RE: [Asterisk-Users] Re: Junghanns and Digium TDM400?

On Thursday, March 30, 2006 8:27 PM Chris Earle wrote: > Someone plase help > This is terrible. Been trying everything for weeks :-( Have you contacted Junghanns directly? They do not read this list. Usually they are very helpful in such issues. Phone them. Mit freundlichen Grüßen Jan-P

[Asterisk-Users] Comparison of Business Edition VS Open Source

Hi,   I would like to find out;   Anyone using Asterisk Business Edition?   If yes, could you give us a brief description of what type of solution you use it for?   What made you to choose the Business Edition over the Open Source?   What advantage/disadvantage you have found?  

RE: [Asterisk-Users] Re: Building Asterisk embedded device

Anyone come across embedded device with FXO built in? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tomislav Parcina Sent: Monday, April 03, 2006 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Building A

[Asterisk-Users] Re: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries!

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > I never so this error. > I am using H323 with Asterisk 1.2.6 Any idea what can be the problem? You are using ooh323 from asterisk-addons-1.2.1? If so, you are experiencing the same but that I have. There is patch (which I can't apply) o

[Asterisk-Users] Re: ooh323 and g729 - any issue?

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > Guyz! > > Is there any known issue with ooh323 and g729? I am experiencing one side > voice okay from ooh323 extension to sip ext, but on reverse side voice > quality is very poor. Any thoughts? I have one question. Are you using ooh323

[Asterisk-Users] Re: oh323 - unable to install

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > I'm and [EMAIL PROTECTED] user - been so now for almost a year. > Lately, I've upgraded to the latest & greatest.. (which is built on 1.2.5) > and am unable to install oh323. I don't like to bring bad news, but I'm unable to install h323

[Asterisk-Users] Re: Building Asterisk embedded device

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > Hi, > > I want to build a PBX base on Asterisk using an embedded device. > Can anyone please recommend an embedded device I can use for doing so? > I will install linux or freebsd in the device. For now, I use Via EPIA PD6000. Pretty sma

[Asterisk-Users] Re: internals and ISDN calls fail when Internet is down

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > Hardcoding IP in /etc/hosts works but it's not reliable if voip > provider's IP change. > > Does someone already as a similar problem and resolved it? I head similar problem, but I didn't solve it. If you find workable solution, please

Re: [Asterisk-Users] Codec Problem

It seems that your asterisk cannot transcoding from ulaw to g729 and vice versa. What is the output from 'show translation' ? Did you allow both codecs in sip.conf ? Regards, Stevanus Il Neofita wrote: I have the license for G729, however I need to use a different codec for the prepaid servi

Re: [Asterisk-Users] cannot set outgoing cid

Hi, it is a PRI. With the old telephone system the extensions were transmitted. Only replaced the telephone systems and whatever I do, only the central dial-in number is transmitted. kind regards Sebastian "Tom Vile" <[EMAIL PROTECTED]> wrote: > Are you allowed to set your callerid with your

Re: [Asterisk-Users] Connecting Asterisk to "traditional" phone central

Hi,I already try that, unfortunately with no success. I wonder what is wrong?Cheers ___ --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] ASTCC: How to reset "in-use" flag automatically ?

I have some troubles with ASTCC. TO often the "in-use" flag remains set. I would like to find a solution, where astcc.agi checks automatically if THIS user is in a call rather than to check the flag. If that is not possible, I would like to have an extension to dial to, and it will after

Re: [Asterisk-Users] Re: caller anounce

Great thanks Steven Job wrote: OK, enough of this.. No reason to bicker about something like this. Here is the URL. http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+cmd+Dial&diff=57 For those of you that do not have a working web browser or cand find it with Google he

Re: [Asterisk-Users] H323 on way voice

[EMAIL PROTECTED] wrote: I am not sure if this debug message is enough information. Try to do what I told. Switch to another H323 channel driver and see what happens. Try first chan_oh323. So instead of solving his configuration problem he should try a new channel driver? Michael Man

RE: [Asterisk-Users] Who is on a call?

