Re: [asterisk-users] Idiot questions

Hello, If you can buy a TDM400, good! Support Digium Cheers, Madhawa Nilesh Londhe wrote: I would suggest buying a very low price FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping. On 8/24/06, *Adam Collard* <[EMAIL PROTECTED]

[asterisk-users] Re: [Users] Mysql problem

[ Answering on both lists ] On Thu, Aug 24, 2006 at 02:52:54PM -0500, Diego Quintana Cruz wrote: > Hi all, > I have a problem, I can't find nowhere the asterisk-mysql package. I'm > using the sarge version of yours. My repository is: > deb http://rapid.sunsite.dk/rapid/ sarge main > > Hope you ca

[asterisk-users] Re: Nokia E60/61/70 and SIP

On 2006-08-23 18:02:52 -0700, El Flynn <[EMAIL PROTECTED]> said: Hi list, Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70 phones? We're trying to see if this would be an OK phone to get for the company, particularly since we're already running Asterisk. Not as

[asterisk-users] Re: Idiot questions

On 2006-08-24 18:10:20 -0700, "kritikus Araklidas" <[EMAIL PROTECTED]> said: So: The FXO car is for the Pots lines (I.E. bellsouth line) so if you need a analog phone cennected to asterisk you need a FXS card, so if you gonna use a SIP Soft Phone (or a regular SIP Phone) you only need a net

Re: [asterisk-users] hint status not updating on inbound

Hints have got a lot better over the last year, so upgrading might be the only was to get this working really well. Kind regards, PaulH AsteriskIT www.asteriskit.com.au On Thu, 2006-08-24 at 17:36 -0600, Damon Estep wrote: > I have the “hint” priority defined for a few SIP phones. > > > >

[asterisk-users] Re: Intercom mode on Polycom and/or SPA9xx

Thanks to BJ Weschke I have now solved this problem by adding the option s, and taking off the option t from app_page like this: I changed the line that reads (by me line 177): snprintf(meetmeopts, sizeof(meetmeopts), "%ud|%sqxdw", confid, ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"); to: snpri

[asterisk-users] Can the codec/format for name/greeting in voicemail be changed?

Hello people, before I go hunting on Wiki and Google, if maybe someone here knows the answer to this. This is in regards to the voicemail system. Is it possible to change the default/native format in which the greetings and outgoing messgaes for a user's mailbox are stored? It seems like (*) rec

[asterisk-users] Re: Attempt to setup paging and intercom

I do not know if this breaks anything or not the way you have it, but you should not have the underscore before the extension. The underscore means that the following is an expression, where X=any single digit and .=any number of digits. I do not know if the underscore also interprets the * as s

Re: [asterisk-users] Snom phones locking up

We're running several 320s on 6.2.2 and haven't had any issues just to toss my two cents in. On Thu, 24 Aug 2006 23:19:48 +0200 (SAST), garth wrote > Hi All > > I have had a problem with a few Snom 320's on several sites locking > up after a few days. I am running application ver 6.2.2 wi

Re: [asterisk-users] Basic Asterisk Setup

On Jul 5, 2006, at 7:58 AM, K Y Iyer wrote: Hi Hi! Am a bit confused about the basic requirements for a simple, small, test Asterisk setup. There are many options... I want to setup a PBX with 8 PSTN lines and 50 extensions. For argument's sake we'll assume all 50 extensions and 8 PS

Re: [asterisk-users] Phone status

Hello, I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) I have made a deamon, which query Asterisk every second for active calls, this works by issuing a "Status" to the manager-interface, and processing the return data and then put th

Re: [asterisk-users] Trunk with multiple IPs?

