[Asterisk-Users] Snom 360 with Firmware 6.1?

Hi, Has anybody experience with Snom360 and Firmware 6.X with Asterisk 1.2.X? I am currently using Firmware 5.5 without serious problems but wanted to make sure 6.X will work as well (including subscription etc.) Kind regards, JP smime.p7s Description: S/MIME cryptographic signature

RE: [Asterisk-Users] Out of Office Auto Reply:

On Thursday, June 22, 2006 8:13 PM Anthony Rodgers wrote: We use MS Exchange too and, as far as I am aware, it is cognizant of mailing list headers and doesn't send OOO notices to mailing list postings. The only mailing list from which I receive my own OOO notices is one that doesn't have the

Re: [Asterisk-Users] show queue ... Invalid

- Denis Shaposhnikov [EMAIL PROTECTED] wrote: I've found the problem. That's because I've loaded app_queue.so before chan_sip.so in modules.conf. That makes perfect sense. Thanks for following up! -- Kevin P. Fleming Senior Software Engineer Digium, Inc.

Re: [Asterisk-Users] sip to h323 ... direct RTP?

- Jeremy McNamara [EMAIL PROTECTED] wrote: The problem is 're-inviting' in H.323-jive is very much a non-trivial task. Ahh, OK, then this is a protocol limitation more than an implementation issue. Never mind :-) -- Kevin P. Fleming Senior Software Engineer Digium, Inc.

Re: [Asterisk-Users] Re: fail to make call

- unplug [EMAIL PROTECTED] wrote: Is it the way for asterisk realtime system? register: UA1 --register-- asterisk1 - store user information in DB UA2 --register-- asterisk2 store user information in DB UA1 --invite-UA2--- asterisk1 asterisk1 query UA2 information in DB

Re: [Asterisk-Users] Re: fail to make call

- unplug [EMAIL PROTECTED] wrote: BTW, do you mean this function will be included in next release? When will be the next release available? No, it will not be in the next release (which is Asterisk 1.4). It may be in Asterisk 1.6, scheduled for January or so of 2007, but as I said in my

RE: [Asterisk-Users] Snom 360 with Firmware 6.1?

Just installed! Use 6.1.1 (beta) because 6.1 has a few of registration problems. Bye -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Koopmann, Jan-Peter Sent: Friday, June 23, 2006 9:24 AM To: Asterisk Users Mailing List - Non-Commercial

Re: [Asterisk-Users] 1.2.9.1 crashed today

- Steve Totaro [EMAIL PROTECTED] wrote: My system never stops processing calls but it slows. On the CLI, it may take five minutes for a command to execute but it does finally execute (such as sip show peers or show channels) And calls to the agents sit in queue for a while before

Re: [Asterisk-Users] SPA-2002 call HANGUP. May be a SIP bug.

- Dmytro Mishchenko [EMAIL PROTECTED] wrote: In this case during all conversation SIP packets contains Call-ID: [EMAIL PROTECTED] but the final BYE packet from adapter contains Call-ID: [EMAIL PROTECTED] Is such scenario correct from SIP protocol point of view? No, it is not

Re: [Asterisk-Users] Asterisk 1.2.7/9.1 mp3 volume is good, wav file of same volume are too loud!

- Matt [EMAIL PROTECTED] wrote: Is there something volume wise different about the internal mp3 and wav players? Probably. People usually use a volume factor of somewhere between .25 and .40 in sox when preparing files for MOH usage. -- Kevin P. Fleming Senior Software Engineer Digium,

Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

On Thu, 2006-06-22 at 20:53 -0400, Steve Totaro wrote: From their README Disclaimer -- The information provided by CDRTool documentation is not always enough to successfully complete the installation and deployment of CDRTool. Most of the configuration tasks are related to

[Asterisk-Users] billing

I am using trixbox ,please any ont knows how to confiure billing on it, I want to make a billing ,I created an account at http://x.X.X.X/a2billing but it does work * No employee or agent is authorized to conclude any binding

Re: [Asterisk-Users] How to set overlap dial timeout in bristuff zaptel?

