[Asterisk-Users] Alarmreciver

Hi,Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked). Sometimes alarmreceiver is able to get some eve

[Asterisk-Users] iax2_poke_noanswer on IP change. Sometimes permanent.

I have 4 asterisk servers which are "Friends" and each one has an account for termination.  A total of 5 peers each. Currently, the setup is as follows iax.conf= [FriendName] type=friend context=server_friend secret=donttell host=friend.dyndns.com qualify=750 =

[Asterisk-Users] get no connection, very often, but not allways, why?

Hi, I have an ISDN phone connected to a hfc-s card. I use it to phone via an iax provider to foreign countries. Inside my country it works reliable, but to other country it happens very often that the other side hears ringing and before it can take the phone the line is dropped. What makes me wond

[Asterisk-Users] Any Polycom dealer willing to help?

Hi All, We are in search of the latest Polycom firmware SIP 1.6.5 and BootROM 3.1.3 as per http://www.polycom.com/resource_center/1,,pw-492,00.html Can someone help? We have legitimately obtained these phones but even our official distributor can't get their hands on updated firmware. The only th

Re: [Asterisk-Users] Any Polycom dealer willing to help?

Eric,     I have a copy of both.  They are at my office.  Send me an email directly and tomorrow I'll forward you a copy.   - Gabe     - Original Message - From: Eric Bishop To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, March 27, 2006 12

Re: [Asterisk-Users] Polycom IP 301 is slow

The worst thing on all Polycom IP phones is the speaker phone's poor quality. You could not have a conference call using the speakers, only the head phone. Denis. Hahaha, clearly this guy is on crack. (no offense) I have uploaded MP3s to my asterisk box and have it programmed to play th

RE: [Asterisk-Users] Alarmreciver

At the risk of being redundant, VoIP and Alarm is known not to mix well. Some of the tones used by an alarm system do not behave in the same way as conventional DTMF. This will vary greatly based on the actual alarm format used (and there are at least thirty different formats.) I don't know the

[Asterisk-Users] Caller ID length

What is maximum length of name in caller ID? How much charters can I put and be sure it will work fine? -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or updat

[Asterisk-Users] AstCC

Hi,I am wondering if it is possible with astcc to make a second call without hangup and be oblige to re-enter all the codes.Any idea how to do?Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

Re: [Asterisk-Users] 7940 with Asterisk?

Skeeve Stevens wrote: I just picked up a Cisco 7940 from an Auction… and would like to use it on an Asterisk box. Can anyone give me a pointer where I should start so I can get it working? http://www.voip-info.org Doug -- Ben Franklin quote: "Those who would give up Essential Liberty

Re: [Asterisk-Users] What codec extensions using now?

Yes. disallow=all allow=g723 Allow only g723 codec. roberto 2006/3/26, Mohammad Salaque <[EMAIL PROTECTED]>: Hello list,Another newbie question,.  if I put  "disallow=all" and  "allow=g723"my sip.cof  does it mean that  extension could only communicate usingg723 ?bellow is one of my extension e

[Asterisk-Users] RE: Snom 360 problems

Detailed info about snom beta firmware can also be found at snom-wiki e.g. http://snom.com/wiki/index.php/Beta_Firmware#Release_Notes Regards, - Usman Tahir snom technology AG ---

[Asterisk-Users] RE: Re: Best GUI for basic HostedPBX service

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > Please stop send me email > > Best Regards, > > Mr.Peeramate Rochanasmita > > Project Manager/General Manager This message was sent to me? -- Tomislav Parcina tparcina#lama.hr ___ --Bandw

Re: [Asterisk-Users] stop monitor on transfer

Anton Krall wrote: > Hi John, yes, Im using native transfer. What I do is use Monitor on the > dialplan of the extension that picks up the call coming from PSTN, so after > that, if the extension forward or transfers the call, monitor keeps > recording all thru the end of the call no matter where i

[Asterisk-Users] Polycom 501 Output volume

Hi Guys   Is there anyway to adjust the output volume on the Polycom 501? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/li

Re: [Asterisk-Users] RE: Snom 360 problems

5.5.1b is neither listed on the snom-wiki nor is any changelog for 5.5.1b listed. -Dan On Mon, 27 Mar 2006, Usman Tahir wrote: Detailed info about snom beta firmware can also be found at snom-wiki e.g. http://snom.com/wiki/index.php/Beta_Firmware#Release_Notes Regards,

