RE: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.
Hi, But it's seems the auto-answer function work on my spa-941. Have you upgrade to the latest firmware version? Regards, kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Friday, May 05, 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work. Asterisk wrote: Hello all, I want to report a BUG with the Linksys SPA94X so it is general knowledge and that we can all make noise about it so it will get fixed sooner.. The handsets do not work with the SIP flag to make them AUTO-ANSWER. (As documented is should) Ie, you cannot use them with intercom or Page features. This works with the Sipura841 fine. So linksys broke it. Um.. interesting is it not, considering it works with there SPA9000 unit... sounds a bit fishy to me.. So any Linksys owners using Asterisk, do pass on some discontentment, and Email linksys tech support at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] And tell them you have this issue.. James Curious, as I tried to get this to work with the 841, and though the phone does auto answer, the called or paged party hears dial tone as well as the page, just as if one went off hook by pressing the speaker button. The Pager does NOT hear dial tone. I sent support some information, but so far no help. They asked for more and I have yet to get back to them Curious, very curious. ___ --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 ___ --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 ___ --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] Cepstral , options to read the contents of a file
Hi, You can call an agi script to convert the text file to wave format. Example: http://www.voip-info.org/wiki/view/swift.agi Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Joseph Sent: Monday, May 01, 2006 7:08 PM To: Asterisk Users Subject: [Asterisk-Users] Cepstral , options to read the contents of a file Hi I had installed Cepstral , and it is working in Asterisk , it workfine for exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Cepstral( This is Just a test ) exten = s,4,Cepstral(Hope u are getting this voices) but instead of the text contents for Cepstral , can I use the file name location , where it can read the file Thanks Joseph John ___ 24 FIFA World Cup tickets to be won with Yahoo! Mail http://uk.mail.yahoo.com ___ --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 ___ --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] CallerID Name problem
Hi, What protocol for your 7960 phone? SCCP or SIP? You can turn on the SIP debug on CLI to make sure the callerid and name pass to your phone. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Tuesday, May 02, 2006 12:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] CallerID Name problem I'm having trouble getting callerid name to show up on my phones (Cisco 7960 and a few softphones) When I look in the CDR database I see the name but not on any phone when being called. I'm running Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC Any help would be great ! ___ --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] treating an incoming call as a local extension
Hi, Check the DISA command. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DISA Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jnuoiqweahf kajhdsff Sent: Thursday, April 27, 2006 12:21 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] treating an incoming call as a local extension I have [EMAIL PROTECTED] running on one machine, with X-lite running on another machine on my local network, with X-lite logged in to asterisk as extension 200. From X-lite, I can dial *97 to hear voice mail for extension 200, dial 201 to call extension 201, etc. I need to be able to accept an incoming call over the voip trunk which I have set up, and have asterisk treat that call as extension 202, so that e.g. I can dial in to asterisk from an external voip line and then as soon as asterisk answers the line, I can enter enter a password and then have the call treated as extension 202 and enter *97 to access the voicemail for 202, or enter 200 to locally call extension 200, etc. How can I do this? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 ___ --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] Re: Pattern matching problem
So sorry, the correct version is 1.2.6 :-) kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Thursday, April 27, 2006 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: Pattern matching problem My * version is 2.1.6. ... Did I miss something? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 ___ --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] Re: Pattern matching problem
Yes, you are correct.I am so sorry. I never use the zap analog card. We only have one digium T1/E1 PCI card in our small office. One more question, The analogue zap channel is fxo port? Or fxs port? Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Thursday, April 27, 2006 9:10 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: Pattern matching problem On Wednesday 26 April 2006 20:54, kevin ling wrote: Same dial pattern on my extension.conf, But it's work great. The Asterisk only match 7 digits number. My * version is 2.1.6. From an analogue Zap channel? Bullshit. Analogue channels do not present the extension in one shot -- they present the digits one at a time, in sequence. When the dialplan matches, it matches. Why do you think the telco needs you to enter 1 for long distance? And why do you think they're moving to ten-digit dialing for so many areas? This is very very basic, standard pattern matching. Analogue channels are very different from digital ones in how the desired extension or telephone number is presented to the switch. -A. ___ --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 ___ --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] Re: Pattern matching problem
Hi Andrew,
Sorry for my english first.
My configuration and hardware: AAH2.7 2.8, Digium TE100P, welltech 4fxo
voice gateway
SIP Phone
|
|
Asterisk Server - TE100P - Telcom1
|
+ Welltech 4FXO voicegateway Telcom2
Actually no matter on the digital interface (TE110P) or analog channels
(4FXO). Bellowing is my outbound routing config. I try to dial 6137451576
number. The asterisk doesn't match this dial pattern. And when I dail
6137451. It's work.
So you mean the analogue channels is analog phone attach on a fxs port?
[outrt-001-outside]
include = outrt-001-outside-custom
exten = _NXX,1,Macro(dialout-trunk,1,${EXTEN},,)
exten = _NXX,n,Macro(outisbusy,)
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Thursday, April 27, 2006 9:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Re: Pattern matching problem
On Wednesday 26 April 2006 20:54, kevin ling wrote:
Same dial pattern on my extension.conf, But it's work great. The
Asterisk only match 7 digits number. My * version is 2.1.6.
From an analogue Zap channel? Bullshit.
Analogue channels do not present the extension in one shot -- they present
the digits one at a time, in sequence. When the dialplan matches, it
matches. Why do you think the telco needs you to enter 1 for long
distance? And why do you think they're moving to ten-digit dialing for so
many areas?
This is very very basic, standard pattern matching. Analogue channels are
very different from digital ones in how the desired extension or telephone
number is presented to the switch.
-A.
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On Wednesday 26 April 2006 20:34, hugolivude wrote:
Thanks, but the problem's with the first extension:
exten = _NXX,1,NoOp(Number dialed ${EXTEN}) exten =
_NXX,n,Dial(Zap/1/${EXTEN})
The problem is I _do_ get a match as you can see by the CLI output,
but it shouldn't match IMO - 6137451576 shouldn't match _NXX but
that line gets executed.
When you dial six one three seven four five one Asterisk says hey! That
matches _NXX! -- the fact that you have five seven six left means
nothing, just as you can dial 1-800-PROGRESSIVE as Eric stated earlier.
On analog Zap interfaces, Asterisk (just like the telco) simply listens
until the digits match. If you don't want a ten digit number to match, then
adjust your dialplan accordingly. This is not a strange error in Asterisk,
it is a mismatch between what you want the system to do and how the system
operates.
Digital Zap channels and VOIP channels do not work this way because the
entire number is sent in one go -- when you dial from a SIP phone,
Asterisk does not see a stream of digits, it sees one message or packet
of information with the entire phone number in it. That is why it doesn't
match with SIP or IAX or PRI channels. (overlap dial excepted.)
-A.
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RE: [Asterisk-Users] Asterisk Voice Problems
Hi, Have you try to install this TDM400P card on another asterisk server? Same problems? Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shyam Gopale Sent: Thursday, April 27, 2006 5:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Voice Problems Hi, I am running Asterisk 1.2.1 using Digium TDM 400P with 4FXO lines to connect to the PSTN world. But, I constantly get clipped voice whenever there is a call placed using Zap channels. I have tried it all the recommended solutions - turned off all non essential services on the machine - ran fxotune - Changed IRQ settings But nothing works. The only thing that works is reducing the rxgain to around -20. But this leads to other issues like the hangup on the PSTN line is not detected by Asterisk. Anyone have a clue about how to fix the bad quality problem. Any help will be highly appreciated. Thanks, Shyam ___ --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 ___ --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] Excessive Asterisk delay to answer on ZAP inboundcall
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Wednesday, April 26, 2006 4:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inboundcall Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __Asterisk fxo line -| -Analog phone The analog phone rings immediately when calling, while asterisk shows the message Starting simple switch on zap... after the first ring and executes the old extension script after the second ring (for example a NoOp instruction). Why does Asterisk wait for these two rings? What is it doing meanwhile? Is it possible to shorten this interval to have an immediate response? TIA Giorgio Incantalupo ___ --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 ___ --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] Excessive Asterisk delay to answer on ZAP inboundcall
Hi, I have make some test. If asterisk can decode the callerid. The asterisk will answer the call after 2 rings. But when asterisk have some problem to get the callerid. Asterisk pickup the call after 3-4 rings. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Wednesday, April 26, 2006 4:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inboundcall Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __Asterisk fxo line -| -Analog phone The analog phone rings immediately when calling, while asterisk shows the message Starting simple switch on zap... after the first ring and executes the old extension script after the second ring (for example a NoOp instruction). Why does Asterisk wait for these two rings? What is it doing meanwhile? Is it possible to shorten this interval to have an immediate response? TIA Giorgio Incantalupo ___ --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 ___ --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] Re: Pattern matching problem
Hi,
Same dial pattern on my extension.conf, But it's work great. The Asterisk
only match 7 digits number. My * version is 2.1.6.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
Sent: Thursday, April 27, 2006 8:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: Pattern matching problem
Thanks, but the problem's with the first extension:
exten = _NXX,1,NoOp(Number dialed ${EXTEN}) exten =
_NXX,n,Dial(Zap/1/${EXTEN})
The problem is I _do_ get a match as you can see by the CLI output, but it
shouldn't match IMO - 6137451576 shouldn't match _NXX but that line gets
executed.