Benjamin Lawetz wrote: Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Origin

Re: [asterisk-users] Idiot questions

So: The FXO car is for the Pots lines (I.E. bellsouth line) so if you need a analog phone cennected to asterisk you need a FXS card, so if you gonna use a SIP Soft Phone (or a regular SIP Phone) you only need a network connectivity between Asterisk and SIP Phone. Cris. From: "joea, j4comp

Re: [asterisk-users] Idiot questions

Thanks for all the replies. I take it that with one of these FXO boards, one would need an "IP phone" as there is no FXS ? BTW, cheapest I've seen is $19.95. Still, not bad. joea Nilesh Londhe<[EMAIL PROTECTED]> Boldly Declared: 8/24/2006 8:47 PM: > I would suggest buying a very low price FX

Re: [asterisk-users] RE: [asterisk-dev] Phone status

So how about inventing a car? The auto industry is much more profitable. The point; there is no point in reinventing the wheel, why are you writing this from scratch? On 8/24/06, Mir <[EMAIL PROTECTED]> wrote: What do you mean? I'm not looking for someone elses work, I'm developing an applica

Re: [asterisk-users] Idiot questions

I would suggest buying a very low price FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping. On 8/24/06, Adam Collard <[EMAIL PROTECTED]> wrote: You will need a TDM400 with an FXO module for each line you want. A TDM400 supports up to four lines or analog statio

Re: [asterisk-users] No outbound with A2Billing

Luciano Moreira wrote: List members, When I dial to a PSTN number, the A2Billing script does all the tasks, until it shutdown without make the dailout by sip trunk set. Lasts outputs fro the a2billing.php debug are: a2billing.php|2: RESFINDRATE::> 0 a2billing.php|2: UPDATE cc_card SET inuse=

[asterisk-users] hint status not updating on inbound

I have the “hint” priority defined for a few SIP phones.   When I make a call OUT from one of the phones I see that the “show hints” picks up a status change from 0 to 1 for the extension, but when I call IN to that extension the hint status is still 0.   This is on a server built back

Re: [asterisk-users] SpanDSP Error

Well I got it resolved. Thanks steve for telling me. I looked all around the filesystem and found some files I did not delete from spandsp0.0.3. I saw that solution before and tried it and it didn't seem to work and I wondered why but now i know. Thank you. Chris On Aug 24, 2006, at 10:54 AM

RE: [asterisk-users] Call Parking Ring Back (Snoms)

Look over there : http://bugs.digium.com/view.php?id=6953 David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de J. Oquendo Envoyé : 24 août 2006 13:54 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Call Parking Ring

RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 with IIS

Title: Message We had a similar problem. Eventually we gave up and just used apache. We found that _exactly_ the same content would not work with IIS, but WOULD work with Apache. -Original Message-From: Phil Menico [mailto:[EMAIL PROTECTED]Sent: Thursday, August 24, 2006 3:06

RE: [asterisk-users] Asterisk t38passthrough

Also, make sure you have a T.38 enabled device at the other end…   Edgar     From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Piper Sent: quinta-feira, 24 de Agosto de 2006 21:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [as

Re: [asterisk-users] Idiot questions

You will need a TDM400 with an FXO module for each line you want. A TDM400 supports up to four lines or analog stations. For two lines, you should get a TDM04B. -Original message- From: "joea, j4computers" [EMAIL PROTECTED] Date: Thu, 24 Aug 2006 14:58:21 -0700 To: asterisk-users@lists.di

Re: [asterisk-users] Idiot questions

a TDM400 card will hold up to four modules, which can be FXO or FXS. depending on the modules purchased you can connect up to four phones (using FXS modules) or four incoming phone lines (using FXO modules), or any combination thereof. joea, j4computers wrote: As a complete newcomer to Asteri

Re: [asterisk-users] Snom phones locking up

I have had a problem with a few Snom 320's on several sites locking up after a few days. I am running application ver 6.2.2 with the latest jffs2 ver and tried the latest 5.x ver with similar results. Is this also experienced with other Snom users? not sure if this will help you identif

Re: [asterisk-users] Idiot questions

joea, j4computers wrote: > As a complete newcomer to Asterisk, Digium and PBX, I have several questions. > > But I'll start with this. > > To setup a simple system with only a couple of POTS lines, I gather I will > need a TDM400 board with FXO and/or FXS modules. > > So, a TDM400 card will sup

[asterisk-users] Attempt to setup paging and intercom

This is my first attempt to setup intercom and paging for some Grandview sip phones per instructions from Grandview. I put the lines below in extensions.conf and did the CLI reload command. When I issue **1 or **2 from a phone I get a 404 error. Shouldn't that be ringing the 3 phones on my list