[Asterisk-Users] registration with different username

Hello, I am trying to register to the asterisk with different phone number, login and password. This is my setting in the sip.conf: [246079011] type=friend context=cisco secret=XXX host=dynamic username=tomas allow=alaw nat=yes canreinvite=no mailbox=246079011 but I get this reply: Mar 27 13

[Asterisk-Users] Call Simulator

Guyz,   I wanna test my asterisk load capability before going to production, anyone know is there any call simulator to test this thing?   Thanks in advance,   Voipman ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] Transfer after group pick-up

I can't transfer call which was picked up with feature - group pick up. I'm running * 1.2.5. The problem is that asterisk doesn't "hear" that I have pressed #1 and doesn't play "transfer" sound for me. "Regular" phone calls I can transfer without problem. Can anybody check is this a BUG? -- T

[Asterisk-Users] Re: Re: Cisco 7960 - Have to press a menu button to dial

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > Absolutely right :) > > "\" escapes the next character, so if you wants *69 to go through > immediately, you'd put "\*69" so that the * gets recognized as a digit. > > "," returns the dialtone sound. When my users hit "9", they like to

[Asterisk-Users] Re: Free g729

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > There is no such thing as a 'free' G.729 - The DSP Group has claimed and > defended the Patents they hold against the algorithm and process. > > Please do not use Asterisk/Digium related resources to exchange this > information - They

[Asterisk-Users] Voicemail to Email

Could anyone provide me some link in order to  voicemail to email working, I believe I have to give SMTP settings but do not know where.   Thx   Voipman ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

Re: [Asterisk-Users] Voicemail to Email

Voicemail uses sendmail on your system. If your machine can send mails using sendmail, so will asterisk. Rudolf On 3/27/06, voipman <[EMAIL PROTECTED]> wrote: > > Could anyone provide me some link in order to voicemail to email working, I > believe I have to give SMTP settings but do not know wh

Re: [Asterisk-Users] Voicemail to Email

Its in vociemail.conf. If you built asterisk with a basic running config there should be examples in there. Dovid --- voipman <[EMAIL PROTECTED]> wrote: > Could anyone provide me some link in order to > voicemail to email working, I > believe I have to give SMTP settings but do not know > where

Re: [Asterisk-Users] iax limit question

> I found a solution... I just has to enter an Answer > line and now it behaves as I wanted. Here is the > working code: > > [inbound] > exten => 1234567,1,Set(GROUP()=limit) > exten => 1234567,2,GotoIf($[${GROUP_COUNT()}>2]?103) > exten => 1234567,3,Dial(Zap/5&Zap/6,25,tT) > exten => 1234567,4,Voi

Re: [Asterisk-Users] Polycom 501 Output volume

MBIT Technologies wrote: Hi Guys Is there anyway to adjust the output volume on the Polycom 501? Yes. I did this over the weekend. Look in your Polycom sip.cfg for a line tx.digital.handset. I had to set mine to -6 before the levels came down within tolerance. There is one for hands

[Asterisk-Users] Re: compiling Zaptel-1.2.4 on CentOS 4.3

I got past this by changing spinlock.h in the /usr/src/kernels/2.6.9-34.EL-x86_64/include/linux/ folder. (I am using 64bit kernel) I changed: #define DEFINE_RWLOCK(x) rw_lock_t x = RW_LOCK_UNLOCKED #define DEFINE_RWLOCK(x) rwlock_t x = RW_LOCK_UNLOCKED to: #define DEFINE_SPINLOCK(x) spinlock_t

RE: [Asterisk-Users] Re: Best GUI for basic HostedPBX service

You are signed up to the list. If you want out go to http://lists.digum.com --- "Peeramate @ SIPPhone Thailand" <[EMAIL PROTECTED]> wrote: > Please stop send me email > > Best Regards, > > Mr.Peeramate Rochanasmita > > Project Manager/General Manager > > SIPphone (Thailand) Co., Ltd. > 644/

[Asterisk-Users] automatic callback when busy

I'm trying to set up the following application: When a SIP extensions calls another one which is busy, the caller would be able to ask for an automatic callback: when the callee becomes available again, asterisk would ring both the caller's and the callee's phones and connect them when both par