There was a cut/paste error with the others BTW. I thought I'd replaced the
defines with the actual numbers for clarity, but I made a mistake. They are
actually this way in my plan:
exten = ${LD_PATTERN},1,Dial(Zap/1/${EXTEN})
exten = ${INT_PATTERN},1,Dial(Zap/1/${EXTEN})
Thanks,
H
On 4/26/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
1) Your exten = _1XX,n,Dial(Zap/1/${EXTEN}) does not start
with priority 1 so it will never match
2) The 10 digit number you dialed does not start with a 1 so it will
never match, even if the priority issue is fixed.
Asterisk knows that once you've dialed 7 digits no OTHER pattern can
match what you are dialing and so it matches the 7 digits you dialed.
For the most part exten = i is only run during IVR (WaitExten,
Background, etc) and not when dialing from a phone.
BTW, this works just like the Telco. You can dial as many extra
digits as you want, and the telco will ignore the extra ones, which is
why you can dial 1-800-PROGRESSIVE it will work (assuming such a number
exists).
hugolivude wrote:
I'm running Asterisk 1.2.7.1 on Red hat 9 and have a strange pattern
matching problem:
I have the following in my dial plan:
exten = _NXX,1,NoOp(Number dialed ${EXTEN})
exten = _NXX,n,Dial(Zap/1/${EXTEN})
Unless I'm missing something, I wouldn't expect the pattern above to
match a 10 digit number, but when I dial 6137451576, I see the
following in the CLI:
-- Executing NoOp(Zap/1-1, Number dialed 6137451) in new stack
-- Executing Dial(Zap/1-1, Zap/1/6137451) in new stack
As you can see, the last 3 digits are truncated in the dial cmd.
This is odd behaviour isn't it? _NXX shouldn't be a match for a
10 digit number!
The other patterns I have are:
exten = _1XX,n,Dial(Zap/1/${EXTEN})
exten = _011.,n,Dial(Zap/1/${EXTEN})
so in fact I would have expected 6137451576 to fall thru to here:
exten = i,1,AbsoluteTimeout(15)
exten = i,n,Playtones(congestion)
exten = i,n,Congestion
exten = i,n,Hangup
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
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RE: [Asterisk-Users] Dialing Ring Groups from the Digital Receptionist-
Hi, I only check the AAH AMP. The inbound routing from-pstn didn't include the context ext-group. So the ring group setting doesn't work when you call from PSTN. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, April 25, 2006 11:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dialing Ring Groups from the Digital Receptionist- Hi! I've got a number of extensions (about 50) on a working Asterisk setup. For each user, I have two extensions configured (for example 11021 for a Cisco 79XX phone and 11022 for X-Lite), and a ring group that ties the two extensions together (for example, 1102). Reason being that if the user is away from his/her desk or working offsite, they can answer the soft phone on the PC. From an inside SIP extension (say 11071) I can dial 1102 and have it ring both 11021 and 11022, and this setup works well. But when I call the external number and get the digital receptionist, I cannot dial 1102 and have it ring both extensions - I have to either specify 11021 or 11022. So my questions: 1. Clearly it is possible to setup an option in the digital receptionist and have it dial 1102 (press 3 for Bob - dial 1102), but this doesn't scale well for 50 users. So is there a way to dial 1102 from the digital receptionist and have it ring both 11021 and 11022? 2. Is there another way to accomplish what I'm trying to do, ie. have two extensions per user, then dial them both simultaneously, and leave it up to the user to decide which one to answer - and do this from a phone NOT connected to the VoIP system? I appreciate your responses. Thanks- --Maxx ___ --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 ___ --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] wellgate FXO unit
Yes, just set the hotline number to an extension number. And disable the welltech IVR function. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Artifex Maximus Sent: Tuesday, April 25, 2006 3:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] wellgate FXO unit May hotline function will help. I never been use with Asterisk just with Welltech FXS device so it's just a hint. artifex On 4/21/06, Jerry Geis [EMAIL PROTECTED] wrote: Anyone know how to set the wellgate unit so incoming calls pass on directly to asterisk? Right now incoming calls ring twice and I hear a recording saying enter the extension. If I go enter the extension it goes on to asterisk just fine. I just want the incoming call to go directly onto asterisk. Anyone found that out? Jerry ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to select Ceptral's Voice in Asterisk'sSwift application??
Hi, Check the script. You can assign the voice by -n option, e.g., /opt/swift/bin/swift -n Diane Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shane Young Sent: Friday, April 21, 2006 9:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to select Ceptral's Voice in Asterisk'sSwift application?? Quoting Pimjai Wesnarat [EMAIL PROTECTED]: Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? [cepstral-demo] exten = s,1,Answer exten = s,n,wait(1) exten = s,n,Cepstral(voice name=DuchessHello and welcome to the world of text to speech using Cepstral. My name is Duchess./voice) exten = s,n,Cepstral(voice name=WalterHello and welcome to the world of text to speech using Cepstral. My name is Walter./voice) exten = s,n,Cepstral(voice name=ShoutyHello and welcome to the world of text to speech using Cepstral. My name is Shouty./voice) exten = s,n,Cepstral(voice name=WilliamHello and welcome to the world of text to speech using Cepstral. My name is William./voice) exten = s,n,Cepstral(voice name=WhisperyHello and welcome to the world of text to speech using Cepstral. My name is Whispery./voice) exten = s,n,Cepstral(voice name=RobinHello and welcome to the world of text to speech using Cepstral. My name is Robin./voice) exten = s,n,Cepstral(voice name=LindaHello and welcome to the world of text to speech using Cepstral. My name is Linda./voice) exten = s,n,Cepstral(voice name=EmilyHello and welcome to the world of text to speech using Cepstral. My name is Emily./voice) exten = s,n,Cepstral(voice name=DianeHello and welcome to the world of text to speech using Cepstral. My name is Diane./voice) exten = s,n,Cepstral(voice name=DavidHello and welcome to the world of text to speech using Cepstral. My name is David./voice) exten = s,n,Cepstral(voice name=DuncanHello and welcome to the world of text to speech using Cepstral. My name is Duncan./voice) exten = s,n,Cepstral(voice name=DamienHello and welcome to the world of text to speech using Cepstral. My name is Damien./voice) exten = s,n,Cepstral(voice name=CallieHello and welcome to the world of text to speech using Cepstral. My name is Callie./voice) exten = s,n,Cepstral(voice name=DogHello and welcome to the world of text to speech using Cepstral. My name is Dog./voice) exten = s,n,Cepstral(voice name=AmyHello and welcome to the world of text to speech using Cepstral. My name is Amy./voice) This message was sent using IMP, the Internet Messaging Program. ___ --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 ___ --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] Sip.conf
Hi Check this setting: bindaddr = 0.0.0.0 :IP Address to bind to (listen on) kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Par?ina Sent: Tuesday, April 18, 2006 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sip.conf In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network can register with specific user? The thing is that I can't use password and I can't use host=ip.of.my.phone. And I have to be sure that no one, from Internet will register on my * like that user. So, please tell me how to do this? -- Tomislav Parcina tparcina#lama.hr ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk hardware for new office suggestion