The 'sip show channels' and 'show channels' command aren't exactly easy to interpret, especially if one of the numbers has pic codes and rate centers inserted (the rest is truncated on the output), or you have a proxy involved in the call. Wish someone with some C knowledge would fix that. Dou

Re: [Asterisk-Users] H323 on way voice

Neofita wrote: -- channelsOpen = 1 There is only ONE channel open. This should be a huge alarm to you. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options vis

[Asterisk-Users] ZapBarge but ability to talk to the agent

Hi, Is there a way to do a ZapBarge, but where the person doing the barge-in would be able to talk to the agent only (whispering)? Thanks, Andre Courchesne - Consultant http://www.net-forces.com ___ --Bandwidth and Colocation provided by Easynew

[Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma

Hello List! I wanted to share to everyone the following compatible connectivity products that my company installed in our Asterisk based soft switch. I already sent these to the Asterisk.org site many days ago but for some reason they still have to post it. 1. Sangoma A101 single port E1/T1/PRI C

Re: [Asterisk-Users] Re: caller anounce

OK, enough of this.. No reason to bicker about something like this. Here is the URL. http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+cmd+Dial&diff=57 For those of you that do not have a working web browser or cand find it with Google here is the text. Dial macros Introd

Re: [Asterisk-Users] Re: caller anounce

eh? what have I done that is rude, I never even made comments about money? Tom Vile wrote: um ok, maybe not since you seemed a bit rude. On 4/2/06, Miles Scruggs <[EMAIL PROTECTED]> wrote: um ok, well do you mind posting it off list to myself, if you haven't caught it I am interested. Mile

[Asterisk-Users] Who is on a call?

I would like to know which extension number is engaged in a call. show channels shows me: *CLI> show channels Channel Location State Application(Data) SIP/asterisk.elmit.com-0 [EMAIL PROTECTED]:2Up Echo() SIP/8807-066

RE: [Asterisk-Users] Reporting?

Nicolas, Do you have any idea what this will cost and when it might be released? Thanks, Doug > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Nicolás Gudiño > Sent: Friday, March 31, 2006 7:50 PM > To: Asterisk Users Mailing Li

Re: [Asterisk-Users] Re: caller anounce

um ok, maybe not since you seemed a bit rude. On 4/2/06, Miles Scruggs <[EMAIL PROTECTED]> wrote: > um ok, well do you mind posting it off list to myself, if you haven't > caught it I am interested. > > Miles > > Tom Vile wrote: > > There is a reason why I am posting it off list and not because of

Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts

What I do is the following and keep in mind I only use one register statement with my provider: exten => 18665551234,1,SetVar(FROM_DID=18665551234) ; exten => 18665551234,2,Goto(from-pstn,s,1) ; exten => 5185551234,1,SetVar(FROM_DID=5185551234) ; exten => 5185551234,2,Goto(custom-ca

Re: [Asterisk-Users] Re: caller anounce

um ok, well do you mind posting it off list to myself, if you haven't caught it I am interested. Miles Tom Vile wrote: There is a reason why I am posting it off list and not because of money. On 4/2/06, Miles Scruggs <[EMAIL PROTECTED]> wrote: _

Re: [Asterisk-Users] Re: caller anounce

There is a reason why I am posting it off list and not because of money. On 4/2/06, Miles Scruggs <[EMAIL PROTECTED]> wrote: > I would also like to know how to do this, it really defeats the whole > purpose of the list if you reply off list. > > Please post that to the list. > > Miles > > Shaun wr

Re: [Asterisk-Users] Re: caller anounce

Actually you are wrong about $$$ Dovid, I did not charge one penny for it. On 4/2/06, Dovid Bender <[EMAIL PROTECTED]> wrote: > My guess (it's only a hunch) as to why he asked for > off list is for $$$. Usually when you see that the > person wants $$$. I know there is a way of doing it in > asteri

Re: [Asterisk-Users] Information about LOCAL/ Channel

Yes I have used it and it looks like this: exten => s,1,Dial(Local/[EMAIL PROTECTED]) exten => s,2,Goto(s-${DIALSTATUS},1) On 4/2/06, Stuart Elvish - Dallas Delta Corporation Pty Ltd <[EMAIL PROTECTED]> wrote: > Hi, > > I have searched the web and found some basic information about the > LOCAL/ c