I have the same problem here. There is an function called TIMEOUT and in 1.2.9.1 it has to be set for specific channel this way: Set(TIMEOUT(response)=seconds) This works on any other but Zap channels. It seem that the default timeout is still back there discarding my custom one. My playground

Re: [Asterisk-Users] PRI Issue - Calls being rejected with unacceptable channel

On 06/23/06 01:22 Andy Brezinsky said the following: Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 15996/0x3E7C) (Originator) Message type: SETUP (5) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 48764/0xBE7C) (Terminator) Message

Re: [Asterisk-Users] GXP 2000 - BLF and Hold/Hangup Answering

I had the same problem some time ago. Make sure call waiting is NOT disabled. This will make the phone receive more calls on the other lines. - Daniel On Jun 23, 2006, at 1:29 AM, Corporate IT Solutions - Michael Dunne wrote: I have a network of GXP 2000 phones and would like to know if

[Asterisk-Users] GXP-2000 and Shared Line Appearances

I have a client with 20 GXP-2000s. Everything seems to be working fine. However, after a couple of weeks of use, the client is having a hard time adjusting to the new IP based phone systems and only misses one feature from their old Lucent system. That is, they had 8 analog lines before

RE : Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

Steve, I do hope ag-projects to keep their products with commercial licences ! Harry PS: most of billing (or not) open source projects provide a good documentation . --- Steve Totaro [EMAIL PROTECTED] a écrit : Alex Robar wrote: Harry, Fisrt ag-projects talk about is product

[Asterisk-Users] Passing DID to external number?

Hi, We run a small switchboard using Asterisk and Free PBX. We have two main extensions and two ring groups. The first ring group rings the two internal extensions. If the internal extensions do not pick up the call after 15 seconds then the second ring group kicks in which should ring

Re: [Asterisk-Users] Snom 360 Passsword Issue

I have had the same problem too, I solved resetting the phone to factory defaults Edward de Zeeuw wrote: I'll take a look first thing tomorrow and let you know what I find. Thanks! Edward Colin Anderson wrote: In the Snom web management page under Advanced make sure Challenge response

Re: [Asterisk-Users] 1.2.9.1 crashed today

Kevin P. Fleming wrote: - Steve Totaro [EMAIL PROTECTED] wrote: My system never stops processing calls but it slows. On the CLI, it may take five minutes for a command to execute but it does finally execute (such as sip show peers or show channels) And calls to the agents sit in

Re: RE : Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

You should have been working with asterisk about four years ago, lol! Talk about no docs. Most opensource projects provide read between the lines and Google the error to get a clue, your brains, trial and error docs. At least in my experience. Want some good docs? Maybe read the docs for

Re: [Asterisk-Users] Hitting * in a queue call hangs up?

On 06/20/06 18:20 Matt said the following: It seems 1.2.9.1 does not correct this behavior... can I correct it somehow? matt, i believe i've already sent this to the list. the bug at http://bugs.digium.com/view.php?id=6897 has the fix for 1.2.x as agent-endcall.patch. apply that, and

RE: [Asterisk-Users] SE Michigan asterisk users group

I'd certainly be up for it, even if it ended up being a small group that met over beer at a local pub. :) Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BerkHolz, Steven Sent: Thursday, June 22, 2006 4:27 PM To:

RE: [Asterisk-Users] TE405P Dropping Calls. !! Got I-frame while linkstate 0

I would say it's almost certainly a cabling issue of some sort. We get the S-frame while link down message all the time on our Asterisk clusters due to the way the T1 failover switch works. It's harmless in our case since really the PRI is connected to the other box in the cluster (unless there

[Asterisk-Users] best hardphone for Asterisk?

Dear Friends,We have implemented "Asterisk" in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost?Thank you.Regards,Chandra.

[Asterisk-Users] Call accounting where calls cross charge zones (code fragment request)

Greetings list, Before I go and write something from scratch, are there any kind souls here who already have a nice code fragment that works our charging for calls split across charging zones? There are essentially 4 possibilities for a call: 1) call is completely within one zone, so it's nice

[Asterisk-Users] fax

How can I support fax at trixbox M. Khaled Chehab Monitoring Operationg Engineer Xplorium Tel: +961 1 868686 Fax: +961 1 808810 e-mail: [EMAIL PROTECTED] * No employee or agent is authorized to conclude any

[Asterisk-Users] asterisk sip listening port

How I can let asterisk listen only at port 5062 since I have ser on the same machine listening to port 5060 , Please from where I can configure it * No employee or agent is authorized to conclude any binding agreement on behalf of

Re: [Asterisk-Users] Routing inboud from ISDN to second * server.

