[Asterisk-Users] Does Atcom AU-200 work with XLite?

Would appreciate any learnings on the AU-200 model. Thanks. Melisa. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user

RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

On Thursday, March 09, 2006 8:18 AM Douglas Garstang wrote: > By 'code for asterisk' are you referring to the Asterisk source code? > If so, step back and think about your statement for a moment. If, for > Asterisk to be enterprise class, it's source code needs to be > modified from it's current

Re: [Asterisk-Users] Any way to change dns timeout value? Asterisk hangs if internet unreachable

The real solution is to implement asynchronus DNS. We are looking into doing that with the C-ares library. No promises yet, it all depends on funding for this development work. If anyone is interested in funding it, please contact me off list. /O --- * Olle E. Johansson - [EMAIL PROTECTED] *

Re: [Asterisk-Users] Real Time Asterisk

8 mar 2006 kl. 17.07 skrev Fernando Lujan: Hi guys, I want to setup a environment where asterisk load all information from a Postgresql database. So here goes my questions: 1) Is real time asterisk stable enough? 2) Where can I found documentation about using it with Postgresql? ( inclu

Re: [Asterisk-Users] parking slot lights - testers wanted

8 mar 2006 kl. 15.19 skrev Dr. Michael J. Chudobiak: Hi all, The "metermaid" patch allows you to use the programmable buttons and LEDs on phones (like the Snom 2xx or 3xx) to view the status of parking slots and transfer to them. This should be really useful for small-office environments

RE: [Asterisk-Users] res_mysql.conf & DNS SRV lookup

If you read the list, you will see that several people have noted the exceedingly long time for posts to appear in the list today. -Original Message- From: Matt Riddell [NZ] [mailto:[EMAIL PROTECTED] Sent: Wed 3/8/2006 11:47 PM To: Asterisk Users Mailing

RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

By 'code for asterisk' are you referring to the Asterisk source code? If so, step back and think about your statement for a moment. If, for Asterisk to be enterprise class, it's source code needs to be modified from it's current content, it's hardly enterprise class, is it? If 'code for asteri

Re: [Asterisk-Users] sending text to display of sip phones

9 mar 2006 kl. 07.50 skrev Matt Riddell [NZ]: Alejandro Vargas wrote: I red that it is possible to send instant messages to the displays of sip phones. How can I do it using Asterisk? You can either do sendtext from an agi on that channel, or using my new patch ( http://bugs.digium.com/vi

Re: [Asterisk-Users] Multiple Subscriptions to SIP accounts at Same Domain

THis is too hard to solve in Asterisk, even though it can be solved. I've answered the question far too many times to answer again - search the mailing lists and the wiki and you will find out how to work with peer matching to fix this issue. In the "sipregister" development branch I am wor

Re: [Asterisk-Users] Professional Recordings

Waldo Rubinstein wrote: > Can anyone recommend a company that does professional Asterisk > recordings for things like IVR, greetings, MOH, announcements, etc? http://www.digium.com/index.php?menu=product_category&category=thevoice -- Cheers, Matt Riddell

Re: [Asterisk-Users] Re: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO???

Dan Miller wrote: > So, when I get no comments on this at all, either here or on any of the > forums, does that mean nobody knows what I'm talking about?? Or does nobody > know the answer?? Or is it just a stupid question and nobody wants to bother > telling me where to look?? > > It *is* a q

Re: [Asterisk-Users] What port mpg123 uses for MoH?

Zach A wrote: > Hi, > > What port does mpg123 uses to play music on when it starts MoH after > asterisk has put called on hold? As far as I'm aware it writes to standard output and reads from standard input (i.e. no ports involved) -- Cheers, Matt Riddell __

Re: [Asterisk-Users] parking slot lights - testers wanted

Dr. Michael J. Chudobiak wrote: > Hi all, > > The "metermaid" patch allows you to use the programmable buttons and > LEDs on phones (like the Snom 2xx or 3xx) to view the status of parking > slots and transfer to them. This should be really useful for > small-office environments. > > Anyway, the

Re: [Asterisk-Users] sending text to display of sip phones

Alejandro Vargas wrote: > I red that it is possible to send instant messages to the displays of > sip phones. How can I do it using Asterisk? You can either do sendtext from an agi on that channel, or using my new patch ( http://bugs.digium.com/view.php?id=6131 ), you can do it from the manager in