Hi, We setup a HP DL360G2 (Xeon 2.8*2, 2G RAM)server Digium TE110P (E1 PRI to telco) for a small office. Include 70 IPF-3000 phone in office 10 phones on another warehouse. Between the office and warehouse we use the G.SHDSL in bridege mode to connect each other. I suggest you can setup a small lab to test the autoprovision on the phone. Include the config file and firmware upgrade. E.g, The ipf-3000 can download the config files new firmare from the tftp server. But this phone always waiting user to press '1' for upgrade new firmware. It's to hard to upgarde 80 phones. So only 1 model phone for office and test the autoprovision functions. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simone Sent: Tuesday, April 18, 2006 2:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk hardware for new office suggestion I want to thank you for the suggestions. The office is in the UK, so probably we will go for the ISDN30. I am trying to get a SDSL 2mbit for the line so that bandwidth should not be a problem, the internal LAN will be Gbit as said so the QoS as suggested will be only on the firewall (linux). I have lowered expenses for other equipment so I was thinking of buying a Dell 1800 or 2800 server 2x2,8Ghz 2gb ram to set up Asterisk, know this is a big server but they'll use the ISDN lines and VoIP so virtually there could be 20/25 simultaneous calls. I'll have a look at the wiki and the phones suggested, we'd definitely like phones with internal ethernet switch and PoE capable, I'll try to get an idea of what could work for us. Thanks again Simone Tim Panton wrote: On 14 Apr 2006, at 11:29, Simone wrote: Hi list, I am in the process of setting up Asterisk for a new office and since this is going to be my first real installation I'd appreciate some advice on the hardware from the real world. We will have 8 channels (still not sure if 4xISDN2 or ISDN30 8 channels, but I will definitely go for a Digium card with echo hw cancelation) and a DSL 2mbit line (QoS on the switch and firewall?), to be configured for both traditional and VoIP usage . I was looking at the Xorcom TS-1 server and I was wondering if you would recommend it for a 30 employees office or if you'd rather build it on a normal server (would a double PIII 1Ghz be enough), and also if you could give a suggestion on the phones (we will get an HP Gbit switch PoE). Thanks, any hint really appreciated Simone I can only base my advice on what we have done for a smaller office. If you want 8 lines it is probably as cheap to go for ISDN 30 as for 4xBRI at least it is here in the UK. We have a single span E1 card from digium without echo can in a small 1U rack mounted server (spec: 1Ghz Via processor and 512Mb ram). The Via might be a bit underpowered for 30 users, but unless you are transcoding, virtually any modern processor would be fine for 8 lines. You need to look out on the DSL line if it is ADSL, since they have low upstream bandwidth. Heavy outgoing mail messages (eg attachments sent to distribution lists) can easily fill the outgoing (256kbit/s) pipe degrading the voice quality. I'm very fond of the SNOM phones - elmeg are selling the old SNOM 190 model which is a decent office phone. For 30 you should be able to get them for less than £70 each. I've got 6 - 4 SNOMs and 2 elmegs - No problems with any of them, but they don't support PoE, so you may want to look at other models. Don't underestimate how much training/doc you will need to provide to get people going on the new system. They may have been using the old one for years and written little cribsheets about how to transfer etc. Tim Panton [EMAIL PROTECTED] ___ --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 ___ --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 ___ --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] voicemail email-from
Hi, Check the vm_general.inc file Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Sunday, April 16, 2006 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] voicemail email-from How can I change this: Asterisk PBX [EMAIL PROTECTED] to: London PBX [EMAIL PROTECTED] ?? I tried several settings in voicemail.conf, without success! bye Ronald Wiplinger ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium TE110P TDM400P - always found new hardware on CentOS 4.3
Hi, I have try install the TE110P or TDM400P on HP Proliant DL380 Server. And use the AAH 2.7 distribution. When I reboot the server. The CentOS always display some hardware removed - Tiger pci card. Found new hardware - Tiger pci card. Is it the digium cards have some compatiable problems with hp DL380 server? Appreciate for any input. Thanks a lot. Kevin ___ --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] Display Confideltial or unknown on called iddisplay
Hi, Have you try to set hidecallerid=yes in zapata.conf? Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andre Courchesne - Consultant Sent: Friday, April 14, 2006 12:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Display Confideltial or unknown on called iddisplay Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display Confidential or unknown as we sometimes see ? Andre ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie MOH and call transfer question
Hi, I use the AAH2.7 (asterisk version 1.2.5). When someone call me and I pickup the phone. If I want to transfer to another extension. Then I dial the # key the system will play the onhold music. After I dial the extension number. The system stop play onhold music and play ringtone. Is it possiable keep play onhold music until someone pickup the phone? Appreciate any input. Thanks. Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to CCM4 SIP Trunk one-way audio problem.
Hello,, I have some of asteriskto CCM4 SIP trunk oneway audio problems. I have setup a asterisk server. It's work great and have no any problemsconnect to local ITSP(using SIP protocol). But we need to build a sip trunk to another CCM4 server. The network typology like this. SIP Phone (192.168.1.100) --- 192.168.1.254 (Asterisk Server) 59.124.xx.xx [INERNET] - PIX firewall - CCM4 1.The CCM4 side user can call to my asterisk extension and work great. 2.When I make a call from SIP phone to CCM4 extension. The CCM4 user can pickup the call but he can'thear me. It's seems have an one-way audio problem. The PIX firewall already mapping the TCP/UDP 5060 and UDP 16384-32768 to CCM4 Server. I have try to install another sipphone usingpublic IP. It's same problems. Anyonehave any suggest on the settings? My sip.conf configuration: [ccm4] context=.. canreinvite=nohost=the_ccm4_server_ipnat=noqualify=yestype=peer I have change a lot of settings. canreinvite=yes or no, nat=yes/no qualify=yes/no. It's doesn't work. Thank youfor your help. Best regards, Kevin ___ --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] Double Call Progress tones
Hi, I have the same problem on TE110P and Taiwan telco PRI line. I think to fine tune the rigntone frequencies not resolve this problem. For example. When I make a call to mobile. I can hear one ringtone like geneate by asterisk or device. And another ringtone like from telco. You known, some mobile will pickup the call and play music before user really answer the call. So I can hear music and mix with ringtone. Regards, kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simone Cittadini Sent: Wednesday, March 22, 2006 11:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double Call Progress tones Ron Wellsted ha scritto: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this suggests the call progress tones are being generated by the SIP device. As a result I have edited sip.conf to set progressinband=never but this has made no difference (even after a total restart). Previously I was running 1.0.7 without this problem, I upgraded to fix a problem with Monitor (the call stopped monitoring when transfered, 1.2.5 has fixed this). Does any one have any suggestions? Configure the ringing frequencies on the sip devices so that is something not udible by human hears (we did that as a quick fix before discovering progressinband some time ago, worked for linksys pap) ___ --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 ___ --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] Double-ring tone
Hi Thanks a lot. It's work for my ArtDio IPF-3000 phone. I have make a lot fine tune on the zapata.conf file. Doesn't have any help. Just add progressinband=no in the sip.conf. Done! Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: Thursday, March 16, 2006 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double-ring tone Could be the same problem I had with my Aastra - progressinband=no worked for me. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 15 March 2006 18:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double-ring tone Not sure it's that weird :O Douglas Garstang wrote: The phone must have transported you to Australia... :) -Original Message- From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 15, 2006 10:05 AM To: asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Double-ring tone I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on 7-4. TIA. Julian ___ --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 ___ --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 ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --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 ___ --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] Cisco Callmanager integration with asterisk
Hi,
The CCM4 behide the PIX firewall?Have you open the
ports for SIP trunk on CCM4 side? (TCP/UDP 5060, UDP
16348-32768)
Kevin
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antonio
Serrano LuqueSent: Wednesday, March 01, 2006 4:18 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Cisco
Callmanager integration with asterisk
HelloWe have integrated cisco callmanager 4.1 with asterisk
and we can dial from cisco to asterisk but we're getting an error if we call
from asterisk to callmanager. This is the error I'm gettinganybody can
help me? Verbosity is at least 3 -- Executing
Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack --
Called cme-pbx/4455 -- SIP/cme-pbx-25ae is
circuit-busy == Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion("SIP/2234-e084", "") in new
stack == Spawn extension (default, 4455, 2) exited non-zero on
'SIP/2234-e084'extensions.conf[cme-pbx]exten
= _4XXX,1,Dial(SIP/cme-pbx/${EXTEN}) exten =
_4XXX,2,Congestionsip.conf[cme-pbx]type=peercanreinvite=yeshost=XX.XX.XX.XX;
This is the callmanager IPRegards.