Re: [asterisk-users] Idiot questions

FXO is coming from the PSTN, FXS is what devices connect to (like a analog phone).If you are using VOIP phone then you dont need the FXS modules, just FXO.On 8/24/06, joea, j4computers <[EMAIL PROTECTED]> wrote: As a complete newcomer to Asterisk, Digium and PBX, I have several questions.But I'll

[asterisk-users] Idiot questions

As a complete newcomer to Asterisk, Digium and PBX, I have several questions. But I'll start with this. To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules. So, a TDM400 card will support up to two analog (POTS) lines? joea

[asterisk-users] Snom phones locking up

Hi All I have had a problem with a few Snom 320's on several sites locking up after a few days. I am running application ver 6.2.2 with the latest jffs2 ver and tried the latest 5.x ver with similar results. Is this also experienced with other Snom users? I know some posts say it could be the n

Re: [asterisk-users] Modems dialing over sangoma a104d

Rich Adamson <[EMAIL PROTECTED]> writes: > Sean Cook wrote: >> I have a sangoma 104d that is our main pbx now( legacy system died ). I >> have replaced every phone in the building and things are going very well. >> We have fax working well and calls are routing properly... All is well... >> >> E

[asterisk-users] Polycom microbrowser issue Error HTTP 406 with IIS

Title: Message I have no where else to turn to so if anyone has an answer please send it.   I am running sip version 1.6.on a Polycom 601 on Asterisk and am unable to get the microbroser to work. The phone returns a 406 error for both idle and services. I can see the file being requested a

Re: [asterisk-users] voicemailmain

Howdy guys, Thanks for your help, it works fine without editing the default line of: exten => a,1,VoicemailMain(${ARG1}) The issue was that I had specified VoicemailMain by the default line, which was way above the rest of my extensions (out of context). Hopefully this will help someone in the

Re: [asterisk-users] voicemailmain

Not sure about that Doug. It should read: exten => a,1,VoicemailMan([EMAIL PROTECTED]) If you put it in the brackets, it becomes part of the variable name instead of part of the argument. On Thu, 2006-08-24 at 16:57 -0400, Doug Lytle wrote: > existx wrote: > > Cristian, > > > > The only other l

RE: [asterisk-users] voicemailmain

Hi: First it at all check if you have a different extension for voicemailmain.? Then use VoiceMailMain syntax. And send me the CLI log when you try to connect to VoiceMailMain. regards. Cristian. From: existx <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discu

Re: [asterisk-users] voicemailmain

Ok you have two optionsthe iax extension is created under default context??? The VoceMilMain could be configured with the options of wich context use like this: extensions.conf: exten => 2999,1,Answer exten => 2999,2,Wait,2 exten => 2999,3,Voicemailmain(@test) Wh

Re: [asterisk-users] voicemailmain

existx wrote: Cristian, The only other line in extensions.conf that references VoicemailMain is this: exten => a,1,VoicemailMain(${ARG1}) This should read: exten => a,1,VoicemailMain([EMAIL PROTECTED]) Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase

Re: [asterisk-users] voicemailmain

Cristian, The only other line in extensions.conf that references VoicemailMain is this: exten => a,1,VoicemailMain(${ARG1}) The error from the CLI is: Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not e

Re: [asterisk-users] Asterisk t38passthrough

Perhaps a stupid suggestion... but did you make sure that the ATA had the T38 selected in the GUI?   bp  On 8/24/06, Ricardo Carvalho <[EMAIL PROTECTED]> wrote: Hi,I've installed Asterisk t38passthrough branch and I'm using oneGrandstream ATA to connect Asterisk to a Fax machine. Every time I send

Re: [asterisk-users] voicemailmain

On Friday 25 August 2006 08:39, existx wrote: > The error from the CLI is: > > Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected > connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not > exist It looks like you have created 2699 in a different context than you

[asterisk-users] voicemailmain

Howdy, I have a Debian box using Debian's Asterisk package. People can leave voicemail for the extensions that are setup in the configuration, and asterisk e-mail's the user a .wav file (voicemail.conf). This works perfect. However, I want to have VoicemailMain sit on an extension so people can

RE: [asterisk-users] Static in Monitor recordings

I turned off hyper threading and it did not help, I also tried telling it to record as gsm files but it still makes an awful noise.  Anyone have any other ideas?   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don Sent: Friday, August 18, 2006 12:04 PM To: Asteris