RE: [Asterisk-Users] Call Simulator

SIPPS is one, I would like to hear of others.   Of course you could create a dialplan that loops calls in and out.   Thanks, Steve Totaro http://www.asteriskhelpdesk.com   From: voipman [mailto:[EMAIL PROTECTED] Sent: Monday, March 27, 2006 6:39 AM To: asterisk-users@li

Re: [Asterisk-Users] registration with different username

--- Tomas Komarek <[EMAIL PROTECTED]> wrote: > Hello, > > I am trying to register to the asterisk with > different phone number, > login and password. This is my setting in the > sip.conf: > > [246079011] > type=friend > context=cisco > secret=XXX > host=dynamic > username=tomas > allow=alaw >

Re: [Asterisk-Users] tsu-600

we are thinking about replacing a median 1 pbx system, we have about 40 phone. i got 4 incoming pot lines (all the same number), i don't know if i can use one tsu600 port as a fxo (for the pots) and all the rest as fxs, or should i use a tdm400p with 4 fxo's (for the pots,inside the asterisk bo

Re: [Asterisk-Users] Alarmreciver

Hi,Thanks for so fast reply.>Now, rather than just being a nay-sayer, let me refer you to the Bosch>C900V2 device.  It takes a signal from just about any panel and converts >it into IP to be received by a Bosch receiver.Is it possible to connect C900V2 with Asterisk, (did you do such a thing, did y

RE: [Asterisk-Users] Alarmreciver

The C900V2 only connects with Bosch receivers. In fact, all of the IP communicators in the industry are proprietary. There is a committee working towards a standard, but my understanding is that we still have a decent wait ahead of us. Bob McDowell From: [EMA

[Asterisk-Users] Re: Cisco 7970

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > Best bet is to get "Asterisk Chan_Sccp" http://chan-sccp.berlios.de/ > > 1.) setup your /etc/asterisk/sccp.conf with something like: > > 2.) setup lines 30/31 as a custom extension in astersik (i used amp) > and had it dial SCCP/30 and

Re: [Asterisk-Users] registration with different username

Well, I did, but the reason is still the same, if the username is different from the phone number, asterisk rejects the registration :-( Dovid Bender napsal(a): --- Tomas Komarek <[EMAIL PROTECTED]> wrote: Hello, I am trying to register to the asterisk with different phone number, login a

[Asterisk-Users] Timeout waiting for response to Originate

Hello,I am using Asterisk-java, the Manager. And I have a problem I don't know howto sort it out!:Sometimes, when I send an OriginateAction my code receives an exception withthis message: "Timeout waiting for response to Originate"I don't know what it means as Asterisk receives the action and then

RE: [Asterisk-Users] automatic callback when busy

I'm postponing this activity indefinitely but I collected some ideas. Try something similar to this recipe: First of all store dialed extension number as exten => _[2-8]XX,102,SetVar(${UNIQUEID}=${EXTEN}) exten => _[2-8]XX,103,Goto(busyphone,s,1) then you can use 3 optio

RE: [Asterisk-Users] stop monitor on transfer

Really? Mmhh seems you got working what I want and I what you want.. Hehehe try using monitor instead of mixmonitor.. Maybe there is a difference in apps. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |John Daragon |Sent: Monday, March 27, 2006 4:

RE: [Asterisk-Users] Polycom 501 Output volume

what do you men adjust? (I guess you already tried the keys on the pad right)? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MBIT TechnologiesSent: Monday, March 27, 2006 4:57 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Polycom 501 Outpu

RE: [Asterisk-Users] Any Polycom dealer willing to help?

If you purchased your phones from an authorized reseller they shouId be able to provide this.   I can help you.  Please contact me off list. -Mike   Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Eric Bishop [mailto:[E

Re: [Asterisk-Users] Alarmreciver

Quoting Andrew Nowrot <[EMAIL PROTECTED]>: > Hi, > > Did anyone try to set up alarmreceiver application over IP network? Which > ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. > Maybe I did something wrong with alarmreceiver.conf (I tried diverse > settings, but nothing w

Re: [Asterisk-Users] RE: Re: OT: Unblocking bloced CID

I have a regular PRI from our CLEC and I *do* get blocked numebrs.. the bit is set to tell me to hide the number. I definately (as the 'phone company') want to be getting all call data for tracing purposes, should we ever need it, but we can certainly honor that bit and not display the number. F

FW: [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?