[Asterisk-Users] Information about LOCAL/ Channel

Hi, I have searched the web and found some basic information about the LOCAL/ channel. I am wondering if anybody has any good web resources they can point me to on this subject as I think I will need to use it in the near future in a solution I am putting together. Alternatively, if anybody

[Asterisk-Users] Codec Problem

I have the license for G729, however I need to use a different codec for the prepaid service, but when the call is started I have this error Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) ___ --Bandwidth and Colocation prov

Re: [Asterisk-Users] G729 codec problems

Hi, I wonder if VoIP providers consume two licenses when one calls via them? One license for my phone to the provider and one license when call is passed to the recepient. Is that possible? Rudolf On 4/3/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > Steve Kennedy wrote: > > > Each channel n

RE: [Asterisk-Users] Connecting Asterisk to "traditional" phone central

Use a set callerid before your dial statement in extensions.conf. -Original Message- From: Andrew Nowrot [mailto:[EMAIL PROTECTED] Sent: Sun 4/2/2006 4:19 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] Connecting As

[Asterisk-Users] Voicemail() - Reading exit or return results

Here my script:    exten => 230,1,Answer  exten => 230,2,NoOp  exten => 230,3,Voicemail(u${EXTEN})  exten => 230,4,NoOp(Need results from VoiceMail() above - should be non-zero and provided to ARG3)  exten => 230,5,NoOp exten => 230,6,GoToIf($[${ARG3} = 0]?s|8)  exten => 230,7,system(/var/l

Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts

Hi, I'm not an expert, but as far as i know, your incoming calls will arrive with DID in ${EXTEN} so the only thing you need is: exten => 1234,1,GoTo(context1,1234,1) ; example for context extension and priority exten => 2345,1,GoTo(context2,2345,1) exten => 3456,1,GoTo(context3,3456,1) Be sure

[Asterisk-Users] Connecting Asterisk to "traditional" phone central

Hi,I am trying to connect Asterisk to traditional central. It must be done over ISDN. I am utilizing Asterisk 1.0.9 from bristuff RC8p. I was able to communicate them with each other, but I have a problem with callerid. The traditional central does not recognize the callerid from the phones connect

Re: [Asterisk-Users] Re: Cisco 7960 nat problems.

Shaun Reitan wrote: I dont have that option in my phone, this is software 8.2 (p003-08-02 if i remember correctly) That is the firmware that I am also running. It's under SIP Configuration, all the way at the bottom. Not within the extension confgs, keep hitting the down arrow until you ge

[Asterisk-Users] Subversion mirrors of Asterisk, Zaptel and libpri rebuilt

Due to an error in the configuration of the mirroring tool we are using to mirror the repositories from our internal commit server to the public read-only mirror, the revision numbers were not being properly kept in sync (so rev 14381 on the internal server was not the same as on the mirror). This

[Asterisk-Users] Asterisk "answering machine" replacement, "WaitForRing()", application return values

This is possibly a dumb question, but I've googled around and poked through the documentation and I'm a bit confused. My initial experiment with Asterisk involves setting it up in place of my old dedicated "answering machine". That means I've still got a regular old phone on the line which we

[Asterisk-Users] can automon work with MixMonitor

Hi list automon now works as Monitor does. But MixMonitor is a better way for most cases, I guess. Any workaround to make automon do that? Any help will be appreciated. Franz Wu ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-U

[Asterisk-Users] Re: Cisco 7960 nat problems.

Dont have that option, i think i remember seeing it in version 7.xx but i cant remember, i may have to downgrade it i guess ~Shaun "Rich Adamson" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > > > Shaun wrote: >> I have a asterisk server running on site listening on a public ip.

[Asterisk-Users] Re: Cisco 7960 nat problems.