Re: [Asterisk-Users] res_mysql.conf & DNS SRV lookup

Douglas Garstang wrote: > Good grief! I posted the message below at 1:17pm... and it appeared on the > list after 8pm. > Nice I just posted mine and it arrived 30 seconds later...from New Zealand. Maybe your mail servers are b0rk3n: hehe :D It varies from time to time, but the mails do te

Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

Douglas Garstang wrote: > Asterisk calls the Business Edition 'enterprise grade'. It's right there on > the Digium website. It's the same dang code as the open source version, just > older. We are using it successfully in quite a few enterprise roll outs. If you are unable to, maybe you should

Re: [Asterisk-Users] Memory Problems

Something like: up2date -i kernel-hugemem Then make the appropriate changes in /etc/grub.conf, reboot, and see if it works. Of course, that's an overly simplified explanation, if this is a production system please research this first. If it's a test system, well what's the worst that could happ

Re: [Asterisk-Users] [Slightly OT] Does TE110P (a 32-bit PCI) fit into PCIe x8 slot?

Josip Gracin wrote: Does TE110P (a 32-bit PCI) fit into PCI Express x8 slot? It turned out that it doesn't. Which leaves me with the question: does Digium produce PCI-express cards? ___ --Bandwidth and Colocation provided by Easynews.com -- Aster

RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe

Best of luck :-D I would be interested in your progress on this. I am having very little problem in convincing ppl to upgrade their multiple BRI cricuits for a single pri. The cost difference between a te110 (or a Sangoma A101) MORE than covers the difference from the customer stand point, especi

Re: [Asterisk-Users] Memory Problems

So how do I enable a High mem Kernel? Do i have to recomplile the kernel to use highmem ??On 3/9/06, Joseph Tanner < [EMAIL PROTECTED]> wrote:The answer's just below the part you bolded.  "Use a HIGHMEM enabled kernel." Joseph TannerOn 3/8/06, Dumpolid Exeplish <[EMAIL PROTECTED]> wrote:> Hello,> T

[Asterisk-Users] re: Billing Package for Asterisk

Hi, Can anyone recommend a good billing package for use with Asterisk? We would prefer something that has a Customer and Provider web interface/access.   Thanks,   Bruce VIC IP Communications   ___ --Bandwidth and Colocation provided

Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

Actually its hardware related. On 3/8/06, Nick Hoffman <[EMAIL PROTECTED]> wrote: > On Thu March 9 2006 03:43, Warren Burstein <[EMAIL PROTECTED]> wrote: > > I have a Linksys PAP2. Identical setups for the two channels in both > > the unit and in Asterisk. In particular, both channels enable g72

Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

Joseph Tanner wrote: The PAP2 can only handle one g729 call at one time. Whether that's a hardware limitation, or licensing, or both, I don't know. Joseph Tanner Hardware. The PAP2 (and SPA2000) can only do one g729 call at a time. Any other call will have to use g711. -- Kristian Kielhof

Re: [Asterisk-Users] System Design

Colin Anderson wrote: This doesn't directly answer your question, because every integration scenario is different, but one of the nice things about Asterisk is that the barrier to entry to get a system working and play around with it is very low. What you might want to consider doing is get you

Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

Warren Burstein wrote: I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (an

[Asterisk-Users] Re: [asterisk-biz] Professional Recordings

http://www.mikesullivan.com/ http://thevoice.digium.com/ On Wed, 8 Mar 2006, Waldo Rubinstein wrote: Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? ___ --Bandwidth an

Re: [Asterisk-Users] MWI, SER and asterisk

On Thu March 9 2006 08:52, [EMAIL PROTECTED] wrote: > Hello, > > You can use ser as an outbound sip proxy and asterisk > as a register server . > > Your sip agents will get MWI, ... > > Harry Hi guys. With that solution, remember that Asterisk can handle a fraction of the number of registrations

RE: [Asterisk-Users] HW Echo Cancellers

Forget the orion.lots of DTMF problemstech support is not Terribly well spoken. Look for ANY of the 257* series... Just ebay for "t1 echo" -D -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, March 08, 2006 2:30 PM To: Aster

RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe

I have received the card. It comes with some closed source capi drivers, which I haven't tried as I don't believe that is in acceptable solution anyway. I had a look at hacking qozap to make it work, but haven't gone there at the moment. What I'm looking at now is visdn. 0.14 doesn't even want to

RE: [Asterisk-Users] Dial command

Exten => 222,1,Dial(SIP/polycom601||20) Exten => 222,2,Dail(Zap/2/ww09123456789# > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Ronald Wiplinger > Sent: Wednesday, March 08, 2006 5:26 PM > To: Asterisk Users Mailing List - Non-Commercial Discuss

[Asterisk-Users] Openline4 and [EMAIL PROTECTED]

Hi all, Can somebody help me get my OpenLine4 card running with [EMAIL PROTECTED] I've got my VPB drivers configured, but can't figure out how to map trunks and channels the typical way in the AMP config interface for [EMAIL PROTECTED] Apparently I'm supposed to use /vpb/1 type commands, but

Re: [Asterisk-Users] Is everyone getting mails except me?

Ron McCarthy wrote: I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? About once a week for the past three weeks I've experienced periods of time where no mail is received from the Asterisk mailing list. After

Re: [Asterisk-Users] Reverse group in zapata.conf

> I have a situation where I have 8 lines from the phone company in a hunt > group coming in to my asterisk box. These are the same lines I'm using > for outgoing calls ( named g0 ). > Is this possible? If it isn't, I plan to reverse the order in which the > lines are connected to my * box, ha

Re: [Asterisk-Users] HW Echo Cancellers

Matt wrote: Tellabs looks a little too up-scale for what I need :). $1k for a single port orion unit might be worth considering for really stubborn installs though. Why? they go for around $100.00 on eBay. What goes for $100 on eBay? Tellabs? or Orion? I can't find any Orion eq

RE: [Asterisk-Users] No DTMF

Try dtmfmode=info and see if that works.   Mark   -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, 9 March 2006 6:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] No DTMF   Some one was on my server making changes to my sip.

Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

On Thu March 9 2006 03:43, Warren Burstein <[EMAIL PROTECTED]> wrote: > I have a Linksys PAP2. Identical setups for the two channels in both > the unit and in Asterisk. In particular, both channels enable g729 and > set it as the preferred codec, and have disallow=all and allow=g729 in > sip.conf

Re: [Asterisk-Users] Memory Problems

The answer's just below the part you bolded. "Use a HIGHMEM enabled kernel." Joseph Tanner On 3/8/06, Dumpolid Exeplish <[EMAIL PROTECTED]> wrote: > Hello, > This is not a question directly related to asterisk. > I am currently rinning ansterisk on a Debian server and i just upgraded my > memor

Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

This ATA can only do 1 g729 call at a time. The sipura 2002 is the same way. It's outlined in the datasheet. On 3/8/06, Warren Burstein <[EMAIL PROTECTED]> wrote: > I have a Linksys PAP2. Identical setups for the two channels in both > the unit and in Asterisk. In particular, both channels ena

Re: [Asterisk-Users] Upgrading Asterisk witk G729 license installed

I would think that it would be OK to upgrade, but to be sure, your old license file should exist at /var/lib/asterisk/licenses/G729-.lic and could be backed up from there. After the install, copy this back in. And make sure you still have your codec_g729.so file to put in the modules

Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

The PAP2 can only handle one g729 call at one time. Whether that's a hardware limitation, or licensing, or both, I don't know. Joseph Tanner On 3/8/06, Warren Burstein <[EMAIL PROTECTED]> wrote: > I have a Linksys PAP2. Identical setups for the two channels in both > the unit and in Asterisk.

Re: [Asterisk-Users] Is everyone getting mails except me?

I have recived 7 mails since that time this morning GMT+10 Ron McCarthy wrote: I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -

Re: [Asterisk-Users] Is everyone getting mails except me?