Re: [asterisk-users] voicemailmain

Aaron Daniel wrote: Not sure about that Doug. It should read: exten => a,1,VoicemailMan([EMAIL PROTECTED]) You are correct. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

[asterisk-users] Re: Nokia E60/61/70 and SIP

> "H" == Haspers <[EMAIL PROTECTED]> writes: H> I've got them all. It registers correctly with Asterisk, and get H> incoming calls, but it complaints about outgoing calls (Connection H> Error). SIP Debug is giving me: SIP/2.0 407 Proxy Authentication H> Required H> But those settings are th

RE: [asterisk-users] RE: [asterisk-dev] Phone status

I guess what Andrew was saying is what are you trying to do specifically that Flash Operator Panel doesn’t already give you     Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mir Sent: Thursday, 24 August 2006 2:48 PM To: Asterisk Users Mailin

Re: [asterisk-users] Re: Zaptel install - Fedora Core 5

On Tue, Aug 22, 2006 at 11:56:26 +0200, Tomislav Parčina <[EMAIL PROTECTED]> wrote: > > The thing is that I can't rely on yum update for asterisk installation. I > would like something that will work like this: I install FC5 from CD/DVD, > install RPM's that I need from my ftp server or from C

[asterisk-users] Re: Working Sipura 3000 or Linksys 3102 configuration?

At 15:12 23/08/2006 -0700, Ron Wellsted wrote: Here are mine (with UK regional settings/A-law). Thanks a bunch :-) After more search, it turns out that if you don't need the router feature of the 3102 (I already have a router), the unit must be connected to the LAN through its... WAN plug. It

Re: [asterisk-users] Phone status

Michael, you might take a look at this, as it does most of that + more: http://micpc.com/eventmonitor/ earl On Thursday 24 August 2006 14:17, Mir wrote: > Hi > > I'm working on a project, where I need the status of every telephone on the > system. (Idle,ringing,busy) > If a phone is bu

Re: [asterisk-users] RE: [asterisk-dev] Phone status

What do you mean?   I'm not looking for someone elses work, I'm developing an application from scratch.   Michael  2006/8/24, Andrew Kirch <[EMAIL PROTECTED]>: Umm… Flash operator panel?   Andrew   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of MirSent: Thursday, August 2

Re: [asterisk-users] Modems dialing over sangoma a104d

Sean Cook wrote: I have a sangoma 104d that is our main pbx now( legacy system died ). I have replaced every phone in the building and things are going very well. We have fax working well and calls are routing properly... All is well... Except for our support modems... we have support people t

[asterisk-users] hotel teledex integration anyone?

All,   I am looking at taking on a project for a hotel that is using Teledex systems. I see that they have a SIP based phone and the information says that there is some CMS server part that appears to be the brains behind the device. My questions are; has anyone out there used this type o

[asterisk-users] RE: [asterisk-dev] Phone status

Umm… Flash operator panel?   Andrew   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mir Sent: Thursday, August 24, 2006 2:18 PM To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com Subject: [asterisk-dev] Phone status   Hi   I'm wor

[asterisk-users] Phone status

Hi   I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end.   I have made a deamon, which query Asterisk every second for active calls, this works by issuing a "Status" to the man

[asterisk-users] Quiet on the list today?

Just gotta check, I've never seen a complete day with no posts ___ --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] Call Parking Ring Back (Snoms)

Quick question maybe someone can point me in the right direction... Caller --> Receptionist --> ParksCall Receptionist makes announcement for individual to pick up parked call. No one picks up so it rings back to receptionist within a minute and a half. Is there any way to change the ringer for

[asterisk-users] Modems dialing over sangoma a104d

I have a sangoma 104d that is our main pbx now( legacy system died ). I have replaced every phone in the building and things are going very well. We have fax working well and calls are routing properly... All is well... Except for our support modems... we have support people that dial out with m

[asterisk-users] Asterisk t38passthrough

Hi, I've installed Asterisk t38passthrough branch and I'm using one Grandstream ATA to connect Asterisk to a Fax machine. Every time I send a fax, it gets sent using codec G711, and never T.38. I added the following parameters in the [general] section as well as in device configurations: t3