I dont know if the extensions.conf is any use here as I am configuring the access via freepbx ? pls let me know if so and I will post the file. Other thing I am thinking of - have seen other posts about diferrence IP subnets must be configured in asterisk to allow thise calls to be routed. Is

Re: [Asterisk-Users] Snom 360 with Firmware 6.1?

Koopmann, Jan-Peter wrote: Hi, Has anybody experience with Snom360 and Firmware 6.X with Asterisk 1.2.X? I am currently using Firmware 5.5 without serious problems but wanted to make sure 6.X will work as well (including subscription etc.) Use the very latest - 6.2.1. It seems quite good.

Re: [Asterisk-Users] asterisk sip listening port

in the [general] section of sip.conf bindport=5062 well documented here: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf -yair On 6/23/06, Khaled Chehab [EMAIL PROTECTED] wrote: How I can let asterisk listen only at port 5062 since I have ser on the same machine

RE: [Asterisk-Users] best hardphone for Asterisk?

snom 300 :-) CS From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy BoySent: Friday, June 23, 2006 7:16 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] best hardphone for Asterisk? Dear Friends,We have implemented "Asterisk" in our

[Asterisk-Users] Antek EGW-804 e *

Hi everybody, I found in the company where I work an Antek EGW-804. I googled to see if it can be configured to work with * and I understood that it is possible, but I don't know how. Can someone help me? Thanks Stefano ___ --Bandwidth and

Re: [Asterisk-Users] fax

That's a vague question Khaled :) First you must have hardware/software to support fax. Faxes on my TDM400P work great with asterisk; they didn't work so great over voip or with my X100P though. Next you need software.. spandsp works ok to me with my fax to email setup. Read all about it here:

[Asterisk-Users] Realtime voicemail not working

Hi, I've been using MySql with the CDR for awhile with no issues at all, but I figured I'd try putting voicemail users in a DB. Using the same DB (very low load), and a user that has been proven to work well using a client GUI, I inserted one user and tried to use realtime. This is what I get

[Asterisk-Users] Setting caller-id when parking call

I have an issue where someone will park a call, and then it will ring back to them, but because the caller-id looks like a regular inbound call, they don't know how to answer the call (these are the receptionists). I've tried to make an extention that I can transfer to that will set the

Re: [Asterisk-Users] best hardphone for Asterisk?

Crazy Boy schrieb: We have implemented Asterisk in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost? I would say a

[Asterisk-Users] Trunk failover

Hi, I'm doing some experiments with SIP vs. IAX2 trunks as an alternative to PSTN and I noticed that, if voip link is down, failover to PSTN is almost immediate with the SIP trunk and VEEERY slow with the IAX trunk. Is there a specific reason? Some timeout to set in iax.conf? Thanks -- Domenico

[Asterisk-Users] calling between contexts

hi all, somebody know a way how to call between contexts which are in a realtime database? i tried to include them wise versa in extension.conf but this is not working. Is there another way? regards rene ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] best hardphone for Asterisk?

Christian Stredicke schrieb: snom 300 :-) Could be a bit hard to get 150 of them at one time imho. ;-) Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] freepbx centos 4 install script?

Michael Collins wrote: Has anyone created a script that will download and install all of the freepbx prerequisites in the INSTALL file automatically on a Centos 4 box? In a manner of speaking the trixbox guys have. Have you ever seen that (or Asterisk @ Home)? There is

[Asterisk-Users] Dial(ZAP with t option for call transfer via *2)

Hi all, using asterisk 1.2.9.x I would like to forward an incoming call to an outbound ZAP target (EuroISDN PRI via Digium TE410P), i.e. an mobile phone. exten = 105,1,Dial(Zap/r1/0171234567,120,rt) I use the Dial() option t as the goal is to enable the called destination to be able to perform

[Asterisk-Users] UK English Sounds

New versions of the Male UK English sound files are now available. We believe these are complete for v1.2.x of Asterisk and v1.2.1 of Asterisk-sounds. The LouisLouis song is missing (which is US anyway) and 7 seconds of silence, but everything else should be there. The vm voice prompts are now

RE: [Asterisk-Users] best hardphone for Asterisk?