On Thu March 9 2006 02:14, "Ron McCarthy" <[EMAIL PROTECTED]> wrote: > I havent got any mails since 2:42 this morning..usually i get at least > the normal 10-15 a hour, if someone gets this can they reply? > > Thanks! > Ron Hi Ron, I've received many emails from the mailing list over the past 24

RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

For the record, Douglas is correct on this point of "enterprise-grade" being on ABE: http://www.digium.com/index.php?menu=product_category&category=software Copied and pasted right from the website, it says: Asterisk Business Edition(tm) Digium(tm), the leader in open source telephony, offers Ast

Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3

Douglas Garstang, Your inability to keep your mouth shut (err hands closed when writing emails) is sometimes astonishing. On 3/7/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: > Docs? Polycom has docs? Where would one find this fabled land of... err I > mean Polycom does stating what ftp server

[Asterisk-Users] MINNESOTA: TwinCities Asterisk Users Group - Saturday 03/11/2006

SPONSORED THIS MONTH BY: SOUND CHOICE COMMUNICATIONS LLC "Keep in touch with the World" Hello, The next Asterisk Users Group meeting has been scheduled for this Saturday March 11th at 11:30am. Meetings are held monthly on the second Saturday of each month, excluding

Re: [Asterisk-Users] 160 analogue phones..

On Sun, 2006-03-05 at 16:05 +0200, Tele Cost Price Reducer wrote: > Conrad, > i would go with following solution: > 1. 6 sets of Audio Codes of 24 FXS ports conected by SIP accounts to > the system. the type is MP 124. then you open the conector on the > initial MDF and then the users have the same

RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel

I sent my reply to this to your off list email to me, which I greatly appreciate. We can send the results once we fix the problem, to the list? Take care, Sina -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Daragon Sent: Wednesday, March 08, 2006

RE: [Asterisk-Users] System Design

This doesn't directly answer your question, because every integration scenario is different, but one of the nice things about Asterisk is that the barrier to entry to get a system working and play around with it is very low. What you might want to consider doing is get your Asterisk box work

Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3

Azfhasterisk wrote: We had the same issue but we found that it was really the MS proxy server that the phone was going though. Set it up to use a different route out to the server and everything worked fine. Had to prove it to the admin at the location too, that was fun! Rick Rick, Even th

RE: [Asterisk-Users] Reverse group in zapata.conf

> Hey all, > > I have a situation where I have 8 lines from the phone company in a hunt > group coming in to my asterisk box. These are the same lines I'm using > for outgoing calls ( named g0 ). > > The problem arises when someone dials our number at the same time > asterisk tries to put a call

Re: [Asterisk-Users] System Design

Lot of questions, lots of variables, but I'll touch base on a few things. 5-10 concurrent calls is hardly anything. A plain T1 will more than handle that, even at ulaw or alaw (non)compression. Throw in a decent codec, and 10 calls won't even put a dent in your T1. Heck, it'd handle all 20 user

RE: [Asterisk-Users] Reverse group in zapata.conf

I think what your asking is pretty easy, just change the lowercase g in your extensions.conf file to an uppercase G. If you have a TRUNK type variable declared, this will be cake. If not you will need to change the little g, as in Zap/g1 to Zap/G1 everywhere you have it used. Hope that helped.

[Asterisk-Users] [EMAIL PROTECTED] Servers Connecting Portugal to Brazil (offices)

Hi all, I'm planning to connect 2 office from one company. I'm the developer, so i hope i can get all the features working well. [EMAIL PROTECTED](Portugal)-IAX2/[EMAIL PROTECTED](Brazil) 1- First i'm integrating Asterisk in Portugal's company office, one [EMAIL PROTECTED] with TE110P c

[Asterisk-Users] overlap dialing with polycom?

Where's the setting for overlap dialing with Polycom IP601? -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

[Asterisk-Users] Professional Recordings

Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? Thanks, Waldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or

[Asterisk-Users] Unicall, Fax and Echo cancellation

    Does Unicall support disabling echo cancellation on an E1 circuit when a fax tone is detected?  I think this is the reason why I cannot send or receive faxes on my Asterisk server. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-9

Re: [Asterisk-Users] sending text to display of sip phones

On Wed, 8 Mar 2006, Alejandro Vargas wrote: I red that it is possible to send instant messages to the displays of sip phones. How can I do it using Asterisk? Your phone needs to support it. Few do. -Dan ___ --Bandwidth and Colocation provided by Easy

RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

Asterisk calls the Business Edition 'enterprise grade'. It's right there on the Digium website. It's the same dang code as the open source version, just older. -Original Message- From: Matt Riddell [NZ] [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 07, 2006 6:48 PM To: Asterisk Users Ma

[Asterisk-Users] Re: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO???