Re: [asterisk-users] Annoying Bristuff

HiCan anyone confirm a working asterisk 1.2 from bristuff with 1 port PCI, hfc-s based ISDN card (zaphfc driver). If so, could you send your configuration. I mean OS (linux distribution) type, kernel version. Thanks in advanceCheersAndrew ___ --Bandwidth

RE: [asterisk-users] Realtime and hints

I don't see how that helps. If you have a portion of the hint still in extensions.conf, then what use is the database? > -Original Message- > From: Aaron Daniel [mailto:[EMAIL PROTECTED] > Sent: Thursday, August 24, 2006 10:24 AM > To: Asterisk Users Mailing List - Non-Commercial Discussi

Re: [asterisk-users] E61

Using STUN isn't a solution to NAT either, as it won't work with symmetrical NAT, which is very common (or for at least to partially use symmetrical). I'll be interested to see how the Paragon Wifi phone fares out when it starts making an appearance in the US. -Brodie On Thursday 24 August 200

[asterisk-users] No outbound with A2Billing

List members, When I dial to a PSTN number, the A2Billing script does all the tasks, until it shutdown without make the dailout by sip trunk set. Lasts outputs fro the a2billing.php debug are: a2billing.php|2: RESFINDRATE::> 0 a2billing.php|2: UPDATE cc_card SET inuse=inuse-1 WHERE username='5

RE: [asterisk-users] Realtime and hints

That's what he was gettin at. Take the second line out, and put the first priority in the database. On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote: > But... you need _both_ in your dialplan. > > My extensions.conf has: > > exten => 2944054,hint, SIP/2944054 > exten => 29

RE: [asterisk-users] Realtime and hints

But... you need _both_ in your dialplan. My extensions.conf has: exten => 2944054,hint, SIP/2944054 exten => 2944054,1, Dial(SIP/2944054) ie two lines for the hint. > -Original Message- > From: Dovid Bender [mailto:[EMAIL PROTECTED] > Sent: Thursday, August

[asterisk-users] Re: channel variable

In article <[EMAIL PROTECTED]>, unplug <[EMAIL PROTECTED]> wrote: > Tony Mountifield wrote: > > > > Try using Set(__testmode=1), but still refer to it as ${testmode}. > > > > The __ tells Asterisk to propagate the variable to created channels. > > Do you mean it is a global variable instead of cha

Re: [asterisk-users] SendText Queue Notification

I know this isn't answering your question, but what I did for queue notification was use softkeys on the phones that call a PHP script on the * box that'll output XML for the phone to parse and display the queue stats on demand. Of course your phone would need to have an XML parser or some other

Re: [asterisk-users] No CLID from PSTN using X100P FXO Card

However, reinstalled the box from ground up and installed 1.2.10 and now CLID isn't working at all. The PSTN line is still transmitting it, as I've plugged in my Uniden cordless with CLID and it shows up fine on there, but getting absolutely nothing inside the ${CALLERIDNUM} and ${CALLERIDNAME}

[asterisk-users] Wellgate 3804a

I want that each call from PSTN goes to Asterisk to the context for this line. Within this context can be a menu or a dial command, ... As more I read, as more I get confused, ... and each try is not working! My sip.conf: [WG88621001] type=friend defaultip=192.168.250.244 insecure=very cont

[asterisk-users] originate from group + dialplan

Hello, is possible execute a dialplan before make a call from Zap/g0/dnis? How can i do that? Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digiu

[asterisk-users] SendText Queue Notification

I was wondering if anyone was able to execute custom commands on a channel once a caller connects to an agent after being in a queue. The reason I ask, is because I would like to use SendText to send a message to the agent receiving the call to let the agent know how many calls are waiting in the

Re: [asterisk-users] Realtime Extensions -- Comments?