Actually, I have tested this here with an Aastra 9133i and an [EMAIL PROTECTED] server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it

[Asterisk-Users] Bluetooth headset in handsfree mode with SJPhone or X-lite

Hi, After much searching I have found that it might be possible to get a bluetooth headset to answer/hangup with SJPhone or Xlite if the headset supports handsfree mode. My Toshiba bluetooth stack supports this but I have not been able to figure out how to enable it. Also Windows XP desktop b

Re: [Asterisk-Users] Polycom 501 Output volume

Anton Krall wrote: what do you men adjust? (I guess you already tried the keys on the pad right)? On my system, when you watch ztmonitor on a channel, it is maxing out the output volume, causing local side echo. Reducing the tx.digital.handset gain bring the graph down to an acceptable range

Re: [Asterisk-Users] RE: Re: OT: Unblocking bloced CID

On 3/27/06, Matt <[EMAIL PROTECTED]> wrote: > I have a regular PRI from our CLEC and I *do* get blocked numebrs.. > the bit is set to tell me to hide the number. I definately (as the > 'phone company') want to be getting all call data for tracing > purposes, should we ever need it, but we can cer

[Asterisk-Users] Re: IAX Incoming/Outgoing

> I could ask why it can't authenticate against the key, but we've already been > there. > > So, if I have 5 asterisk systems, and I want to have a different key on each, > and each system has a user and a peer section, and I have to use different > usernames... oh boy... this sounds like a horrib

RE: [Asterisk-Users] Bluetooth headset in handsfree mode with SJPhoneor X-lite

Try replacing the XP Bluetooth stack with the widcomm drivers...google is your friend! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, March 27, 2006 6:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [As

Re: [Asterisk-Users] RE: Re: OT: Unblocking bloced CID

On Monday 27 March 2006 08:58, Matt wrote: > Further, and here is where the legal question comes in. Is it legal to > 'unblock' to the end user, that blocked number?Personally I feel > it SHOULD be.. after all it is my time I'm about to spend picking up a > phone and talking to someone, I want

[Asterisk-Users] Small - Medium Billing Software needed

Hi,   Our asterisk installation will be a "man-in-the-middle" providing local,long,international VOIP services to our customers and our asterisk will be connect via VOIP to international carriers. We use asterisk 1.2.5 with mysql in centos 4.2 Kernel 2.6   I have looked at astbill and it sounds int

Re: [Asterisk-Users] RE: Re: OT: Unblocking bloced CID

Who it is legal for or not to display those numbers is not realy the point here, as in a law suit you will both (you and your provider) be held liable. But the law clearly states that the end user should NOT see that number if the number is blocked. On 3/27/06, Andrew Kohlsmith <[EMAIL PROTECTED]>

[Asterisk-Users] Who hangup.

Hello people. I`m running asterisk 1.0.9. In a phone call, I want to know who hangup, the caller or the callee. It this posible? Thanks in advance. José Luis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UN

Re: [Asterisk-Users] Caller ID length

For the US PSTN network the limit seems to be 15 characters. For Asterisk you can safely use 20 characters with most VOIP phones. MATT--- On 3/27/06, Tomislav Parčina <[EMAIL PROTECTED]> wrote: > What is maximum length of name in caller ID? How much charters can I put and > be sure it will work

[Asterisk-Users] after-queues

Hi,   I have the following requirement.. after a customer is answered by a Queue, I want him to be redirected to another extensions, where an IVR would answer and ask for his opinion about the analyst who just solved his issue.   Is there a way to redirect him automatically, or do I have to

Re: [Asterisk-Users] after-queues

On 3/27/06, Dov Bigio <[EMAIL PROTECTED]> wrote: > > Hi, > > I have the following requirement.. after a customer is answered by a Queue, > I want him to be redirected to another extensions, where an IVR would answer > and ask for his opinion about the analyst who just solved his issue. > > Is there

[Asterisk-Users] Config File Management

I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? Some sort of revision control such as cvs,rcs or subversion? A central 'config server' where you edit the files and then rsync them