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RE: [Asterisk-Users] Problems registering Linksys SPA941 with * via SIP
Hi, "Admin Login" "Ext 1" "Proxy and Registation / Proxy" enter the asterisk ip address here. Kevin From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damian FunnellSent: Tuesday, April 11, 2006 8:14 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionCc: James HsiaoSubject: [Asterisk-Users] Problems registering Linksys SPA941 with * via SIP Hi all,Having trouble registering Linksys SPA941 with our * box. We can't find an entry in the config screens to allow us to put the IP address of the SIP server (i.e. the * box) in.We can find an entry for a SIP proxy in the phones set up, but we're not using one (SIP connections are direct across the LAN). Phone using static IP and no DNS present on network.Appreciate any help.Cheers,Damian. ___ --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] App Page() in 1.2.5
Hi, It's work on my spa-941. I just add belowing line before dial the extension. exten = s,3,SIPAddHeader(Call Info: Anwser-After=0) ; This is for the Snoms and Others Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Tuesday, April 11, 2006 3:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] App Page() in 1.2.5 Anyone have this working with the Sipura 841? I can page the phone, but it auto answers the page with dial tone, which isn't heard by the paging phone, John Novack Alexander Lopez wrote: Please look at: http://www.sineapps.com/news.php?rssid=1130 SNIP... ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco Callmanager
Hi, Our company replace the CCM3 with Asterisk this month :-) Take a look here: http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Voic email+Integration Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry Wade Sent: Tuesday, April 11, 2006 3:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and Cisco Callmanager Hi Guys I have just come from a customer that is looking to install 13 Cisco CallManagers into all their branches, (i tried to convince them to go *). They are looking for a voicemail solution. Now as Kinesis and Unity are way too expensive (apparently cisco is launching a cheap voicemail system too) I was thinking of installing * as the voicemail solution. Lots of goggling i have found plenty information telling me that this is possible. I just wanted to know if there are any success stories out there and whether or not i need any additional hardware, other than a PC. I know that one could use ztdummy as a timing source. Is this the best way, or is there some wise words out there one should be heeding. TIA Cheers Terry ___ --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 ___ --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] App Page() in 1.2.5
Hi, // sipura spa-941 auto-answer paging test exten = *63,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = *63,2,Dial(SIP/203) exten = *63,3,congestion I have make some test to paging extension 203. It's work on SPA-941. Can you test on your SPA-841? Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack (port) Sent: Tuesday, April 11, 2006 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] App Page() in 1.2.5 kevin ling wrote: Hi, It's work on my spa-941. I just add belowing line before dial the extension. exten = s,3,SIPAddHeader(Call Info: Anwser-After=0) ; This is for the Snoms and Others Kevin Is the mis spelling of Answer required to make it work? Problem in my case is it DOES auto answer, but the phone receiving the page also has dialtone out of the speaker, whereas the party sending the page does not hear it. How to make the phone NOT play dialtone? John Novack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Tuesday, April 11, 2006 3:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] App Page() in 1.2.5 Anyone have this working with the Sipura 841? I can page the phone, but it auto answers the page with dial tone, which isn't heard by the paging phone, John Novack Alexander Lopez wrote: Please look at: http://www.sineapps.com/news.php?rssid=1130 SNIP... ___ --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 ___ --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] Psgw
Hi, I have download the uplink and test with skype 1.4 2.0. not lucky to me. Only connect on first call then hang. I need to reboot my windows xp everytime. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Wednesday, March 29, 2006 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Psgw Haven't tried this product myself, but according to their spec it's only 1 call. There's another free SIP-Skype gateway from www.nch.com.au called uplink. http://www.nch.com.au/skypetosip/index.html Giordano Grandis wrote: Hi all, anyone never used PSGW as gateway beeween * and SkyPe? If yes, how does it works? How many session could I have on a single user ? Thanks all Giordano Thanks This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. --- - ___ --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 ___ --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 ___ --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] Uplink Skype2Sip
In my remember. The uplink install a virtual sound card. So uplink can auto answer the call from skype or sip side and redirect to another side. No matter what kind of onboard audio card do you have. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: Saturday, April 08, 2006 2:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Uplink Skype2Sip I cant make the proggie link my sip to skype, but skype to sip work great. Im running winxpsp2 with a cheapo onboard sound card. On 4/7/06, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all, anyone get it worked ? Uplink route me the call incoming from skype but when i answer, my skype go in error on sound card ? I also set in my hosts this value: 127.0.0.1 pgp01.televolution.net 127.0.0.1 stun01.sipphone.com This is my sip.conf [skype] language = it username = skype secret = password host = dynamic defaultip = lan_ip_address_of_uplink port = 5060 type = friend context = from_eth canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 fromuser = skype_username insecure = very qualify = yes callerid = Test 999 allow = all Thanks all Giordano ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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 ___ --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] Planet VIP-320 DECT gateway with Asterisk?
Hi Sorry for my chinese engish first. The VIP-320 seems like a SIP ATA+DECT Phone product. Please check you have registered to the asterisk server first. Because the VIP-320 built-in H.323/SIP dual mode. The web config tool not so clear. Reference the Page.31. Input the SIP:asterisk_ip_address in the *1 field. And try to sip debug peer and capture the sip message in the CLI mode. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Louis-David Mitterrand Sent: Thursday, April 06, 2006 11:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Planet VIP-320 DECT gateway with Asterisk? Hello, I just received what seems to be a nice SIP-DECT gateway but can't make it work with asterisk. The manual is very unclear (written in chinese english) and the web configurator is ambiguous as well. Has anyone succeeded in making one of these babies work with * ? info: http://www.planet.com.tw/product/product_dm.php?product_id=367menu_id=3 ___ --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 ___ --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] welltech Wellgate 3804 in SIP mode
For 3804 default setting. Each 3804 fxo port can register to asterisk server. And you can dial each extension and hear the second dial-tone. But you can config the 3804 to one-stage dialing and just pass the phone number to 3804. The 3804 will callout directally. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: Friday, March 31, 2006 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] welltech Wellgate 3804 in SIP mode one-stage calling function? On 3/30/06, kevin ling [EMAIL PROTECTED] wrote: Yes, Same configuration as Martin. 1.for incoming call just set the 3804 hotline to one sip extension number. 2.for outgoing call, you just using regular dail command to pass the phone number to 3804 (3804 is a 4FXO port device, the call from ip side always pass to FXO Port). You can telnet to the 3804 and enable the one-stage calling function. Regards, Kevin From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Sunday, March 26, 2006 12:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] welltech Wellgate 3804 in SIP mode On Mar 25, 2006, at 6:26 PM, Erick Perez wrote: Martin, i guess im in dumb mode today because i don't get what you say, may also be because this will be my first welltech to configure. what im trying to do is: Sorry, I don't know if the 3804 actually uses a similar config setup to my 3701a, so it might not be you at all. remote_voip_gatewayasterisk---fox/fxs/and_internatio nal_voip_providers call will always go - this way, no incoming calls from the right of the diagram side to asterisk. Ok, then if it is similar to mine, all you need to do is set the SIP configuration screen for the wellgate. Look under SIP Config and security config. Then to dial, you just use (repeated from below): For outbound ones, I think I just have a regular old Dial(SIP/[EMAIL PROTECTED] in my dialplan where 2003 is the FXO port. Good Luck, Marty PS I had to reflash my device with the SIP formware, as it arrived with H323 firmware. On 3/23/06, Martin Joseph [EMAIL PROTECTED] wrote: On Mar 22, 2006, at 10:24 PM, Erick Perez wrote: Hi, does anybody have a working config or tips to connect the welltech wellgate 3804 (4fxo) unit to asterisk via SIP ? I think I register it via SIP with my * box, but when sending calls from * to the wellgate the unit does not pass the call to any of the fxo ports. I am using the 3701a, which is a 1 FXO 1 FXS deal. The trick for me was the routing table or something like that from the Web based configuration screen. There I changed the default for the FXO to point to IP, and the IP to default to the FXO. Then I also have the line configuration set so the extension I want to ring ie 2020 is the hotline for the FXO. For outbound ones, I think I just have a regular old Dial(SIP/[EMAIL PROTECTED] in my dialplan where 2003 is the FXO port. HTH? Marty ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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 ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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 ___ --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] Re: FreePBX AAH
AAH 2.8beta1 include the FreePBX. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Thursday, March 30, 2006 9:44 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: FreePBX AAH In article [EMAIL PROTECTED], Jim Houser [EMAIL PROTECTED] wrote: I wanted the user interface of FreePBX over what is provided in the latest version of AAH. I installed the latest version of AAH and then just installed FreePBX over the top. It went fantastic and I do like the FreePBX web interface better than the latest AAH. Presumably, the next version of AAH will include FreePBX. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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 ___ --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] wellgate 38XX FX FXS voip gateways with outgoingcall files