If you do not have a budget, grandstream is not bad You do not get what you do not pay But you do not allways get what you paid for t. Jacobson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Victor Sent: vendredi 23 juin 2006 14:25 To:

Re: [Asterisk-Users] Out of Office Auto Reply:

Dean Collins wrote: You got to be freaking kidding, a month of this? Cant we get an easy process for the list owner to take care of these? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent:

Re: [Asterisk-Users] best hardphone for Asterisk?

I have not had a problem getting SNOMs. Keep in mind that SNOM phones have a penguin inside. m,, Penguins.. On 6/23/06, Christian Victor [EMAIL PROTECTED] wrote: Christian Stredicke schrieb: snom 300 :-) Could be a bit hard to get 150 of them at one time imho. ;-) Chris

Re: [Asterisk-Users] Dell PowerEdge 1650

Sean, I do not have one up and running, however I do remember seeing various posts earlier this year about incompatibilities with Dell motherboards and Digium hardware. IIRC the solution was to either change to a different server or to change to a Sangoma board. You might want to search the

[Asterisk-Users] SIP - PSTN calls not connecting properly

Hi, I've got a problem with my asterisk set up which has been going on for a while (months). I'm currently running 1.2.7.1 on a gentoo box with the topology below: +-+ PSTN -+ * +- Service Provider (wctdm400p) +-+-+-+ IAX

Re: [Asterisk-Users] Realtime voicemail not working

2006/6/23, Mike [EMAIL PROTECTED]: Hi,I've been using MySql with the CDR for awhile with no issues at all, but Ifigured I'd try putting voicemail users in a DB.Using the same DB (verylow load), and a user that has been proven to work well using a client GUI, I inserted one user and tried to use

Re: [Asterisk-Users] calling between contexts

2006/6/23, René Enskat [Teamware GmbH] [EMAIL PROTECTED]: hi all, somebody know a way how to call between contexts which are in a realtime database? i tried to include them wise versa in extension.conf but this is not working. Is there another way?I solved this using the Goto and

Re: [Asterisk-Users] best hardphone for Asterisk?

I find the Polycom Soundpoint 301 and 501 models to be great phones. Christian Victor wrote: Crazy Boy schrieb: We have implemented Asterisk in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff.

Re: [Asterisk-Users] problem - DSL line and Digium card

I was wondering if anyone has seen a similar problem... I have a DSL line that doubles as a voice line to my Asterisk box. When the Digium card answers that line, the DSL modem is disconnected until the line releases. The phone line is split at the DSL modem then run to an FXO

Re: [Asterisk-Users] Dell PowerEdge 1650

I can't comment on the 1650, but we are running three 2850s in production, with heavy load, and they are working fine. In one I have 3 PRI cards, in another, only 1. On 6/22/06, Sean Cook [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Anyone have a 1650 running

Re: [Asterisk-Users] asterisk sip listening port

On 14:42, Fri 23 Jun 06, Khaled Chehab wrote: How I can let asterisk listen only at port 5062 since I have ser on the same machine listening to port 5060 , Please from where I can configure it sip.conf -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key:

[Asterisk-Users] Kernel 2.4 / 2.6 and timer

I've read in different places that if I want to do trunking and meetme on Asterisk I need to have a reliable timer. People have recommended that I install a Digium board, even if I don't have any circuits connected to it, just to get a reliable timer. However, I've also read that if I'm

Re: [Asterisk-Users] SIP - PSTN calls not connecting properly

I had this same problem. For me, the Cisco phone wasn't detecting that the call was connected. Turn on VAD, and maybe bump up the rx gain on the PSTN. Hope that helps, Brian On Jun 23, 2006, at 8:04 AM, Ronan Mullally wrote: Hi, I've got a problem with my asterisk set up which has been

RE: [Asterisk-Users] Realtime voicemail not working

That is what I have in my config file. Pretty much the same thing as the CDR file [general]dbhost=localhostdbname=dbdbuser=userdbpass=pw;dbport=3306;dbsock = /tmp/mysql.sock Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin StockerSent: June 23, 2006 9:07

[Asterisk-Users] Meetme max users

Does anyone knows what is the max of users that meefme can handle. I am using Iax2 clients to connect to the conference. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options

[Asterisk-Users] Asterisk 1.4 on schedule?

