATOR *** On 14/08/2003 at 09:19 Dave Cotton wrote: >On Wed, 2003-08-13 at 10:59, John Todd wrote: >> This is starting to sound like a feature request, > >Absolutely, that would be the real icing and cherry on the cake all in >one go. > >Totally seamless coms behind a NAT firewall. >IAX, Analog, ISDN, SIP and H323, etc..., if Pavlov could see me now. > >-- >Dave Cotton <[EMAIL PROTECTED]> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
Errm, no... does that mean you'll personally check to see if my line is busy or not ;P will try it now... Andy *** REPLY SEPARATOR *** On 14/08/2003 at 09:58 Martin Pycko wrote: >Did you try BUSYDETECT_MARTIN in asterisk/Makefile ? > >regards >Martin > >On Thu, 14 Aug 2003, Andy Powell wrote: > >> Hi Dave, >> >> I have a similar problem, I tried using busydetect and busycount but >calls kept being dropped >> at random intervals. It didn't seem to matter what i set the busycount >to. I guess it's a case >> of deciding which is more important... You can also limit the length of >the voicemails using >> >> ; Maximum length of a voicemail message >> maxmessage=180 >> >> in voicemail.conf >> >> which cuts down the length of the recorded dial tone... >> >> Andy >> >> >> >> >> >> >Thanks Andy. The stage I'm at at the moment is that I've removed the >> >code for the US and so dmesg will show CTR21 without the modprobe >> >option. But I don't think that is the whole problem. >> > >> >Last night I posted showing that the problem is repeatable and only >> >occurs in one certain circumstance. I think it is within voicemail.c. If >> >the caller exits voicemail by pressing # the line is dropped correctly, >> >if they just hang up voicemail continues to record. I put some debugging >> >statements into voicemail.c and I think that a condition statement is >> >never reached so the line is held up. The routine in question is 14 >> >pages long, so it reminds me of my Cobol days when we used to lay the >> >printouts along the corridor to debug them. >> > >> >As far as the option to modprobe is concerned couldn't zaptel.conf be >> >used for this as it would be more obvious, I only heard about the option >> >from your post last night. >> > >> >-- >> >Dave Cotton <[EMAIL PROTECTED]> >> > >> >___ >> >Asterisk-Users mailing list >> >[EMAIL PROTECTED] >> >http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
>> FCC mode is for the US. CTR21 is for Europe - you even pasted the info >> in your message! > >Exactly, the question really is how do you change it? > modprobe wcfxo opermode=1 HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] list proposal
Perhaps there is another way to cut down on increased traffic... Specifically, I would go back to the suggestion of a collaborative website for documentation. Collecting info and organizing into Howto's would reduce the number of times people ask the same questions. Also, the documentation could grow as quickly as the project. Unfortunately, I don't have a place to host it currently. Ideally, the list would just be for issues that aren't already addressed. Any one else interested in this? Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven > Critchfield > Sent: Friday, August 08, 2003 1:25 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] list proposal > > > With the increased traffic as of late, I'm wondering if it is time to > split the list again. Specifically I am wondering if it should be split > along the various VoIP protocols and zap hardware, then leave a general > list that does configuration other than VoIP related? > > The hope is that those asking SIP or H323 questions could get help from > the various supporters while the main list can deal with transport > neutral content like extension logic and voicemail configs. > > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] list proposal