So, when I get no comments on this at all, either here or on any of the forums, does that mean nobody knows what I'm talking about??  Or does nobody know the answer??  Or is it just a stupid question and nobody wants to bother telling me where to look??   It *is* a question that I have to ans

Re: [Asterisk-Users] MWI, SER and asterisk

Hello, You can use ser as an outbound sip proxy and asterisk as a register server . Your sip agents will get MWI, ... Harry --- Christian B <[EMAIL PROTECTED]> a écrit : > Hi Sharon! > > This is pretty difficult, i was not able to > implement it so far(though > my ser-skills are pretty basic).

[Asterisk-Users] Re: System Design

Date: Tue, 7 Mar 2006 18:26:12 -0500From: "Jason Adams" < [EMAIL PROTECTED]>Subject: [Asterisk-Users] System DesignTo: "Asterisk Users Mailing List - Non-Commercial Discussion"Message-ID:<[EMAIL PROTECTED]>Content-Type: text/plain; charset="us-ascii

Re: [Asterisk-Users] MWI, SER and asterisk

Method 3 is the one I was speeking of. As long as you plan to continue to have SER in front of Asterisk it should be fine. David On 3/8/06, Christian B <[EMAIL PROTECTED]> wrote: > Hi Sharon! > > This is pretty difficult, i was not able to implement it so far(though > my ser-skills are pretty bas

[Asterisk-Users] Random Zap port going crazy When channel released after a flash.

On 1.2.x I have a random problem when a Zap/x channel flashes to transfer or make a three way call. The Zap/x-2 channel is created and the transfer or three way proceeds, but on hangup the Zap/x-1 channel fails to destroy the old bridge and asterisk goes crazy logging the problem. Here is an e

Re: [Asterisk-Users] sending text to display of sip phones

I think its sendtext, to cinfirm do a show applications like text from the CLI On 3/8/06, Alejandro Vargas <[EMAIL PROTECTED]> wrote: > I red that it is possible to send instant messages to the displays of > sip phones. How can I do it using Asterisk? > -- > Alejandro Vargas >

Re: [Asterisk-Users] OT: Polycom Registration Weirdness

Are the Polycoms doing this on a different network than the Polycoms not doing this? On 3/7/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: > This is a SER/Polycom question, but I hoped we may have some SER guru's > here... > > I have a series of Polycom phones that are tying to register with Ope

Re: [Asterisk-Users] Re: ON DEMAND call Recording

Tomislav Parcina wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: 7. ozujak 2006 18:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: ON DEMAND call Recording On Mar 7, 2006, at

[Asterisk-Users] Dial command

I have an ZAP extension number 222 which is connected instead to a phone to a FXS/FXO converter and from there to a CDMA gateway. To dial my mobile phone I use: 222 (wait 2 seconds) 09123456789 I cannot figure out how to write this into the dialplan as a default number! 222 as above I will use

Re: [Asterisk-Users] Asterisk Prepaid Card

why not use astcc ? it comes with asterisk and does all that you have requested. we have scripts running. one that works via CID and one the user enters the number. --- leonimar cape <[EMAIL PROTECTED]> wrote: > Hi group, > > I am currently looking for a prepaid application > that > can do the f

Re: [Asterisk-Users] Asterisk download file locations

we mirror all the files our selves so our scripts work flawlessly. --- Alistair Cunningham <[EMAIL PROTECTED]> wrote: > This is a request to the website manager for > asterisk.org. > > The build scripts for our ITSP product include the > URLs to download the > Asterisk files, such as: > > wge

RE: [Asterisk-Users] res_mysql.conf & DNS SRV lookup

Good to know I'm not the only one... I thought perhaps I had been expelled from the list... Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Tuesday, March 07, 2006 10:44 PM To: Asterisk Users Mailing List - Non-Comme

[Asterisk-Users] Problem ChanSpy

Sorry, This is a mistake, sip.conf:   [302]canreinvite=no [301]canreinvite=no  Any idea? Thanks          ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visi