Is Asterisk 1.4 on schedule for release in July? /Obelix ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Realtime voicemail not working

Weird. I just tried with 127.0.0.1 instead of localhost and it worked. Can't explain why How can I force it to go through a db socket instead? Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin StockerSent: June 23, 2006 9:07 AMTo: Asterisk Users Mailing List

RE: [Asterisk-Users] SE Michigan asterisk users group

Steven, I am in Livonia. Please let me know if there is interest. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of BerkHolz, Steven Sent: Thursday, June 22, 2006 4:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SE Michigan asterisk

Re: [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings

Tom, just to make sure im on the right track. What files do you tweak? sip.conf, the ones from avaya and anything else? On 6/22/06, Tom Lynn [EMAIL PROTECTED] wrote: Nope. Let me know if you do. I've suspended my efforts until I see a new version of firmware available on the Avaya web site.

[Asterisk-Users] Re: Showing Current Calls

There is also: show channels verbose show channels concise These may be easier for you to interpret. -- -- Steven http://www.glimasoutheast.org Douglas Garstang [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Can someone recommend the best way to view current calls in progress on

[Asterisk-Users] Re: Out of Office Auto Reply:

Exchange changes http://www.microsoft.com/exchange/techinfo/tips/mailtip01.asp -- -- Steven http://www.glimasoutheast.org Koopmann, Jan-Peter [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Thursday, June 22, 2006 8:13 PM Anthony Rodgers wrote: We use MS Exchange too and,

Re: [Asterisk-Users] Kernel 2.4 / 2.6 and timer

You don't need any boards. only zaptel and ztdummy module loaded and working http://www.voip-info.org/wiki-Asterisk+timer+ztdummy .../zaptel-1-2-X/README.udev litte modyfication in ztdummy.c that was all i needed I've read in different places that if I want to do trunking and meetme on Asterisk

Re: [Asterisk-Users] Realtime voicemail not working

2006/6/23, Mike [EMAIL PROTECTED]: That is what I have in my config file. Pretty much the same thing as the CDR file [general]dbhost=localhostdbname=dbdbuser=userdbpass=pw;dbport=3306 ;dbsock = /tmp/mysql.sock In logger.conf, enable debug mode: debug =

Re: [Asterisk-Users] Meetme max users

We've had over 100 participants spread across 30 meetme rooms on a single server before, and the most we've had in a single meetme room is 46. I don't know of a hard limit for meetme participants and I haven't seen a limit in the code. You would most likely be limited by the resources on your

[Asterisk-Users] How to use G729 decoded voice files?

If you need to use prerecord voicer files for G729 codec, how do you configure them? Do they have to be specially named, copied to their own folder or something? Can asterisk automatically find them even if you use standard names? /Obelix ___

RE: [Asterisk-Users] Realtime voicemail not working

I think you just comment out dbhost statement and leave the dbsock statement in, and it assumes its a localhost connection on the socket specified. Cheers, Adam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: 23 June 2006 14:51 To: 'Asterisk Users

[Asterisk-Users] Tribox - Unistim9.4 Makefile

Anyone had any luck getting the Unistim channel driver to install on Tribox 1.0.5? Ian Cowley Network Security __ This email has been scanned by the MessageLabs Email Security System. For more information please visit

Re: [Asterisk-Users] voip to voip bridge

Extracted from http://www.voip-info.org/wiki-Asterisk+cmd+Dial: ' When options /t/, /T, h, H, w, W or L (with multiple arguments) are applied, Asterisk will remain in the media path, even if /canreinvite=yes'' (a SIP channel option) has been specified.' Then how is it possible to limit a

RE: [Asterisk-Users] best hardphone for Asterisk?

I'll second that. I really like the provisioning features. My customers prefer the 501 because they like the layout and speaker phone functionality. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua West Sent: Friday, June 23, 2006 10:20

RE: [Asterisk-Users] Re: Out of Office Auto Reply:

Should be part of the FAQ for the list, as well as the setting for Exchange 5.5 which a *lot* of orgs still run (we do too) I wonder if the list SW can be modded to automatically plonk any mail with the subject string: Out of Office -Original Message- From: Steven [mailto:[EMAIL

[Asterisk-Users] call quality statistics?