I dont have that option in my phone, this is software 8.2 (p003-08-02 if i remember correctly) ~Shaun "Doug Lytle" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > Shaun wrote: >> I have a asterisk server running on site listening on a public ip. >> Tonight I attempted to connect

Re: [Asterisk-Users] incoming triggers seperate outbound

I will post this info to the list when I get a chance. --- Miles Scruggs <[EMAIL PROTECTED]> wrote: > Hey, > > I would like in the course of dial plan logic, to > trigger a separate > outbound call. If that outbound call is answered, > and if that certain > key response is detected then it wi

RE: [Asterisk-Users] Asterisk 2.0 Where to download

> Think people will fall for it again next year too? Newbies will __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation p

RE: [Asterisk-Users] Asterisk 2.0 Where to download

Think people will fall for it again next year too? > > Hello All > > > > > > I read in www.sineapps.com have Asterisk 2.0 rewritten C# > and run on > > windows, any body could be mail or send to me URL to download. > > > > Thanks > > > > > > Tin Trung Nguyen > > Technical Team

RE: [Asterisk-Users] Blacklist out bound numbers from file

You can do the following. Create a file called blacklist.conf . In it create the following. [BlackList] Exten => 5551212,1,Playback(num-blacklist) Exten => 5551212,2,Hangup Exten => 1900.,1,Playback(num-blacklist) Exten => 1900.,2,Hnagup You get the idea. In your extensions.conf put in the top #i

Re: [Asterisk-Users] Asterisk hosted solution

--- Thorben Jensen <[EMAIL PROTECTED]> wrote: > http://voip-info.org/wiki/view/Easy+PABX > > With Easy PABX you can create your own virtual PABX > online in just minutes. > Easy PABX is based on Asterisk and best of all - > it's completely free. > > Regards > thorben.dk > Whats the catch ? Wh

Re: [Asterisk-Users] Zap channels - help

Hi Tzafrir, thank´s for your help. My configurations: #zaptel.confspan=1,2,0,cas,hdb3cas=1-31:1101loadzone=usdefaultzone=us span=2,1,0,ccs,hdb3bchan=32-46dchan=47bchan=48-62loadzone=usdefaultzone=us #zapata.conf[trunkgroup] [channels]context=defaultswitchtype=euroisdnsignalling=pri_net;rxwink=300u

Re: [Asterisk-Users] Asterisk 2.0 Where to download

Your just one day too late. Some one did the same last year. I have the email some where on my server. Goto be on the game and original. --- Nguyen Trung Tin <[EMAIL PROTECTED]> wrote: > Hello All > > > I read in www.sineapps.com have Asterisk 2.0 > rewritten C# and run on windows,

Re: [Asterisk-Users] Re: caller anounce

I have searched high and low on the wiki, the problem is that it can fit under a number of different names, since this process can be used for tons of different applications. If you find it let me know, I really want this info, as it is the last piece of the puzzle for my asterisk box. Thanks

Re: [Asterisk-Users] Re: caller anounce

My guess (it's only a hunch) as to why he asked for off list is for $$$. Usually when you see that the person wants $$$. I know there is a way of doing it in asterisk. My friend has it. Too lazy to get the code from his box. When I do I will post it. (I also believe that it is on the wiki). Dovid

Re: [Asterisk-Users] G729 codec problems

Steve Kennedy wrote: > Each channel needs TWO licenses, one for each way (I think). Nope. The encoder/decoder licenses are counted separately, and each license you purchase entitles you to one encoder and one decoder. ___ --Bandwidth and Colocation prov

Re: [Asterisk-Users] no audio

--- Luis herrera <[EMAIL PROTECTED]> wrote: > Hi. I have a [EMAIL PROTECTED] setup at my home. My problems is > with > phones outside my network. I call the extensions > without a problem, it rings but when they answer I > can't hear them and they can hear me. > I set up in the SIP.CONF > nat=yes

[Asterisk-Users] no audio between sip channels * 1.2.6

Hello all, I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until recently all was good. on Friday I was running 1.2.5 when I added the fourth phone. I have to admit to initially wiring the rj11(crossed wires) wrong the first time but other than that nothing I can think of. Ad

Re: [Asterisk-Users] FreePBX on Debian

Hi, okay, i managed to install it :) but I have some Problems with mISDN. I have set up two mISDN-extensions, that can phone each other, or the mailboxes. But i can't phone those special numbers like *60 for weather or *98 for message center, what should i do? stoffell wrote: On 4/2/06, Chr