I was pondering on this question, and have to agree, splitting mailing list just means yet another list to join (since there may one day be something relavant) and filter locally. What might appear to be a good solution is a privately run newsgroup on a digium server eg news.digium.com the groups would be the same as the lists plus one of two more asterisk-announce asterisk-groups asterisk-dev asterisk-users asterisk-forsale asterisk-consultants of course then you have the potential problem that people can't read these groups because of blocked firewall ports at their offices etc.. all IMHO Andy *** REPLY SEPARATOR *** On 09/08/2003 at 10:28 WipeOut . wrote: >My concern with wanting to split the list into so many mini-lists is that >some will die from lack of membership and it will mean that people with >problems in that area will be left in the cold.. Similar to how so many >news groups just sit there unused now days.. > >If there is really a need to split the list into so many components then >somthing like phpbb may be better so that there is no management of list >membership.. > >I know the topic of a BB style system has come up before and many did not >like the idea but if one or two mailing lists become 10+ then mailing >lists can become a headache to look after where a BB system will do it all >for you.. especually the poor "MS Outbreak" users.. :) > >Just my thoughts... > >> >> I think that's a very good idea. When I started to become active in * >last >> December the list was much less congested and Mark usually responded to >> requests, comments and patches within a few hours. Now things are >clearly >> taking off - good for * and Digium but it's sort of losing the community >> spirit. >> >> Splitting the lists by the channel drivers seems to be a good idea but I >> think there needs to be a strong link with the development team. In the >> case of SIP, it is clearly becoming the protocol for VoIP and many >people >> would like the channel driver enhanced or may have patches for it. I'm >not >> sure the bug tracker is the right place for this. It would be better to >> have a SIP list moderated by a developer where changes could be >discussed. >> >> Doubtless the same reasoning could apply to ISDN, IAX etc. >> >> Iain >> >> >> >> --On Friday, August 08, 2003 13:25:10 -0500 Steven Critchfield >> <[EMAIL PROTECTED]> wrote: >> >> > With the increased traffic as of late, I'm wondering if it is time to >> > split the list again. Specifically I am wondering if it should be split >> > along the various VoIP protocols and zap hardware, then leave a general >> > list that does configuration other than VoIP related? >> > >> > The hope is that those asking SIP or H323 questions could get help from >> > the various supporters while the main list can deal with transport >> > neutral content like extension logic and voicemail configs. >> > >> > -- >> > Steven Critchfield <[EMAIL PROTECTED]> >> > >> > ___ >> > Asterisk-Users mailing list >> > [EMAIL PROTECTED] >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> >> >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users > >-- >__ >http://www.linuxmail.org/ >Now with e-mail forwarding for only US$5.95/yr > >Powered by Outblaze >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Newbie ...
Fabia,
The only numbers you should be able to dial from that config are
1945
1943
2999
and nothing else...
The entry under bogon-calls (isn't it bogus calls?) should read
exten => s,1,Congestion
rather that using the _. ...
HTH
Andy
*** REPLY SEPARATOR ***
On 10/08/2003 at 15:13 Fabia wrote:
>Hi ;)
>
>I'm a french newbie and i installed asterisk 1 day ago.
>I've got an ATA186 and a computer with Sjphone installed.
>
>If i want to call the sjphone from the ata or call the ata from de sjphone
>everything is ok.
>My problem is ,that i can't call the voicemail or any other phone number
>..as 600 for exemple from the ata or the jphone.
>I don't know why but i looked after a long time..
>
>here a copy of my extension.conf , sip.conf and voicemail.conf.
>
>Thanks for your help.
>Julien.