In my remember. You can't select the port by dial command. The welltech 38xx voice gateway pick a free port and dial-out. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Friday, March 24, 2006 5:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] wellgate 38XX FX FXS voip gateways with outgoingcall files I am interested in the wellgate 38XX FXO and FXS gateways (and other similiar units). My question is can outgoing call files use these devices??? Can I fashion an outgoing call file with a channel like: SIP/WellGate-1/5551212 (for the first port) or SIP/WellGate-2/5551223 (for the second port) TO these devices behave this way? Of course incoming calls I dont see as a problem. It's asking the devices to place calls for me and the give some automated message. Does this work? THanks, Jerry ___ --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 ___ --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] welltech Wellgate 3804 in SIP mode
Yes, Same configuration as Martin. 1.for incoming call just set the 3804 "hotline" to one sip extension number. 2.for outgoing call, you just using regular dail command to pass the phone number to 3804 (3804 is a 4FXO port device, the call from ip side always pass to FXO Port). You can telnet to the 3804 and enable the one-stage calling function. Regards, Kevin From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin JosephSent: Sunday, March 26, 2006 12:13 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] welltech Wellgate 3804 in SIP mode On Mar 25, 2006, at 6:26 PM, Erick Perez wrote: Martin, i guess im in dumb mode today because i don't get what you say, may also be because this will be my first welltech to configure.what im trying to do is:Sorry, I don't know if the 3804 actually uses a similar config setup to my 3701a, so it might not be you at all. remote_voip_gatewayasterisk---fox/fxs/and_international_voip_providerscall will always go - this way, no incoming calls from the right of the diagram side to asterisk.Ok, then if it is similar to mine, all you need to do is set the SIP configuration screen for the wellgate. Look under "SIP Config" and "security config". Then to dial, you just use (repeated from below):For outbound ones, I think I just have a regular oldDial(SIP/[EMAIL PROTECTED] in my dialplan where 2003 is the FXO port.Good Luck,MartyPS I had to reflash my device with the SIP formware, as it arrived with H323 firmware. On 3/23/06, Martin Joseph [EMAIL PROTECTED] wrote: On Mar 22, 2006, at 10:24 PM, Erick Perez wrote: Hi, does anybody have a working config or tips to connect the welltech wellgate 3804 (4fxo) unit to asterisk via SIP ? I think I register it via SIP with my * box, but when sending calls from * to the wellgate the unit does not pass the call to any of the fxo ports.I am using the 3701a, which is a 1 FXO 1 FXS deal.The trick for me was the "routing table"or something like that fromthe Web based configuration screen.There I changed the default for the FXO to point to IP, and the IP to default to the FXO.Then I also have the "line configuration" set so the extension I wantto ring ie 2020 is the "hotline" for the FXO.For outbound ones, I think I just have a regular old Dial(SIP/[EMAIL PROTECTED] in my dialplan where 2003 is the FXO port.HTH?Marty___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- ---Erick PerezLinux User 376588http://counter.li.org/(Get counted!!!)Panama, Republic of Panama ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] aah 2.7 / BRI
Hi, AAH 2.7 + Digium TE100P work great on our lab. Some example. file: /etc/zaptel.conf # Span 2: ZTDUMMY/1 ZTDUMMY/1 1 # Global data span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 file: /etc/asterisk/zapata.conf [channels] language=en context=from-pstn signalling=pri_cpe switchtype=national ... Depend on your telephone service provider, maybe you must change some parameters. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans Witvliet Sent: Tuesday, March 28, 2006 10:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] aah 2.7 / BRI First encounter with * Just downloaded installed aah-2.7 Started up AMP, but i can not find any reference towards isdn. I presume there has to be some configuration done for my Eicon-Diva-pro. Does aah actually support isdn-bri? On the mail-archive i found some references, but these are rather old ( they speak about the coming release of aah-2.1) aah-handbook (version 1.6) doesn't spill a single character about bri and tfot doesn't spill much paper of the subject either ;-( Any suggestions/pointers Hans -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --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 ___ --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] What SW/HW phones support sendtext feature (tryingto send speech recognition results back to user)?
In my remember, the artdio ipf-3000 phone support instant message. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman Sent: Thursday, February 23, 2006 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] What SW/HW phones support sendtext feature (tryingto send speech recognition results back to user)? Hi, we've proof of conecpt system for speech recognition on Asterisk. We would like to send results of recognition back to user in standard way. Currently we're considering using sendtext command and it works with Firefly. But I'm curious what soft or hard ip phones that can connect to Asterisk support such feature ? Also what softphone would be most suitable for further work in adding such feature and possibly something more in this field ? Any other good way to send results (text) back to user ? Thanks in advance, regards, Rob. ___ --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 ___ --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 config help? Wellgate 3701a
Hi,
I have another model 3702a (2FXS 2FXO) voice gateway. You can implement
one-stage dialing on this device.
1. using sip show peers to make sure two ports (1fxo/1fxs) was registered
to asterisk.
2. login 3701 web and change the defualt routing. The factory default is IP
- FXS port. Change to IP - FXO port
That's all. Just try Dial(SIP/[EMAIL PROTECTED])
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph
Sent: Sunday, February 26, 2006 3:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Newbie config help? Wellgate 3701a
Hi again,
Kind of sheepish about asking for help, as I have only spent a day banging
my head off this...
I got my new Welltech 3701a, 1FXS,1FXO gateway.
I flashed it with what is seemingly the appropriate firmware (SIP V1.04).
This seems to have gone ok, and it is now registering both ports ok with
asterisk. For 1 minute I thought I was home free and and everything was
just going to work like magic. Wrong.
Problem #1) With my previous FXO (HT-488) I was using the following in my
dialplan:
exten = _NX,1,Dial(SIP/@2003,60,D(w${EXTEN}))
The above doesn't work for the Wellgate. So I tried:
exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED]
This doesn't work either.
I am hoping to dial 7 digit numbers directly through the FXO.
Problem #2) Something is amiss with the DTMF. I can call in to asterisk
from my IAX phones and use Comedian mail fine (tones sent out of band). If
I dial the extension of my FXO I hear dial tone, but the DTMF tones aren't
heard, so I can't dial... Very odd, this also worked perfectly with the
HT-488
The documentation from Wellgate is a complete joke, with many choice
nonsense sentences.
If anyone has experience with this device, or ones that sound similar, I
would love some help and or ideas...
Thanks much,
Marty
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RE: [Asterisk-Users] Sendmail with exchange
Hi, Can you make some test to send voicemail to other mail account? (e.g, @yahoo.com, @hotmail.com...). If it's work. I think not a SMTP authetication problem. Or you can check the asterisk maillog first. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan Novak Sent: Saturday, February 11, 2006 5:42 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sendmail with exchange I am using Asterisk to send Voicemail out as Email. I am running into a problem I believe to be caused by the exchange server requiring SMTP authentication. I cannot get the sys admin's to turn it off. Does anyone know enough about sendmail to help me. I am assuming that the default mail client is sendmail. It will also send to other non-SMTP authenticated servers. Your help is much appreciated. Jordan Novak ___ --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 ___ --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] cisco 7940 firmware upgrade
Hi, I have success upgrade two 7960 phone from sccp to sip. Some tftp server doesn't work. You can try this tftp serverand post your tftp logs. http://www.solarwinds.net/Tools/Free%5FTools/TFTP%5FServer/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kris EdwardsSent: Thursday, January 19, 2006 1:01 AMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] cisco 7940 firmware upgrade Hi Ron, Thanks for the reply. I used your config and still no upgrade. Using that file, the phone doesn't ask for the 7960-font.xml, but rather it just loops between the .tlv and the SEPMAC.xml. It never requests OS79XX.txt. I'm starting to thing that contrary to what I've read, a blank CTLwhatever.tlv file is not sufficient. Do you (or anyone on the list) have a sample of 7960-font.xml 7960-tones.xml ? Thanks, Kris On 1/17/06, Ron Wellsted [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE-Hash: SHA1Kris Edwards wrote: I have been trying for sometime to upgrade this phone to either sip, or the latest sccp (at this point I don't care) with no success.It has an sccp image (don't know the version off the top of my head) that is incompatible with chan_sccp.When I boot up the phone, it asks the tfpt server for the .tlv file, then the SEPmac file which it receives successfully.It then asks for English_United_States/7960- font.xml which I don't have, nor can I find a sample of this file.If I send an empty file, the phone says something like Invalid Glyph and then asks for United_States/7960-tones.xml. Then I begin an infinite loop.If I don't have the empty 7960-font.xml, It asks for it a few times, then loops. I got this phone from ebay, so maybe that's why it's asking for 7960 files when it's a 7940.Or perhaps there is an error in my SEPmac file?Here's the contents of that file:- - - - 8 KrisYour existing SEPmacaddress.xml file has far too much in it.In orderto upgrade to SIP, just useDefaultloadInformation8model="IP Phone 7940"P0S3-07-5-00/loadInformation8 loadInformation7model="IP Phone 7960"P0S3-07-5-00/loadInformation7/DefaultThis should trigger the upgrade.HTH- --Ron Wellsted[EMAIL PROTECTED] http://www.wellsted.org.ukN 52.567623, W 2.137621 Linux Counter No. 202120FWD:519961 -BEGIN PGP SIGNATURE-Version: GnuPG v1.4.2 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.orgiQEVAwUBQ81XGUtP/KMNOfRbAQKPSggAogZnAM5MzVP6FZtiLBhQK5ywTa2rifLvu8l0o66YFYsaTidru4v8eTnvsTlhCVg9+G6X4QxoeJa4p/B995TEYT38yumWTX8r G0RrQbM/zNJOqk9G3Yk+NjH9BhfZfW5OZyjEqGFniX6Tq0jsSo/fhvyyibnufT6c J8wLvmNEU3IsFdKK72k/qIxHOgTipBAtmW0M5koG8gUqXq9orL4Q2OwHZTPQ3BGw9CiAUfs748Zspkg6ZBsEs1o+EWENoIW1/DWoVvNH2CDx2uFAs+zevF5ASHi3FQAyNv+xTUr+h+qV+DwJO0ZrVTSNgUDB+ifNaOshu1s2Pi0hfjPH94sGpg== =s9Wv-END PGP SIGNATURE- ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Ita erat quando hic adveni ___ --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] festival-script.pl... howto change language?