[Asterisk-Users] Inaudible voice and sleepy voice

What could have caused a system(on the same side of NAT on our LAN ) that have been working perfectly ie you can call and both parties can hear themselves very well to start having the problem described below (1) the caller can hear the other party very well ,but the other party hears cracked

[Asterisk-Users] Background() App From AGI

I have the following python AGI script. I know it's been abstracted, but it's still pretty easy to see what's happening. self.agi.channelAnswer() self.agi.wait(1) self.agi.execCmd("background","enter-conf-call-number","") self.agi.execCmd("Read","confNum|||","")

Re: [Asterisk-Users] Config File Management

I'm using CVS. I only have one server right now. I use it on other clusters to sync files and it works for me.. On 3/27/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: > I'm curious (ok, well I admit it - it's for perosnal gain) what methods > people are using to manage asterisk config files when

[Asterisk-Users] Testing asterisk faxing functionality

I have asterisk with rxfax txfax modules.I want to test fax sendig and reciving in one asterisk instance, in extensions.conf I have : exten => 1234567,1,rxfax(/home/patryk/fax-new.tif|debug) exten => s,1,Dial(1234567) exten => s,2,txfax(/home/patryk/fax.tif|caller|debug) but I doesn't seem to w

RE: [Asterisk-Users] Re: Best GUI for basic HostedPBX service

FreePBX allows you to set up multiple companies as well as determine what level of access each user has. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com > -Original Message- > From: [EMAIL PROT

RE: [Asterisk-Users] Voicemail to Email

Just remember that a lot of email systems don't accept email from unverifiable domains. If your using a domain for your Linux/Asterisk server that does not resolve to a public IP then you may not be able to receive voicemail to email. I know that Hotmail WILL work no matter what so try that fir

Re: [Asterisk-Users] Testing asterisk faxing functionality

I was playing with the fax stuff over IP on Friday. Unless you're receiving faxes from a PSTN circuit, it doesn't work so well. Also, I don't think you can chain txfax and rxfax like that. When you hit the s,2 part, it's going to play the fax out to the handset you dialed from. You'll need somethi

RE: [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?

I tried again and you are correct. It does work on the Aastra 9133i but takes about an hour with no way to change that that I can find. The GXP2000 happens a lot sooner. I think it can be configured on the GXP2000. Turns out the problem I had is that the Aastra 9133i does not resubscribe to a

Re: [Asterisk-Users] Testing asterisk faxing functionality

You could always use System() to copy a call spool file to launch the outbound fax call. I don't really think a 3rd party app is necessary. -Corey On Mon, 27 Mar 2006, Gary Richardson wrote: > I was playing with the fax stuff over IP on Friday. Unless you're > receiving faxes from a PSTN ci

[Asterisk-Users] CLI Echo

Hello All: I used the Authenticate command against a list of 4 passwords, however is there anyway I can get these to echo in CLI> for debugging purposes? My auth line looks like this: exten => s,2,Authenticate(/home/listofnumbers|[|a]) ___ --Bandwidth

Re: [Asterisk-Users] CLI Echo

Jeremy wrote: Hello All: I used the Authenticate command against a list of 4 passwords, however is there anyway I can get these to echo in CLI> for debugging purposes? My auth line looks like this: exten => s,2,Authenticate(/home/listofnumbers|[|a]) show application NoOp __

[Asterisk-Users] Polycoms and hints

How does the hinting work on the polycoms? I've got a polycom set up with hinting, I can see when the shared line rings, but I can't tell if someone's on the line. Any suggestions? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 __

[Asterisk-Users] Authorization by ip

Can somebody send me a config of how to authorize SIP client by IP? Sam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asteris

RE: [Asterisk-Users] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2?

Can Somebody send a working instruction to me on how to install g729 and 9723.1? I could not open the http://aussievoip.com.au/tiki-index.php?page=G729-Install Thank you, Goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Peeters (Asteri

[Asterisk-Users] SIP caller id

I am using some Cisco 7940s with the 8.0 CM SIP image on them, and was wondering if there is a way to have the caller ID display as just ”NAME” number as opposed to ”NAME” [EMAIL PROTECTED].   The way it currently is, the missed calls directory can’t be dialed, and my users really want th