>
>Extension.conf
>
>[general]
>
>static=yes
>writeprotect=yes
>
>[bogon-calls]
>exten => _.,1,Congestion
>[from-sip]
>exten => 1943,1,Dial(SIP/1943,5)
>exten => 1943,2,Voicemail(u1943)
>exten => 1943,102,Voicemail(b1943)
>exten => 1943,103,Hangup
>
>exten => 1945,1,Dial(SIP/1945,6)
>exten => 1945,2,Voicemail(u1945)
>exten => 1945,102,Voicemail(b1945)
>exten => 1945,103,Hangup
>
>exten => 2999,1,VoicemailMain(${CALLERIDNUM})
>
>
>-
>sip.conf
>
>[general]
>
>port = 5060
>bindaddr = 0.0.0.0
>allow=all
>context = bogon-calls
>
>[1943]
>
>type=friend
>username=1943
>secret=1943
>host=dynamic
>context=from-sip
>mailbox=1943
>
>[1945]
>
>type=friend
>username=1945
>secret=1945
>host=dynamic
>context=from-sip
>mailbox=1945
>---
>voicemail.conf
>
>[general]
>
>format=wav
>
>[local]
>
>1943 => 1943,Essai 1,[EMAIL PROTECTED]
>1945 => 1945,Essai2,[EMAIL PROTECTED]
>
>___
>Asterisk-Users mailing list
>[EMAIL PROTECTED]
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IP phone recommendation
Nathan, I am using the Pingtel phones at a customer site. I should be able to give a report in a couple of days Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Nathan > Littlepage > Sent: Wednesday, August 13, 2003 8:15 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] IP phone recommendation > > > Has anyone had the opportunity to use a PingTel phone with Asterisk? > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
Hi Dave, I have a similar problem, I tried using busydetect and busycount but calls kept being dropped at random intervals. It didn't seem to matter what i set the busycount to. I guess it's a case of deciding which is more important... You can also limit the length of the voicemails using ; Maximum length of a voicemail message maxmessage=180 in voicemail.conf which cuts down the length of the recorded dial tone... Andy >Thanks Andy. The stage I'm at at the moment is that I've removed the >code for the US and so dmesg will show CTR21 without the modprobe >option. But I don't think that is the whole problem. > >Last night I posted showing that the problem is repeatable and only >occurs in one certain circumstance. I think it is within voicemail.c. If >the caller exits voicemail by pressing # the line is dropped correctly, >if they just hang up voicemail continues to record. I put some debugging >statements into voicemail.c and I think that a condition statement is >never reached so the line is held up. The routine in question is 14 >pages long, so it reminds me of my Cobol days when we used to lay the >printouts along the corridor to debug them. > >As far as the option to modprobe is concerned couldn't zaptel.conf be >used for this as it would be more obvious, I only heard about the option >from your post last night. > >-- >Dave Cotton <[EMAIL PROTECTED]> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] To Switch or not to Switch... that is the question....
Hi, when using multiple * boxes, there appear to be 2 choices as to how to go about sharing cards and dialplans 1) using switch 2) using dial fail fall-through ie exten => 1,1, dial(xyz) exten => 1,2, dial(otherpbx/xyz) As i see it switch could end up being recurrsive resulting in a wild ooc pbx .. anyone got any opinions ? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New SIP Phone
It's just a proxy service like fwd it will work with asterisk... The phones they are selling with the deal are Grandstreams. It's very likely that they just have been preloaded with their settings, and probably point to their own tftp server. simply create fake dns entries and a static route so that it looks on your network for the ip of their tftp server... I'd guess that you could still do firmware upgrades... so that's a Grandstream for $65 ... Andy *** REPLY SEPARATOR *** On 06/08/2003 at 13:38 George Pajari wrote: >Michael Robertson, founder of both MP3.com and Lindows, has launched a >new company to supply inexpensive SIP phones ($129 for two) and related >services. See today's press release at >http://www.sipphone.com/tiki-index.php?page=SIPphone%20Inc > >My question for the list is who will be the first to report on the >compatibility and usability of the SIPphone with Asterisk? The >functionality described in the press release suggests the phones are >pre-configured to work with Robertson's service so one issue to be >resolved is how to reconfigure the phones to use one's * server. > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
On 13/08/2003 at 17:46 Dave Cotton wrote:
>in the file wcfxo.c the following structure is initialised as below
>which would suggest that FCC is wrong for France or pretty well all of
>Europe.
errm,
FCC mode is for the US. CTR21 is for Europe - you even pasted the info
in your message!