FYI, http://www.cepstral.com/ You can download the english and spanish voice files for test first. And modify the festival-script.pl to using cepstral swift program. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Otero Sent: Thursday, January 19, 2006 8:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] festival-script.pl... howto change language? excuse for to find a desperate solution... but i'm boried to spent hours... ;) not in asterisk !!! ;) i use the festial-script.pl of Donny Kavanagh... but i want to change the language that festival uses, depending on a variable for the callerid. English/Spanish How i can tell the script the correct voice that festival needs to use?¿ like --language spanish --language english ...in normal cmd Can you help me? Realy thanks _ Acepta el reto MSN Premium: Correos más divertidos con fotos y textos increíbles en MSN Premium. Descárgalo y pruébalo 2 meses gratis. http://join.msn.com?XAPID=1697DI=1055HL=Footer_mailsenviados_correosmasdiv ertidos ___ --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 ___ --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] attended call transfer
Hi, You need the unattended transfer (blind transfer) featuer. That implemented in Asterisk (#) button. Not attended transfer. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Artner Sent: Friday, February 10, 2006 8:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] attended call transfer Hi! I am new with asterisk and I have my first problem with the attended call transfer feature. When a call comes in, i take the call and i would like to transfer it. So I press the * button (mapped for the attended transfer in features.conf) and the number for the receiving extension. The receiving extension rings and the call can be taken there. So far so good. Now to my problem: If I hook on the handset BEFORE the receiving extension take the call, the caller from outside will be disconnected and the receiving extension stops ringing. Shouldn't the receiving extension keep on ringing until the call is taken? Independent of hooking on the handset or not! (as it is with the blind transfer feature) The incoming line and all of the extensions are POTS, connected on a tdm400p card. I use asterisk 1.2.4 and zaptel 1.2.3 Hope someone could help me. Thx, Tom ___ --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 ___ --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] Welltech USA? and Wellgate Products?
FYI, I have two welltech SIP products. 3702a (2FXS+2FXO) and 3804 (4FXO). And all upgrade to the 12/2005 latest firmware. Actually the new release firmware most of bug fixed. The FXO ports still can't detected the barecore callerid in Taiwan telcom. These two voice gateway work fine with the asterisk box. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Thursday, February 09, 2006 3:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Welltech USA? and Wellgate Products? On Feb 8, 2006, at 9:22 AM, Ariel Batista wrote: I normally don't like talking bad about products. But I would like to say that the Welltech/Wellgate are not products that are support to work with asterisk. I have invested many hours of work in getting there device to work with Asterisk. They do not. And also as of Last Nov. They told me that they did not plan on supporting Asterisk. Good luck if you are able to get them to work since they go and sell there product with other names please post the settings you get for them to work. I have 2 of them as paper holders. And since there really bad I will not even sell them on ebay. FYI, They released newer firmware as of 12/2005 that is supposed to make most of there devices Asterisk compatible. If you try it, please let us know... If you have the 3701a unit (1FXS 1FXO) or really any FXO unit that you want to get rid of, please contact me off list, and I will take my chances and experiment with it a bit. Thanks for the feedback, Marty ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk native sounds now available!
In my remember, when playback a file. The Asterisk will automatically choose the audio file with the lowest conversion cost. Not always looks the filename.gsm. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Thursday, February 09, 2006 5:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk native sounds now available! Yes you can copy them into the same directory as the current files. Kris recommends that you move your existing files for safety only. The mode (ULAW, GSM etc) is selected by Asterisk depending upon what mode the current caller is using. Have you noticed that you don't have to put a file extension on the end of a Playback instruction? This is because Asterisk looks for filename.mode when trying to play a file. In the event it can't find filename.mode it looks for filename.gsm. If the file it's playing is not encoded using the current mode it has to transcode the gsm file into whatever is required. This not only adds computing overhead to the call in progress but degrades the quality of the file as all such transactions are lossy. Understand? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --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] Welltech USA? and Wellgate Products?
FYI. The welltech 3702 (or 3804) latest firmware didn't have sip register problem on fxo port. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vahan Yerkanian Sent: Tuesday, February 07, 2006 8:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Welltech USA? and Wellgate Products? Martin Joseph wrote: Any feedback on this brand and in particular on doing business with WelltechUSA? Don't worry they're OK. I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I am hoping to replace the near worthless Grandstream HT-488. I'd personally recommend to get a Sipura SPA-3000 instead. You're going to have problems trying to register the FXO port with username/password into Asterisk. Last time I checked, both ports used same CallID, same with the rest of Wellgate products. This company is telling me that I need to wire $ directly into there bank account. Most unusual. We bought all the samples from them via wire transfer too. Samples are collecting dust now though. HTH, Vahan ___ --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 ___ --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] Web based SIP client
yes, it's work. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roberto PereyraSent: Thursday, January 12, 2006 8:15 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Web based SIP client HiI found this http://www.etntalk.com/callto/loginany/Somebody has used it?roberto 2006/1/11, Derek Whitten [EMAIL PROTECTED]: Miguel wrote: Roberto Pereyra wrote: Hi Someone knows a free web based SIP client for use with any provider ? Thanks roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte técnico ISPs Jabber ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi Roberto, im looking for a similar solution,i found this on the archives http://www.microappliances.com/site/html/index.php It seems very complete to me (look at the customers page), does anyone here have it in production? Any comment? thanks in advance --- Miguel ___ --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 There was someone here on the lists a while ago that had a java basediax client..might find it if you search the archives..--.-BEGIN GEEK CODE BLOCK-Version: 3.1GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w--PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y--END GEEK CODE BLOCK--.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID: [EMAIL PROTECTED]For reliable and professional DNS, use DNS Made Easy!http://www.dnsmadeeasy.com/u/14989 ___ --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] Welltech USA? and Wellgate Products?
Hi, Have you try to search on eBay? I found some welltech devices for sale. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Tuesday, February 07, 2006 3:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Welltech USA? and Wellgate Products? Any feedback on this brand and in particular on doing business with WelltechUSA? I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I am hoping to replace the near worthless Grandstream HT-488. This company is telling me that I need to wire $ directly into there bank account. Most unusual. Thanks for any feedback on this, Marty ___ --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 ___ --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] Free IAX login
Not sure answer your question? Try to write some html code and let user register the username password online. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy Sent: Tuesday, February 07, 2006 7:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Free IAX login how to set up iax.conf , so IAX clients with any user name and any secret can login to * ? ( no authorize for login ) ___ --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 ___ --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] TDM04B FXO [EMAIL PROTECTED]
Hi, In my remember. The AAH will automatic config the TDM04B card. try the CLI command"zap show channels" regards, kevin From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nelson GranadosSent: Tuesday, February 07, 2006 7:18 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] TDM04B FXO [EMAIL PROTECTED] I would like to install a TDM04B with [EMAIL PROTECTED] 2.4 and [EMAIL PROTECTED] 1.5 but I didn't finddocumentation about this installation. Thanks in advance, Nelson ___ --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] French and German translations?