Re: [Asterisk-Users] Asterisk 2.0 Where to download

Kristian Kielhofner wrote: That was last years April Fools joke. I still haven't heard this years... And how much do you want to bet that it'll be this years as well. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, dese

Re: [Asterisk-Users] G729 codec problems

On Sun, Apr 02, 2006 at 09:32:09AM +1000, RumaTech wrote: > Sorry, I was "out of action" for some time. > I am using Voipstnt to plae calls to USA/Canada and I bouhj 1 copy og G.729. > This was mainly to get one of the local Australians VoIP providers working. Each channel needs TWO licenses, one

Re: [Asterisk-Users] Re: caller anounce

I would also like to know how to do this, it really defeats the whole purpose of the list if you reply off list. Please post that to the list. Miles Shaun wrote: Sent you a email ~Shaun "Tom Vile" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] I have a script that will do

Re: [Asterisk-Users] Asterisk 2.0 Where to download

Nguyen Trung Tin wrote: Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download. Thanks Tin Trung Nguyen Technical Team Mobile: 084-91.365.4857 website: www.daivietcontrol.net

Re: [Asterisk-Users] Asterisk 2.0 Where to download

On 02/04/06, Nguyen Trung Tin <[EMAIL PROTECTED]> wrote: > > Hello All > > > I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on > windows, any body could be mail or send to me URL to download. > That version is a year and a day old now, isn't it? Peter -- Peter Bowyer Email: [EM

[Asterisk-Users] Asterisk 2.0 Where to download

Hello All     I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download.   ThanksTin Trung NguyenTechnical TeamMobile: 084-91.365.4857website: www.daivietcontrol.net___ --Bandwi

Re: [Asterisk-Users] H323 on way voice

Is the SIP phone behind NAT?  That's one of the common reasons for one way audio.  You might want to try forwarding some port ranges if you are behind NAT just to eliminate that as a possiblity.  The SIP port ranges should be something like: SIP: 5060-5061RTP: 1-2KyleOn 4/1/06, Il Neofita <

Re: [Asterisk-Users] DID registration status

Try sip show registry from the asterisk console.Kyle On 4/2/06, Giridhar Reddy Bandi <[EMAIL PROTECTED]> wrote: HII have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ?i generally look at the /var/lo

Re: [Asterisk-Users] DID registration status

Giridhar Reddy Bandi wrote: HI I have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ? i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this How about

[Asterisk-Users] DID registration status

HII have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ?i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this thanksGiridhar Bandi _

Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts

Steve Gladden wrote: What version of asterisk? (been lots of changes happening to the sip code over the last year) SVN-branch-1.2-r9156 Have you looked at the sample configs in /usr/src/asterisk/configs? Yes I have and my own configs are pretty much copies of them. They do not detail, do o

Re: [Asterisk-Users] vmail access problem

Evar,I tried to duplicate this on my system but wasn't able to.  Do you have the same password set for both voicemail boxes?  Have you already logged into both of them individually (maybe it's a cookie thing?) KyleOn 4/1/06, Ever Zalazar <[EMAIL PROTECTED]> wrote: Hi everybody..I have th

Re: [Asterisk-Users] H323 on way voice

Hello, As I know from my experience with Chan323 and OH323. I setup Asterisk on Redhat 9.0 i386 and it is working without any problem with Chan323, OH323 libraries required. I never tried OOH323 come (0.4) with Asterisk. If possible I would like to know how to use newest version of OOH323 (0

Re: [Asterisk-Users] Cisco 7960 nat problems.