[Asterisk-Users] Any way to change dns timeout value? Asterisk hangs if internet unreachable

I don't have the most reliable internet connection in the world. Whenever it goes out, I can't receive any incoming calls at all, not even from pstn. When it first goes out I can still make outgoing calls through pstn, but eventually that fails too (as does voicemail, everything's out). Yes, ast

[Asterisk-Users] List Problems

Is anyone with a yahoo account having problems recieving emails from the list. I have not recieved any emails in about 8 hours and I posted something about 3 hours ago. If anyone knows please email to asteriskdigium _AT_ yahoo.com Thanks __ Do You Y

[Asterisk-Users] Calls forwarding to numbers only in user's context

Hello, I'm trying to do call forwarding based on this: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding In the extensions.conf file I have several context defined (local, longdistance, mobile, international and so on). Each user can be associated with different context (so can mak

[Asterisk-Users] More 7940 Questions

Does anyone know why putting an outbound proxy in the SIP.cnf file causes the phone to not pull it's logo from logo_url? Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visi

[Asterisk-Users] Cisco 7960 SIP - Displaying Time

Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information.   Thanks,   Ben Blakely ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or upd

Re: [Asterisk-Users] Calls between Asterisk servers using SIP?What about IAX (got it working w/ IAX but I have questions)

Does anybody have any experience with capabilities here?  I need to know if IAX is able to handle more than that.  I think I might just benchmark this with a bunch of .call files between servers to see how they are handled.   Any input?   - Gabriel Afana   - Original Message - F

[Asterisk-Users] Zap not installing

I have decided to move on from [EMAIL PROTECTED] and start compiling asterisk myself now. I got a dedicated box and put my X100P in it. I installed the server version of CentOS 4.2 (2.6.9-22.EL kernel) barebones. The box is a dell GX270 workstation with 1GB of RAM. I got a fresh copy of O’R

Re: [Asterisk-Users] HW Echo Cancellers

> > Tellabs looks a little too up-scale for what I need :). $1k for a > > single port orion unit might be worth considering for really stubborn > > installs though. > > > > Why? they go for around $100.00 on eBay. What goes for $100 on eBay? Tellabs? or Orion? I can't find any Orion equipment o

[Asterisk-Users] Faxing with MFC/r2

I am having a problem when trying to send a receive faxes on an E1 running with unicall on an asterisk 1.2.4 x64 server. The same server has a TDM02 card and if I send or receive faxes through there there is usually no problem. I am afraid that my customer insists that he wants to use the DI

[Asterisk-Users] Softphone for Windows CE 3.0

Hi, I've found several softphones for Windows Mobile 2003, but does anyone know of a softphone (or older version of a current softphone) that will run on Windows CE 3.0? ~ Matt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users ma

[Asterisk-Users] System Design

Hey Everyone,   We are in the works of planning a new * installation for our company.  We have 20 users in our main office and 5 users in a remote office a couple of states away.  Our call volume for the main office will be anywhere from 5-10 concurrent calls.  The remote office will have

Re: [Asterisk-Users] Setting Vaaibles

Figured it out. It was simple had to add Answer and hangupDovid Bender <[EMAIL PROTECTED]> wrote: Helo List,First I would like to apologize for my bad spelling aswell as that I did not search the wiki first. I onlyhave email access at the moment.I am having trouble setting both variables and glob

[Asterisk-Users] No DTMF

Some one was on my server making changes to my sip.conf files. I am now having trouble with DTMF. No matter what I use (inband,auto,rfc2833) the dtmf tones seem to not come thru. I compared it to the wiki and all the configs seem to be in order.   Here is my sip.conf   [general]disallow=all;all

Re: [Asterisk-Users] Polling Asterisk for Life

> > Are you guys perchance using Local/[EMAIL PROTECTED] in your installations? > > > > -- > > Cheers, > > > > Matt Riddell > > ___ > > > > Is there a known issue when using the Local/[EMAIL PROTECTED] > > thanks, This is how I would read it.. but yes..