Hi, http://www.voip-info.org/wiki/view/Asterisk+sound+files+international -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lang Sent: Monday, February 06, 2006 3:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] French and German translations? Hi, Are there good (and complete, also for the voicemailmain application) french and german translations available for Asterisk 1.2? -- Philippe Lang, Ing. Dipl. EPFL Attik System rte de la Fonderie 2 1700 Fribourg Switzerland http://www.attiksystem.ch Tel: +41 (26) 422 13 75 Fax: +41 (26) 422 13 76 Email:[EMAIL PROTECTED] ___ --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] Call Waiting x100P and Cisco IP Phone
Hi, In AAH, you can setup the Incoming Calls to ring your extension. Or to ring extensions in a ring group. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Smith Sent: Friday, February 03, 2006 6:53 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Call Waiting x100P and Cisco IP Phone OK I have looked everywhere and I can't get a clear understanding on how to do this. If I have an x100P card connected to my home phone line and I am receiving calls on my Cisco 7940 IP phone with a SIP firmware loaded on it. How can I send the hook flash to the x100P card to switch to the call coming in from the PSTN? I am using [EMAIL PROTECTED] 2.4. I can hear the call waiting tone coming over the line but the phone doesn't recognize it. Thanks ___ --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 ___ --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] XLite dtmf issue?
set dtmfmode=rfc2833 in sip.confand try again. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AislingSent: Wednesday, February 01, 2006 11:03 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] XLite dtmf issue? Hi, Im wondering if anyone has experienced an issue with the XLite softphone and asterisk accepting dtmf? I can listen to my voicemail perfectly from my hardphone. However when I dial the voicemail number from my XLite softphone and enter the password at the voicemail prompt, an error appears vm-incorrect and I get an Unable to read password message on the asterisk console. Has anyone experienced issues with XLite dtmf? Many thanks, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ --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] Gateways
Hi,
I didn't have this gateway, But on welltech 4fxo gateway. You can just dial
SIP/[EMAIL PROTECTED] Even the gateway didn't register to the
asterisk server.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Corne
Vermeulen
Sent: Monday, January 30, 2006 10:15 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Gateways
I have a DrayTek Vigor3300V gateway with 8 FXO ports. I am trying to
configure asterisk to dial out on the gateway. I have one of the FXO ports
configured on sip account 100. If I dial the sip account then the router
gives me dial tone, with which I can dial a number. Unfortunately this is
not the behaviour I desire. I want to setup the FXO port as a trunk with out
it giving me a dial tone. I have tried the Dial(sip/100, D(${NUMBER}))
command, but it doesn't work. Any ideas please.
Corne.
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RE: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM 4.0SIP Trunk
Hi, Maybe buy 7912 phone and register to CCM is another choice. or integrated CCM with asterisk voicemail system. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sys readSent: Tuesday, January 24, 2006 11:28 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM 4.0SIP Trunk Hi guys,I want to leave messages on our unity box. I have already converted a couple 7940s to SIP, but I can't give them out to our users because I don't want to have to deal with two voicemail systems.we have licenses for all our users on unity as is. we're about to buy a bunch more 7940s, but I don't want to cause they're expensive. I'd rather buy a cheaper SIP phone and have it rollover to the unity vm. On 1/23/06, Gary Richardson [EMAIL PROTECTED] wrote: You can run a SIP image on a 7940. [EMAIL PROTECTED] has pretty goodsupport for it. Check the voip-info.org wiki for instructions onswitching the firmware.Hopefully that will take a step out of the plan -- you could completely ditch your Cisco system :)On 1/23/06, sys read [EMAIL PROTECTED] wrote: Hi, I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and about 45 SCCP phones on the ccm, and 200 users on unity. we want to migrate all users to IP Phones to ditch our ancient phone system. I would love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet and run sip to an asterisk server, but have their voicemail on Unity. these phones are $150 each, the alternative is cisco 7940s ( around $250 ) running SCCP through the CCM.at the quantities I'm talking about, $100 is significant. Does anyone have any idea how to get this done? I've tried this: exten = 123,1,Dial(SIP/sipphone,20) exten = 123,2,Dial(SIP/ccm/3040) where 3040 is our VM pilot for ccm.but all it does is take us to the main greeting. we have smartnet, but they haven't been helpful at all I called digium to see if they could help if we paid, but they said they've never heard of cisco unity help? thanks. ___ --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 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] oh323 configuration
Hi, To call the extensions registered on Asterisk. You don't need th gatekeeper. In your H.323 devices just set the gateway to Astiersk IP. I have test on ooh323 channel drive netmeeting. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M Sent: Thursday, December 29, 2005 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] oh323 configuration El jue, 29-12-2005 a las 05:40 +0500, Rehan Ahmed escribió: Hi, What exactly would you like to do, how would you like asterisk to talk with GNUGK I'm a little confused about the use of ooh323. I want to register some elesign h.323 hardare with gnugk to call to sip devices conected with asterisk. It's possible ? Rehan On 12/28/05, Guillermo Salas M [EMAIL PROTECTED] wrote: It's possible to register oh323 with gnugk ? Any one knows one good oh323 how to? Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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 -- Rehan Ahmed AllahWala http://www.SuperTec.com - Tommrow's Technology, Today. http://www.didx.net - DID Number Exchange and Peering Service. -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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 ___ --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] OOH323 Configuration with Cisco FSX ports, no Gatekeeper
Hi, You mean Cisco FXS Port? Can you describe more detail about your network configuration? Regards, Kevin _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Timothy R. McKee Sent: Thursday, January 12, 2006 6:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OOH323 Configuration with Cisco FSX ports,no Gatekeeper Has anyone used the OOH323 driver to connect with the FSX ports on a Cisco router *without* the use of a Gatekeeper? If so could you share your OOH323 and Cisco configs? Thanks, Tim McKee File: ATT00246.txt attachment: winmail.dat___ --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] Why remotely reboot SIP phones?
Sometime are phone's configuration change. Because Cisco or Polycom sip phone download the settings from the tftp server after reboot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Langstaff Sent: Wednesday, January 11, 2006 11:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Why remotely reboot SIP phones? Do you mean changes to the phone's configuration, or changes to Asterisk's configuration? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Douglas Garstang Sent: 11 January 2006 15:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Why remotely reboot SIP phones? Polycom phones need a reboot after making configuration changes. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 7:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Why remotely reboot SIP phones? We have to reboot our phones sometimes when we do something server side, mainly because the cisco firmware doesn't seem to handle everything very well. Usually it's just to pull new configs though, as we test more features and roll them out. Aaron Steve Langstaff wrote: Over the last couple of weeks I have seen a thread about remotely rebooting SIP phones from Asterisk. Is there something inherent in Asterisk that *requires* that SIP phones to be rebooted in a particular scenario, or is it just so that phones can pickup new firmware and/or configuration from their boot server? TIA. ___ --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 ___ --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 ___ --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 ___ --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 ___ --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: RE : [Asterisk-Users] name that vendor...
The company name: WellTech, and the model number: WellGate 3804 (4FXO). Support H.323 or SIP. You can download these firmware from there site. http://www.welltech.com.tw I have one 3804 on my desk now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, December 31, 2005 7:13 PM To: 'Jeffery Chen'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE : RE : [Asterisk-Users] name that vendor... Sorry, but I don't remember the name of this chinese company. I have meet it once time at a Cebit exhibition at Hannover in Germany few years ago. Francois BERGERET, France. -Message d'origine- De : Jeffery Chen [mailto:[EMAIL PROTECTED] Envoyé : samedi 31 décembre 2005 10:26 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] name that vendor... yes, right ? do your who make this box ? On 31/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hey men, I know this box ! You can see them at : www.ges.fr/voip/ This gateways are exported from Taiwan by Micronet and probably other brand/company. This are made in China and work well (H.323/SIP firmwares). GES is a french distributor and can provide you with a lower price than displayed on their public osCommerce web site for integrators/resellers. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cory Andrews Envoyé : samedi 31 décembre 2005 04:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] name that vendor... Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --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 ___ --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 ___ --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 ___ --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 -- Jeffery Tel: 1-700-576-1311 FWD: 728150 ___ --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 ___ --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: RE : RE : [Asterisk-Users] name that vendor...
This device only support FSK (Bellcore) ETSI callerid. But I have the same problem and test with there RD now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Sunday, January 01, 2006 2:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: RE : RE : [Asterisk-Users] name that vendor... That's right, it's a welltech. I have one working but when people call in the ringing is not typical of American installations (indications?) and it freaks people out. Also, I don't get callerid. Where can I get the upgraded firmware? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --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 ___ --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] Getting Yoda unit to register all four ports
Hi, I download the guide from yoda site. It's seems the original vendor is Accel AmiGate Elite 400 (http://www.accel.com.tw/frame/frame_age400.htm) I have one H.323 model and can't upgrade to SIP firmware. So what is your firmware version? Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Friday, December 30, 2005 5:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Getting Yoda unit to register all four ports I have a sample of the Yoda VG400 and I am having a devil of a time trying to get all four channels to register to Asterisk. I have an Asterisk 1.2.1 server. I have tried adding one at a time and rebooting it, but it stops after the first. http://www.yoda.com.tw/model.php?type=Enterprise_VoIPpname=VG400 Anyone had success with this? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --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 ___ --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] Re: Re: Remotely reboot SIP Phones ?