See below
Andy
>
>static struct fxo_mode {
>char *name;
>int ohs;
>int act;
>int dct;
>int rz;
>int rt;
>int lim;
>int vol;
>} fxo_modes[] =
>{
>{ "FCC", 0, 0, 2, 0, 0, 0, 0 }, /* US */
>{ "CTR21", 0, 0, 3, 0, 0, 3, 0 }, /* Austria, Belgium, Denmark,
> Finland, France, Germany, Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands,
> Norway, Portugal, Spain, Sweden, Switzerland, and UK */
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New SIP Phone
In fact the is is not required, see below: On 06/08/2003 at 15:50 Michael Robertson wrote: >The phones are completely preconfigured, but not locked in any way to >the SIPphone service. Owners are free to change any settings they want. > >-- MR > >Andy Powell wrote: > >>Hi, >> >>Might seem an obvious question but are the phones locked to your >>service? or can i mess about with them to connect to an Asterisk box >>and dial your service via that? >> >>Thanks >> >>Andy >> On 07/08/2003 at 00:20 Andy Powell wrote: >It's just a proxy service like fwd it will work with asterisk... The >phones they are selling >with the deal are Grandstreams. > >It's very likely that they just have been preloaded with their settings, >and probably point to their >own tftp server. simply create fake dns entries and a static route so that >it looks on your network >for the ip of their tftp server... > >I'd guess that you could still do firmware upgrades... > >so that's a Grandstream for $65 ... > >Andy > >*** REPLY SEPARATOR *** > >On 06/08/2003 at 13:38 George Pajari wrote: > >>Michael Robertson, founder of both MP3.com and Lindows, has launched a >>new company to supply inexpensive SIP phones ($129 for two) and related >>services. See today's press release at >>http://www.sipphone.com/tiki-index.php?page=SIPphone%20Inc >> >>My question for the list is who will be the first to report on the >>compatibility and usability of the SIPphone with Asterisk? The >>functionality described in the press release suggests the phones are >>pre-configured to work with Robertson's service so one issue to be >>resolved is how to reconfigure the phones to use one's * server. >> >>___ >>Asterisk-Users mailing list >>[EMAIL PROTECTED] >>http://lists.digium.com/mailman/listinfo/asterisk-users > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
Can't find the message in a search.. but below is a msg retreved from my archive.. this is what Mark sent a little while ago I have no idea if it actually does anything to the card, but on a modprobe I do get a msg saying it's using CTR21 Andy > >I'm in Paris right now and can't test this change, but I've been >researching the DAA and there are a few international settings I can >change, so I've changed the driver in CVS so that you can specify >the operational mode. Try "modprobe wcfxo opermode=1" if you're in most >of Europe and that should switch to CTR21 mode which slightly modifies a >few of the electrical characteristics of the DAA. > >As we add modes you'll be able to see them with "modprobe wcfxo >opermode=-1" and then doing a dmesg. > >Anyway all you folks that had some trouble like this try it out and let me >know if it makes any difference. > >Mark > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > *** REPLY SEPARATOR *** On 14/08/2003 at 20:48 Richard Scobie wrote: >Andy Powell wrote: >>>>FCC mode is for the US. CTR21 is for Europe - you even pasted the info >>>>in your message! >>> >>>Exactly, the question really is how do you change it? >>> >> >> >> >> modprobe wcfxo opermode=1 >> >> HTH >> >> Andy >> > >This switch (opermode=1) is redundant with the current X100P cards, as >it changes register contents that are specific to the "Global" version >of the chipset on the card. > >The X100P currently out there uses the "USA & Japan" chipset, and thus >does not achieve the intended result. > >The register concerned deals with the impedance presented to the line >connected to the card - 600 ohms (US) vs various complex impedances used >in other countries. > >For internationalised FXO cards, see Mark's recent comments, in the >thread "Does Wildcard x100p support BT Caller ID in UK?" > >Regards, > >Richard > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
Ok, I've changed it to use BUSYDETECT_MARTIN and I'll see how it goes and let you know.. thanks for the tip... Andy *** REPLY SEPARATOR *** On 14/08/2003 at 09:58 Martin Pycko wrote: >Did you try BUSYDETECT_MARTIN in asterisk/Makefile ? > >regards >Martin > >On Thu, 14 Aug 2003, Andy Powell wrote: > >> Hi Dave, >> >> I have a similar problem, I tried using busydetect and busycount but >calls kept being dropped >> at random intervals. It didn't seem to matter what i set the busycount >to. I guess it's a case >> of deciding which is more important... You can also limit the length of >the voicemails using >> >> ; Maximum length of a voicemail message >> maxmessage=180 >> >> in voicemail.conf >> >> which cuts down the length of the recorded dial tone... >> >> Andy >> >> >> >> >> >> >Thanks Andy. The stage I'm at at the moment is that I've removed the >> >code for the US and so dmesg will show CTR21 without the modprobe >> >option. But I don't think that is the whole problem. >> > >> >Last night I posted showing that the problem is repeatable and only >> >occurs in one certain