Maybe he is tryin to make it work. This is much better than the old Doug. Also if he needs features that currently dont exist maybe some one will create it and then we will all benefit from it :) . - Original Message - From: "Brian Capouch" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing

Re: [asterisk-users] Realtime and hints

Doug I have Exten => 10,hint,SIP/11010 and in mysql I have exten => 10,1,Dial(SIP/11010) - Original Message - From: "Douglas Garstang" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, August 21, 2006 3:37 PM Subject: [asterisk-users] Real

[asterisk-users] need help with error code

Can anyone give me some insight as to what this message means from /var/log/asterisk/full Aug 24 10:32:02 DEBUG[3016] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 1 Aug 24 10:32:02 DEBUG[3016] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on ch

RE: [asterisk-users] SSH connection hangs on logout?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Tzafrir Cohen > Sent: Thursday, August 24, 2006 2:32 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] SSH connection hangs on

Re: [asterisk-users] SpanDSP Error

Christian Jensen wrote: I have not found any solution to the problem I am talking about in the archives Steve. I have reinstalled and downgraded from pre26 to pre21 of SpanDSP 0.0.2 and still to no avail. I have followed all the instructions and ways of fixing this problem and have found n

Re: [asterisk-users] Cisco Router QOS and IAX2

The majority of the sample qos policies seem to be based on either five or seven qos queues, and most folks don't need all of that. What I've shown as a sample only has three queues; one for voip, one for my outbound web traffic, and the default queue that everything else falls into. You can

Re: [asterisk-users] E61

here is the link (german): http://www.my-s60.com/de/news/news/newswann_kann_die_e_serie_nat_traversal/back/105/cHash/bac3645f5e/index.html STUN will come in fall 2006 TURN and ICE in 2007 Thomas Artner wrote: > Hi! > > I can't find the link anymore where it was a statement from the nokia

Re: [asterisk-users] Active Directory Listing Feauture

Are you asking about the LDAP stuff or voicemail directory? On 8/24/06, Mohamed A. Gombolaty <[EMAIL PROTECTED]> wrote: Dear All, I am currently very stumped on the subject of Active Directory listing, as I am unable to find any documents regarding this feature thus I am unable to configure

Re: [asterisk-users] E61

Hi! I can't find the link anymore where it was a statement from the nokia support that they are working on a STUN implementation. A firmwareupdate (with STUN support) will be available in fall 2006. tom Andreas Sikkema wrote: >> Anyone here use the Nokia E61 ? I am looking to invest in a >> w

Re: [asterisk-users] Re: Nokia E60/61/70 and SIP

Hi,Yes, the proxy and registrar settings are identical in my set-up. Getting registered is the hard part but you've done that. I'd be looking at the Asterisk end of things rather than the E61 to see why that authentication issue is arising. WOn 8/24/06, Haspers <[EMAIL PROTECTED]> wrote: I've

Re: [asterisk-users] Cisco Router QOS and IAX2

Thank you so much. After fighting with a large/extensive QOS policy from Cisco's SDM tool, I used your sample and tweaked it for my needs and everything started working fine.Bruce On 8/23/06, Rich Adamson <[EMAIL PROTECTED]> wrote: Bruce Reeves wrote:> I'm needing some pointers from anyone who has

Re: [asterisk-users] SpanDSP Error

On Thu, 2006-08-24 at 09:29 -0400, Christian Jensen wrote: > I have not found any solution to the problem I am talking about in > the archives Steve. I have reinstalled and downgraded from pre26 to > pre21 of SpanDSP 0.0.2 and still to no avail. I have followed all > the instructions and way

Re: [asterisk-users] E61

Hi Andreas,I'm on 1.0610.04.04 19-04-06 RM-89WOn 8/24/06, Andreas Sikkema <[EMAIL PROTECTED] > wrote:Simon,> That is incorrect. It works just fine through NAT providing: >> - The server is proxying RTP as it has no support for STUN etc.> - The NAT is the basic domestic router style, not a full> blo

RE: [asterisk-users] E61

Simon, > That is incorrect. It works just fine through NAT providing: > > - The server is proxying RTP as it has no support for STUN etc. > - The NAT is the basic domestic router style, not a full > blown firewall requiring port mappings Strange, then you must have some other firmware, because

[asterisk-users] Monitoring/Listening In (Scott Pinhorne)

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[asterisk-users] About IVR and Oracle (Tim Panton)

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RE: [asterisk-users] Re: Nokia E60/61/70 and SIP

I've got them all. It registers correctly with Asterisk, and get incoming calls, but it complaints about outgoing calls (Connection Error). SIP Debug is giving me: SIP/2.0 407 Proxy Authentication Required   But those settings are the same (Proxy Server/Registrar Server). So what could be t