Thanks a lot. It's work :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Tuesday, January 10, 2006 10:18 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Re: Remotely reboot SIP Phones ? Figured it out :) Basically, you have to have a file called syncinfo.xml in the tftp root directory, with the following contents: SYNCINFO IMAGE VERSION=* SYNC=1/ /SYNCINFO Also, in SIPDefault.cnf or the phone's configuration file, stick: sync: 0 somewhere so the phone's sync value doesn't match the value in syncinfo.xml. If you make a change of sorts, just run sip notify reboot-cisco username at any time in asterisk and it'll send the notify to the phone. If the phone is in use, it waits until it's idle, once it is, it waits 20 seconds and then checks the syncinfo.xml file, and if the values of sync are different, it reboots :) Aaron Tomislav Parcina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Yeah, that should theoretically work, but I've got about 60 cisco phones that don't respond to the check-sync. If you ever make it work, please anounce it on the group. ___ --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 ___ --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] Draytek Vigor 2900 Asterisk
Hi, Draytek 2900 is a great router. Easy to setup stable. I want known more detail of your network configuration. I can setup it and make some test. Regards, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Attwood Sent: Saturday, January 07, 2006 9:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Draytek Vigor 2900 Asterisk I'm in conversation with Draytek's pre-sales dept.. Here's the most recent reply: Hello, We really don't know of anyone who has run an Asterisk server on a Vigor2900. There are doubtless people around, but it's relatively rare. Most people don't run SIP servers. Regards, All I want to know is, if I buy one of these routers, will it break my setup or not - ie. assuming I set up the relevant port-forwarding, can I expect any one-way audio issues. Can't get a definitive answer from suppliers or the manufacturer, so I hope someone here uses this model with Asterisk.? ___ --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 ___ --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] Draytek Vigor 2900 Asterisk
Now draytek have some SIP embeded router (e.g., 2100VG, 2900VG...). Maybe you can try these new router. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergio Chersovani Sent: Saturday, January 07, 2006 9:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Draytek Vigor 2900 Asterisk Jonathan Attwood wrote: I'm in conversation with Draytek's pre-sales dept.. I bought a 2600 2 years ago and I had alot of NAT problem, because the SPI was changing the externhost (sip.conf) ip address with the local private address forwarding the packets, so the audio stream was failing. I sent all the debug logs to the draytek dev team, but they were slow on updates to I bought a new and different brand router. Hope they fixed that issue in the new firmwares Good luck Sergio ___ --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 ___ --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] wich IAX soft client allow to specify a differentserver port?
Try this. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antonio Gallo Sent: Saturday, January 07, 2006 8:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] wich IAX soft client allow to specify a differentserver port? I still having problem with remote SIP client, trying to use IAX client instead but i've to specify TCP port 8080 (because of firewall). I did this on server in bindport=8080 in iax.conf but i cannot find a soft client that allow to set wich server port to use. Any idea? Thanks, Antonio ___ --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 ___ --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] Re: [Asterisk-biz] (Fwd) hi there
It's the welltech wellgate 3804 4FXO gateway. More info: http://www.welltech.com/product_e_03.htm I have another model 3702 (2FXO+2FXS). Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta Sent: Monday, January 02, 2006 2:09 PM To: [EMAIL PROTECTED]; Commercial and Business-Oriented Asterisk Discussion Cc: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: [Asterisk-biz] (Fwd) hi there Hi, Not very reliable for commercial setups, they do have issues hanging up ports etc. Quintum over Antek any day. Regards, Sahil Gupta VoiceValley On Mon, 2 Jan 2006, Rehan AllahWala wrote: www.antek.com.tw Had 4 port fxo, for around 200 to 250$ They are OEM, and can change things if u need. I tested it breifly in there office last year in Computex 2005 You can contact [EMAIL PROTECTED] for wholesale. Rehan On Fri, 2005-12-30 at 17:53 -0800, [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? I bet it has an fcc id which can be looked up at fcc.gov. If you get the first 3 letters it tells you who the vendor is. Maybe a ruse about not believing that it has all those compliance certifications and you want to guarantee the FCC certification for use in the US ... I would google for the name on the sticker, which is 'fxo-04'. This returns people talking about teh Asotel(Dinamyx) fxo-04. There is also a 'stargate fxo-04'. On and on ... If I had to guess I would say it looks like: http://www.chinanetphone.com/newchanpin/fxo-04.asp or http://www.repotec.com/voip/RP_FXO02A.htm My guess is that you should be able to find out more on your own :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group --- End of forwarded message --- --- End of forwarded message --- Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk FXO Panasonic PBX
http://www.alibaba.com/catalog/10886425/Fxs_fxo_Port_Converter.html I have one and bad voice quality. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, January 02, 2006 8:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk FXO Panasonic PBX On 1/1/06, VoIP Newbie [EMAIL PROTECTED] wrote: There are 4 options for your consideration: 1. use 2 x 1-port FXO gateway 2. use 2-port FXS gateway with FXS to FXO converter What is an FXS to FXO converter? you have any URLs? 3. use a 4-port FXO gateway. 4. use 2 x X100P cards You can get them from www.broad-tel.com On 12/21/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I'm looking for a reliable 2 FXO-port gateway to connect a Panasonic PBX to Asterisk. Can anyone recommend a stable and reliable one? Thanks, Waldo ___ --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 ___ --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 ___ --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 ___ --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] offline g.729 transcoding
try this: http://www.asteriskguru.com/audio_conversion.php From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim HarrisonSent: Tuesday, January 03, 2006 10:52 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] offline g.729 transcoding I'm trying to get some of the sample asterisk gsm files into a g.729 encoding. Is there an offline way of doing this (without a specialized card?) Can someone point me in the right direction?Thanks,-Tim ___ --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] offline g.729 transcoding
FYI: http://redice.krisk.org/ g729: http://www.readytechnology.co.uk/open/ipp-codecs/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bogdan MoldovanSent: Tuesday, January 03, 2006 1:46 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] offline g.729 transcoding Hello Kevin, Matt Ridell also replied to a another message with this link (10x both)... But is there a way to do that using a command line like sox? Can sox enc/decode from/to g.729? WIth an external/builtin library? Or something similar to sox? Thanks, Bogdan MoldovanMODULO Consulting"The Future Is Not What It Used To Be"http://www.modulo.ro From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kevin lingSent: Tuesday, January 03, 2006 5:31 AMTo: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] offline g.729 transcoding try this: http://www.asteriskguru.com/audio_conversion.php From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim HarrisonSent: Tuesday, January 03, 2006 10:52 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] offline g.729 transcoding I'm trying to get some of the sample asterisk gsm files into a g.729 encoding. Is there an offline way of doing this (without a specialized card?) Can someone point me in the right direction?Thanks,-Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Christmas Help request
How do I change the time zone for Asterisk? Currently the system time is correct but when I dial *60 it reports a different time (out by many hours). In [EMAIL PROTECTED] console type config type to change time-zone ___ --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] Redency of Asterisk
Hi, Asterisk High Availability Solutions http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SteveSent: Saturday, December 17, 2005 10:14 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Redency of Asterisk Hi, i have two [EMAIL PROTECTED] 2.2 server. i want if one of my asterisk server down. other is taken control of my first server and call goes through. Is it possible in asterisk. ___ --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] Alternative source for Asterisk-IM
The hyperlink work now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Takayuki Uehara Sent: Friday, December 16, 2005 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Alternative source for Asterisk-IM I tried to download the Aserisk-IM software from the URL below but the server returns 404 not found response. http://www.jivesoftware.org/wildfire/plugins/asterisk-im.jar Does anybody know any alternative source for downloading Asterisk-IM? Thanks in advance, Ooey -- Takayuki Ooey Uehara [EMAIL PROTECTED] 090-1426-4482, Skype ID: tuehara ___ --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 ___ --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] sharing a line w/multiple extensions
Have you try first blind transfer to a meetme meeting room. Then multiple user can join in. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: Thursday, December 15, 2005 3:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sharing a line w/multiple extensions Sean Cook wrote: also you can ring multiple extensions: Dial(SIP/101SIP/102SIP/103) I have that but once one extension picks up, others can't join in. Well, at least when I tried it with mixed SIP and Zap, it didn't work. Maybe all SIP does but I need it to work for all phones SIP and analog (via Zap). ___ --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 ___ --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