Re: [Asterisk-Users] Polycoms and hints

On 3/27/06, Aaron Daniel <[EMAIL PROTECTED]> wrote: > How does the hinting work on the polycoms? I've got a polycom set up with > hinting, I can see when the shared line rings, but I can't tell if > someone's on the line. Any suggestions? > Shared lines still don't work with Asterisk on the po

[Asterisk-Users] FreePBX & AAH

Does anyone know if FreePBX can be installed on a Linux box that was built using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are

RE: [Asterisk-Users] Installing the none commercial intel g729 codecsinto [EMAIL PROTECTED] 2.2?

http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ I think this give a pretty good how to on installing the g729 and 723. Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ADEGOKE ARUNA Sent: Tuesday, March 28, 2006 12:55 AM To: 'Asterisk Us

Re: [Asterisk-Users] Re: Cisco 7970

Yes, my mistake in /tftpboot/SEP.cnf.xml. Having said that, Please double check that you have set the line: permit=192.168.1.90/255.255.255.255 ; This device can register only using this ip address or in your case: permit=10.0.0.175 /255.255.255.255 ; This device can register only

Re: [Asterisk-Users] FreePBX & AAH

Yes, you can. On 3/27/06, Jim Houser <[EMAIL PROTECTED]> wrote: > Does anyone know if FreePBX can be installed on a Linux box that was built > using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over > the AAH build. I have just not had good luck building an Asterisk system >

Re: [Asterisk-Users] FreePBX & AAH

Worked fine for me. I did lose my MAINT link off the Portal, but I simply added it back. Thank you, Jyran Glucky Advisory Programmer BlueWare, Inc. Strategic HealthWare Solutions 3060 W. 13th Street Cadillac, MI 49601 Phone: (231) 779-0224 ext. 111 Fax: 231-779-1002 Skype: Jyran Glucky AIM: Jyr

Re: [Asterisk-Users] Bluetooth headset in handsfree mode with SJPhoneor X-lite

Hi, I am not having trouble with the bluetooth stack since the Toshiba stack has the headset profile which supports a subset of AT commands from GSM 07.07 for minimal controls including the ability to ring, answer a call, hang up and adjust the volume

[Asterisk-Users] Unicall Question

Hi    I have a litle question, what is then version stable, in the web server i can see  unicall version x.2.x and version x.3.x,  and the  time is  same unicall-0.0.2e/ 11-Nov-2005 18:33 unicall-0.0.3pre8/ 11-Nov-2005 18:37Where i can find the change log or the diference from t

Re: [Asterisk-Users] FreePBX & AAH

Pardon the question, but what I understand of FreePBX is that it's basically Asterisk with a web interface and some additional modules. Is that correct? Can you install FreePBX on a system which ALREADY has asterisk up and running or does it require ITS version of asterisk? Thanks, Waldo O

Re: [Asterisk-Users] automatic callback when busy

OK, if I see well, this is the key idea here: exten => 9,1,DBPut(CallBack/${${UNIQUEID}}=${CALLERIDNUM}) that is, putting the caller and callee number into AstDB under the CallBack family. Can you confirm that Asterisk takes care of the rest? If there is a record like this in the database wi

RE: [Asterisk-Users] FreePBX & AAH

My understanding is you can install it on any Linux server running Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Monday, March 27, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [A

Re: [Asterisk-Users] automatic callback when busy

How can I edit the DB? Tamás Bondár wrote: OK, if I see well, this is the key idea here: exten => 9,1,DBPut(CallBack/${${UNIQUEID}}=${CALLERIDNUM}) that is, putting the caller and callee number into AstDB under the CallBack family. Can you confirm that Asterisk takes care of the rest? If

RE: [Asterisk-Users] FreePBX & AAH

FreePBX is a configuration manager for Asterisk. It is NOT its own version of Asterisk, it is simply a GUI to manage the config files. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com > -Original Me

Re: [Asterisk-Users] * Meetme Freeze patch found

On Friday 24 March 2006 16:05, Benoit Panizzon wrote: > Hi all > > Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: > > http://bugs.digium.com/view.php?id=5884 > > Haven't tried it out yet. I can now confirm: No freezes/crashes anymore since I applied the patch. -Benoit-

[Asterisk-Users] Searchable forums

Where can I do a keyword search of the posting in biz and users forums? asterisk.org just links to http://lists.digium.com/pipermail/ and that doesn't let me do a string search across all postings.   thanks, -- ---Erick PerezLinux User 376588http://counter.l