circumstance. I think it is within voicemail.c. If >> >the caller exits voicemail by pressing # the line is dropped correctly, >> >if they just hang up voicemail continues to record. I put some debugging >> >statements into voicemail.c and I think that a condition statement is >> >never reached so the line is held up. The routine in question is 14 >> >pages long, so it reminds me of my Cobol days when we used to lay the >> >printouts along the corridor to debug them. >> > >> >As far as the option to modprobe is concerned couldn't zaptel.conf be >> >used for this as it would be more obvious, I only heard about the option >> >from your post last night. >> > >> >-- >> >Dave Cotton <[EMAIL PROTECTED]> >> > >> >___ >> >Asterisk-Users mailing list >> >[EMAIL PROTECTED] >> >http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] list proposal
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven > Critchfield > Sent: Sunday, August 10, 2003 10:31 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] list proposal > > > On Sun, 2003-08-10 at 21:25, Andy Hester wrote: > > Perhaps there is another way to cut down on increased traffic... > > > > Specifically, I would go back to the suggestion of a > collaborative website > > for documentation. Collecting info and organizing into Howto's > would reduce > > the number of times people ask the same questions. Also, the > documentation > > could grow as quickly as the project. Unfortunately, I don't > have a place > > to host it currently. Ideally, the list would just be for issues that > > aren't already addressed. Any one else interested in this? > > While it still needs to be done, the majority of those type questions > will still happen as the newest users still don't use google until told > to do so. There will always be some of this, and I agree that most people won't search for an answer if they can have someone else hand it to them. However, the idea is to make it such that they don't have to search for the answer among thousands of other posts, but rather just read the instructions. And like you said, it needs to be done anyway, so why not try to make the documentation as complete as possible and maybe post a large warning, don't ask questions of the list if you haven't read the documentation! And place this documentation or a link to it before the info on how to sign up for the mailing lists. I imagine many new comers never even see the google search box down the page. It doesn't really bother me that much when those questions are posted mind you, but I just think it is a really bad idea to use a mailing list as the repository for documentation. I know that documentation exists, but it is so incomplete that someone is always telling documentation seekers to "search the list". We should try to move the documentation away from the list and into a seperate place. As new or unaswered questions come up, the documentation can be easily updated if it is collaborative. Anyway, I don't mind splitting the lists as I will just subscribe to those as well. It might work out well as I had also suggested that the documentation be split out by technology also. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running Asterisk behind NAT?
> >I think this is a good idea but at least for FWD users can't they just use >the FWD proxy that is >designed to handle clients behind NAT with no special software on the >client. The ones that >allows even Windows Messenger to work behind NAT. > Sadly this doesn't work otherwise they'd all be using it. I think it's a specific * SIP impelmentation issue. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?
Hi Mark, Short of taking my board out of my * box is there any way to check what revision of the TDM400P I have? It was purchased in May of this year. Is the pricing likely to be the same or similar to the add-on FXS ports? Does this also mean that as we'd be able to get away with not using the x100p we could add another TDM400P card taking the maximum up to 3, ie 12 ports? Below is what is dumped from lspci -vv on my system, hopefully it would help identify the board I have many thanks Andy 01:09.0 Network controller: Tiger Jet Network Inc. Model 300 128k Subsystem: Tiger Jet Network Inc. Model 300 128k Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- TAbort- SERR- > I just hope that the price difference is small enough for Digium to >> consider this other chipset. From the lists it is obvious that there is >> a lot of interest for their hardware outside the US/Japan market. Same >> goes for the rumoured 4port fxo cards, of course. > >The FXO modules we've designed for the TDM400P's are definitely designed >for worldwide operation and should be able to detect polarity reversal. >Also there will be an option for hardware echo cancelation. The first rev >of the FXO module is in layout now so hopefully we'll have some prototypes >in the next few weeks. > >Clearly, the TDM400P is aspiring to be the logical platform for FXS/FXO in >low density. We've done some very substantial improvements on the TDM400P >to obtain extremely low noise on the "Rev E" boards, and to eliminate >other problems people were reporting (e.g. bus master aborts, etc). > >Mark > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Newbie ...