SV: [asterisk-users] E61

I also have this phone, and have stumbled in to the same problem. I just think that it isn't in nokia's interest to change this, as it forces consumers to have some sort of local hardware, that (possibly) only the telecom provider can give them. This forces the users away from using cheaper servi

Re: [asterisk-users] E61

Hi Andreas,That is incorrect. It works just fine through NAT providing:- The server is proxying RTP as it has no support for STUN etc.- The NAT is the basic domestic router style, not a full blown firewall requiring port mappings SimonOn 8/24/06, Andreas Sikkema <[EMAIL PROTECTED]> wrote: > Anyone

Re: [asterisk-users] SpanDSP Error

I have not found any solution to the problem I am talking about in the archives Steve. I have reinstalled and downgraded from pre26 to pre21 of SpanDSP 0.0.2 and still to no avail. I have followed all the instructions and ways of fixing this problem and have found none to be a solution. -

RE: [asterisk-users] E61

> Anyone here use the Nokia E61 ? I am looking to invest in a > wifi phone and I want to get the best. Is it good as far as > reception ? That is of most importance to me. Thanks. I've tried it in the last couple of days. The biggest issue for me ist that it HAS to be on the same side of a NAT

[asterisk-users] E61

Anyone here use the Nokia E61 ? I am looking to invest in a wifi phone and I want to get the best. Is it good as far as reception ? That is of most importance to me. Thanks.   Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

Re: [asterisk-users] Active Directory Listing Feauture

Mohamed A. Gombolaty wrote: Dear All, I am currently very stumped on the subject of Active Directory listing, as I am unable to find any documents regarding this feature thus I am http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Directory Doug -- Ben Franklin quote: "Those who

[asterisk-users] Active Directory Listing Feauture

Dear All, I am currently very stumped on the subject of Active Directory listing, as I am unable to find any documents regarding this feature thus I am unable to configure it or know how to use it. Does anyone have any useful info or documents regarding this feature in terms of how to or guides I

RE: [asterisk-users] quintum Calling Card

Abdul, it doesn’t sound like you need to do anything to the Quintum. I would recommend making your dial plan execute the AGI script of your choice no matter what number is dialed from the context where the quantum users land.   -Jonathan   From: [EMAIL PROTECTED] [mailto:[EMA

[asterisk-users] quintum Calling Card

Hi all,Could anyone provide me some usefull link or some idea, how to configure quintum as calling card purpose with Asterisk.Already i created AGI script which working with SIPURA well. But i do not have the idea about quintum how to configure so quintum will dial our asterisk calling card number.

Re: [asterisk-users] Monitoring/Listening In

I wish to setup asterisk for training purposes so that I am able to listen in to an extension while a call is going on? http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge and http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy ___ --Bandwidth and

[asterisk-users] Multiple lines in body of UserEvent

Hi everybody, I'm trying to send a user event from the dialplan like this: exten => s,n,UserEvent(EventName|var1:value1^var2:value2) The event is sent just fine, but the body is not split in two lines as it should be according to http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+UserE

Re: [asterisk-users] IAX2 extn not registering on 4569

[EMAIL PROTECTED] wrote: > Hi all, > > Just having a strange situation with no clues how to solve. > > I have an Asterisk/TRIXBOX located in US and an IAX extn running on > PA168V ATA in another country. All my configs seems to be on 4569 but > i see my extn connected at a different port like 1

Re: [asterisk-users] Re: Nokia E60/61/70 and SIP

Hi Haspers,Makes sure you have created an 'Internet tel' profile. It doesn't appear to do anything but was vital in getting it working for me. The other settings in the how to look sensible.Simon On 8/24/06, Haspers <[EMAIL PROTECTED]> wrote: Strange,What settings do you use? I followed this linkht

Re: [asterisk-users] Re: Nokia E60/61/70 and SIP

Simon Woodhead wrote: > Hi Benny, > > The E61 handles this just fine. With SIP as the default channel to > dial and no WiFi coverage, you get a message asking if you'd like to > dial by cellular. Works nicely other than a few stability issues. > The E60 appears to handle WPA2 fine, with roaming acr

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