Re: [Asterisk-Users] * Meetme Freeze patch found

I'm a bit newbie, could you tell me how to i apply the patch? Thanks in advance Marco Mouta On 3/27/06, Benoit Panizzon <[EMAIL PROTECTED]> wrote: > On Friday 24 March 2006 16:05, Benoit Panizzon wrote: > > Hi all > > > > Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: > > > > http:

Re: [Asterisk-Users] Config File Management

Sorry for thread breaking... I'm on digest. >> I'm curious (ok, well I admit it - it's for perosnal gain) what >> methods people are using to manage asterisk config files when they >> have multiple asterisk systems? > >I'm using CVS. I only have one server right now. I use it on other >clusters

[Asterisk-Users] Receptionist Phones (was 3Com Phones)

Thanks for all the comments on the 3Com phones. Thankfully, there is a large number of phones out there to dig through looking for the right solution. What I have not been able to find, after spending all weekend looking, is a good solution for an attendant console. We have 2 receptio

RE: [Asterisk-Users] * Meetme Freeze patch found

http://www.google.com/search?sourceid=navclient&ie=UTF-8&rls=GGLD,GGLD:2 004-48,GGLD:en&q=apply+patch+linux patch -p0 < patch-file-name-here Thanks, Steve Totaro http://www.asteriskhelpdesk.com > -Original Message- > From: Marco Mouta [mailto:[EMAIL PROTECTED] > Sent: Monday, March 27,

Re: [Asterisk-Users] automatic callback when busy

On Monday 27 March 2006 20.07, Daniel wrote: > How can I edit the DB? > This may be a starting point for you: http://www.voip-info.org/wiki/view/Asterisk+database Or the related section of the book Asterisk: TFOT http://safari.oreilly.com/JVXSL.asp?x=1&mode=section&sortKey=rank&sortOrder=desc&

Re: [Asterisk-Users] Searchable forums

Hi Erick - > Where can I do a keyword search of the posting in biz and users forums? > asterisk.org just > links to http://lists.digium.com/pipermail/ and that doesn't let me do a > string search across > all postings. I'm guessing you mean the mailing lists rather than the forums. If so, you

Re: [Asterisk-Users] Receptionist Phones (was 3Com Phones)

We would be interested in the same. We have had only limited success getting Snom's phones to do this. And, you're right, this is such a common thing, there MUST be something out there that can do the job. Darrell S. Long BestWeb Corporation Daniel Hazelbaker wrote: Thanks for all the

Re: [Asterisk-Users] Testing asterisk faxing functionality

"You could always use System() to copy a call spool file to launch the outbound fax call. I don't really think a 3rd party app is necessary." Could You explain this please? Or maybe some links to documentation and examples ? Thanks Patryk. ___ -

RE: [Asterisk-Users] Receptionist Phones (was 3Com Phones)

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Hazelbaker Sent: Monday, March 27, 2006 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Receptionist Phones (was 3Com Phones) Thanks for all the c

Re: [Asterisk-Users] Polycom IP 301 is slow

On Sun, Mar 26, 2006 at 08:03:55PM -0600, Darrick Hartman said: > Denis Galv?o - iSolve wrote: > >The worst thing on all Polycom IP phones is the speaker phone's poor > >quality. You could not have a conference call using the speakers, only > >the head phone. > > WHAT! The Polycom phones that h

[Asterisk-Users] Asterisk 1.2.6 and Zaptel 1.2.5 Released

The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.6 and Zaptel 1.2.5. Both of these releases include a number of important bug fixes, and users are encouraged to upgrade their systems when possible. See the included ChangeLog files for more details on what has been fi

[Asterisk-Users] Asterisk 1.2.6 and Zaptel 1.2.5 Released

The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.6 and Zaptel 1.2.5. Both of these releases include a number of important bug fixes, and users are encouraged to upgrade their systems when possible. See the included ChangeLog files for more details on what has been fi

Re: [Asterisk-Users] Searchable forums

Superb replies.   Thanks to Jon and Noah     On 3/27/06, Noah Miller <[EMAIL PROTECTED]> wrote: Hi Erick -> Where can I do a keyword search of the posting in biz and users forums? asterisk.org just> links to http://lists.digium.com/pipermail/ and that doesn't let me do a string search across> all

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