Shaun wrote: I have a asterisk server running on site listening on a public ip. Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed. My home network is simple, Cox cable connection hooked to a linksys wrt router. The firewall on the linksys router is

Re: [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)

Rich Adamson wrote: Is this worthy of opening a bug assuming the above comment is still valid? Would the individual(s) maintaining res_snmp want to log into either of these internet accessible boxes to identify the root cause? The module loader in trunk is undergoing changes that will eliminate

Re: [Asterisk-Users] Re: How is Teliax ?

end-to-end path. Each step through the tracert process does nothing more then issue an icmp echo request, measuring the response time and displaying it. maybe on windows it does icmp echo but no unix does this (at least not by default). i recommend you study what unix traceroute actually doe

Re: [Asterisk-Users] G729 codec problems

Or, could he be seeing an outstanding issue with counting (eg, start with 0 or 1)? Seems like that might be the case. I've got about ten g729 licenses and never see any warning messages, but then again this is a small system and I don't think I could consume all of them if I tried. Alyed Tzo

Re: [Asterisk-Users] Cisco 7960 nat problems.

Shaun wrote: I have a asterisk server running on site listening on a public ip. Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed. My home network is simple, Cox cable connection hooked to a linksys wrt router. The firewall on the linksys router is d

Re: [Asterisk-Users] Asterisk box with unreliable ping/latency

Hi Bjorn - Everything you mentioned seems to point to the problem being a hardware issue, or more specifically the way that FC and CentOS are using your hardware. Why not use different hardware and/or OS? Maybe FC and CentOS just use faulty driver for your NIC? - Noah __

Re: [Asterisk-Users] Building Asterisk embedded device

Hi Sam - > Thanks for your link. how to build asterisk into > this hardware? As mentioned earlier, have a look at astlinux: http://www.astlinux.org/ There are pre-built versions for soekris/wrap, and general x86 computers. Kristian (the astlinux developer) made this run on a gumstix, too, but

Re: [Asterisk-Users] OT: Polycom IP501 and Speed Dials

Hi Again Avi - > Sadly, that doesn't work -- the Polycoms store their > directories locally as well and re-upload them on reboot. Another idea: Can you create the -directory.xml files as symlinks to the central file? Maybe if the phone sees a directory file already there it will not overwrite i

Re: [Asterisk-Users] polycom overlap dialing?

> Is there any way to get a polycom 601 to do overlap dialing? I can't find anything on the subject, which confirms my initial hunch: I really doubt it. You could probably work something up in asterisk, though. - Noah ___ --Bandwidth and Colocation pr

[Asterisk-Users] polycom overlap dialing?

Is there any way to get a polycom 601 to do overlap dialing? -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] Line Pick up Problem

Could anyone shed some light on this problem: Running Asterisk @ Home 1.7 When call comes in (Zap Clone 100 Card), the extensions ring but when lifting the receiver on an IP phone (BudgeTone 100), I just hear a bit of crackle sound in time with the ringing of the other extensions. The external c

Re: [Asterisk-Users] H323 on way voice

I am not sure if this debug message is enough information. Try to do what I told. Switch to another H323 channel driver and see what happens. Try first chan_oh323. Michael Mansos(or something like that) and other guys have been done a good job. Isamar ___

Re: [Asterisk-Users] FreePBX on Debian

On 4/2/06, Christian Gröger <[EMAIL PROTECTED]> wrote: > need that Zaptel stuff? It always prompted errors so i am now using > mISDN -without errors, is there a module for freePBX for mISDN? to use mISDN with freepbx, you can "Add custom trunk" in the Trunks menu. > Anyway, is there a good manual

Re: [Asterisk-Users] H323 on way voice

This is my debugtried with netmeeting I can still hear but when I talk nothing happenmygw*CLI> h.323 debugH323 debug enabledmygw== New H.323 Connection created.    -- Setting up Call    -- oCall token:  [ip$192.168.1.219:1057/8226]     -- oCalling party name:  [myPersonal]    -- oCalling party numb

[Asterisk-Users] Cisco 7960 nat problems.

I have a asterisk server running on site listening on a public ip.  Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed.  My home network is simple, Cox cable connection hooked to a linksys wrt router.  The firewall on the linksys router is disabled an

Re: [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)

On 2 Apr 2006, at 04:27, Rich Adamson wrote: Kevin P. Fleming wrote: Rich Adamson wrote: Is this worthy of opening a bug assuming the above comment is still valid? Would the individual(s) maintaining res_snmp want to log into either of these internet accessible boxes to identify the root