[Asterisk-Users] Memory Problems

Hello,This is not a question directly related to asterisk.I am currently rinning ansterisk on  a Debian server and i just upgraded my memory from 1GB to 2GB. However, my linux OS does not recognise the memory upgrade. The BIOS does, but the Debian Linux refuses to use the entier memory, currently,

RE: [Asterisk-Users] Receiving Multiple calls on asterisk at home

Do you have call-limit parameter set to 3 in sip.conf or possibly sip_additional.conf on AAH? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rolf Brusletto Sent: Tuesday, March 07, 2006 1:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re

[Asterisk-Users] impact of qualify=yes

Anyone have any information on the performance impact of using qualify=yes for hundreds (500ish) of SIP UAs?   I have seen tidbits on qualifyspreading=yes, but not enough to understand what it does. I assume lessens the peak load of qualify sip options queries?   Thx!

RE: [Asterisk-Users] Receiving Multiple calls on asterisk at home

Do you have the phone specific config file for the polycom set to something like this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rolf Brusletto Sent: Tuesday, March 07, 2006 1:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk

Re: [Asterisk-Users] Changing REINVITE status of the channel dynamically

> I'd like to know if it's possible to set the REINVITE on or off dynamically, > based on the extension being dialed. Define two peers in sip.conf, one with canreinvite=yes and the second with canreinvite=no. Then you can route your calls with or without reinvites depending on the dialed number. L

[Asterisk-Users] Putting caller in queue and dialing an extension simultaneously

Hi,   Is it possible to do this in extensions.conf to put a caller in queue and dial an agent’s extension so that he knows that somebody is in queue waiting to be answered. This agent will be a remote agent and extension will dial his cell phone.   Thanks   Zach A.   __

[Asterisk-Users] What port mpg123 uses for MoH?

Hi, What port does mpg123 uses to play music on when it starts MoH after asterisk has put called on hold? Zach A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://l

RES: [Asterisk-Users] Inserting access codes as prefixes to CID

There’s the SetCallerID cmd that you should read about.   http://www.voip-info.org/wiki-Asterisk+cmd+SetCallerID   It has others links to clarify your ideas.   Tell us if you get something.     Filipe Mordhorst   Brazil-SC De: [EMAIL PROTECTED] [mailto:[EMAIL PRO

[Asterisk-Users] Upgrading Asterisk witk G729 license installed

I've an Asterisk 1.2.4 installation, where I've also installed the G729 codec license. I'd like to upgrade that installation to 1.2.5, but I'm not sure if I'll lost the license in the process (and if I'll be able to recover it later!!!). Is there any special consideration I've to keep in mind

[Asterisk-Users] PAP2 won't make two g729 calls at the same time

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we

RES: [Asterisk-Users] pap2 Dial plan

You’re almost right. The PAP2 has some features that are factory default. I don’t remember the section in the web interface, but here’s what you going to do:   Find the section that contains a lot of features name with values like this *56 or *78. Erase all of them. Letting’ this filled

[Asterisk-Users] Location of MeetMe Recordings

In Asterisk 1.2.4 is love being able to recording conferences. However, using the default variables, the files are being written to /var/lib/asterisk/sounds instead of /var/spool/asterisk/meetme. If I change MEETME_RECORDINGFILE variable to something different in works, bit I lose the ability to d

Re: [Asterisk-Users] grandstream handytone 286 sometimes dials out wrong number

Hi Martin, I have 3 choices on my ATA webpage and I chose SIP INFO: /Send DTMF: / in-audio via RTP (RFC2833) via SIP INFO This is the only point I can make changes since it is connected to my asterisk box through a TDM400P: asterisk box <--->TDM400P <-(telephone cable)-> HT-288 <---> LAN <--->

[Asterisk-Users] pickup last ringing phone

Hello,I am using pickup, i can pickup an extension from outside of the queue, but i cannot pickup any call comes to queue.queue strategy=ringallWhat is the problem with queue?Is there anyway to pickup last ringing phone?-erkaN Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breez

[Asterisk-Users] Cisco Call Manager SIP trunk + Asterisk

Hi,   I setup a SIP trunk in a brand new Cisco Call Manager and I try to place the calls using Asterisk… but I get error:   “<-- SIP read from 192.168.11.10:5060: SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 192.168.10.199:5060;branch=z9hG4bK2e7ca9c9;rport From:

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