Julien,
try adding defaultip= in your sip.conf for each phone definition
Andy
*** REPLY SEPARATOR ***
On 10/08/2003 at 16:28 Julien wrote:
>Yes, the voice mail is at 2999 , but it doesnt work when i call it from
>the
>ata .I talked about the 600 (echo test) but i removed it from the
>extension.conf, sorry.
>
>In sjphone , i've got this error
>
>15:06:42 INFO Session rejected. Reason: 404 Not Found
>15:06:42 INFO Call 153 ended: Session rejected: 404 Not Found
>15:06:42 INFO SIP: Session terminated.
>
>And the configuratin of sjphone works, i can call the ata from this pc.
>On the console the 2999 is available, i can acces my voicemail.
>
>I think something is missing ... but it's difficult for me to understand
>how
>everything works in asterik. I read all night long documentation on the net
>, and the handbok-draft without succes.
>For the moment i don't want to use a card to connect to my phone line ...
>Just 1 ata with 2 phones , and a soft phone with their own voicemail.
>
>Julien.
>
>- Original Message -
>From: "Florian Overkamp" <[EMAIL PROTECTED]>
>To: <[EMAIL PROTECTED]>
>Sent: Sunday, August 10, 2003 3:47 PM
>Subject: Re: [Asterisk-Users] Asterisk Newbie ...
>
>
>> At 15:13 10-8-2003 +0200, you wrote:
>> >If i want to call the sjphone from the ata or call the ata from de
>sjphone
>> >everything is ok.
>> >My problem is ,that i can't call the voicemail or any other phone number
>> >..as 600 for exemple from the ata or the jphone.
>> >I don't know why but i looked after a long time..
>> >
>> >[from-sip]
>> >exten => 1943,1,Dial(SIP/1943,5)
>> >exten => 1943,2,Voicemail(u1943)
>> >exten => 1943,102,Voicemail(b1943)
>> >exten => 1943,103,Hangup
>> >
>> >exten => 1945,1,Dial(SIP/1945,6)
>> >exten => 1945,2,Voicemail(u1945)
>> >exten => 1945,102,Voicemail(b1945)
>> >exten => 1945,103,Hangup
>> >
>> >exten => 2999,1,VoicemailMain(${CALLERIDNUM})
>>
>> Call me silly, but this does mean your Voicemailbox is at extension 2999,
>> not 600. Or did I misunderstand you ?
>>
>> BTW, any other phonenumber not being callable would make sense, since you
>> simply don't have anything else in the [from-sip] context. You could
>> include the IAXtel gateway or add a device to connect to your phone
>line...
>>
>> Florian
>>
>> ___
>> Asterisk-Users mailing list
>> [EMAIL PROTECTED]
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>___
>Asterisk-Users mailing list
>[EMAIL PROTECTED]
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?
Mark, if the capability for line reversal detection is in the hardware (X100P) then does this mean that the detection of DTMF style caller-id as used in the following countries would ber trivial? or am I hoping too much... Finland, Denmark, Iceland, the Netherlands,India, Belgium, Sweden, Brazil, Saudi Arabia, Uruguay and Japan Imagine all those happy people!! Andy On 09/08/2003 at 10:44 Mark Spencer wrote: >> Maybe you could open a bug for it, and attach the specs / a link to >> those specs? Also, I suggest you reply to this message: > >That's a great idea. The other thing is that we have to detect polarity >reversal or we'll constantly be scanning for CID. > >Mark > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