Is it possible to set up some sort of call-quality statistics reporting/logging for IAX2 calls? Something that can keep track of dropped packet / jitter trends? (I know iax2 show channels shows this info for active calls.) Suggestions appreciated! - Mike

RE: [Asterisk-Users] freepbx centos 4 install script?

I'm curious what a manner of speaking is. If I go that route what am I losing? I really just want to make sure whichever route I go I will be able to come here for help and not get blown off because of something non-standard in the packaging I chose. By manner of speaking I mean

[Asterisk-Users] Echocancelwhenbridged

Hi, Can someone tell me what are the valid parameters for the option echocancelwhenbridged? Is it just yes or no, or does it support 128 as well? Also is thier any differnce with using echocancelwhenbridged=128 as opposed to echocancelwhenbridged=yes (assuming that 128 is a valid option).

[Asterisk-Users] Asterisk Users Group - Portugal

Boa tarde, Após alguma experiência com o Asterisk, e com muito ainda para aprender, gostaria de saber se há alguém nesta mailing list que pretenda criar um Asterisk Users Group para Portugal. Visto que acaba sempre por ser uma enorme aprendizagem ( valor acrescentado) a partilha de

[Asterisk-Users] Passing DID to external number?

You already posted this. I answered it yesterday also?p<[EMAIL PROTECTED]><[EMAIL PROTECTED]>From: "Brian McCarey" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Fri, 23 Jun 2006 10:27:19 +0100Subject:

[Asterisk-Users] ISDN

Hi, since last august, I worked on Asterisk and PRI lines, making a good experience. Now I need some information about ISDN BRI: I know that there is no native support in Asterisk and I need some third-part driver. Then I read often about different cards (Junghanss, Eicon, Beronet, etc), different

[Asterisk-Users] New to the list.

Hi all, First of all I wanted to give my salutations to everyone in the list. In second hand, let's see if anyone may help me with a problem i've got. I bought a Developer Kit from Digium (TDM11B + ST2030 + Asterisk book) I installed a new server over [EMAIL PROTECTED] with the

RE: [Asterisk-Users] GXP-2000 and Shared Line Appearances

That feature is called Bridged (or Shared) line appearance. That is one of the things Asterisk cannot do and nobody seems very interested in making it do that because it is apparently not easy. There has been some talk about implementing it but so far there does not seem to be any progress. I

Re: [Asterisk-Users] call quality statistics?

try iax2 show netstats On 6/23/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote: Is it possible to set up some sort of call-quality statistics reporting/logging for IAX2 calls? Something that can keep track of dropped packet / jitter trends? (I know iax2 show channels shows this info for

Re: [Asterisk-Users] Asterisk Users Group - Portugal

Marco, boa tarde (in Brazil) Na realidade tenho interesse em estar participando deste sua idéia também, posso contribuir no que for necessário para que avance a idéia. Cumprimentos Josué I have interest inparticipating of this its idea , I can contribute in that will be necessary so that it

Re: [Asterisk-Users] best hardphone for Asterisk?

My fellow employees like their Polycom 600s even more. Andrei (MPI) Jonathan k. Creasy wrote: I'll second that. I really like the provisioning features. My customers prefer the 501 because they like the layout and speaker phone functionality. -Jonathan -Original Message- From:

[Asterisk-Users] Re: Asterisk-1.2.9.1 e MOH

But to register in the list, I compiled asterisk-1.2.9.1 with addons and he functioned. Thank'sJosué 2006/6/23, Josué Conti [EMAIL PROTECTED]: Hi All Somebody knows asresolv the error below? Already I compiled asterisk-addons-1.2.3, but exactly thus it reports this error, could help me? --

Re: [Asterisk-Users] ISDN

Mimmus wrote: Could some goodwill man summarize this topic for me before I engage myself in the rediscovery of warm water? Read a topic posted a few days ago: ISDN BRI NetJet You will find good advice there. If you need to buy a Cologne chipset card, check here:

Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances

shadowym wrote: That feature is called Bridged (or Shared) line appearance. That is one of the things Asterisk cannot do and nobody seems very interested in making it do that because it is apparently not easy. There has been some talk about implementing it but so far there does not seem to

RE: [Asterisk-Users] best hardphone for Asterisk?

I love my Aastra 9133i with v1.4 firmware. Pretty much everything just works with Asterisk right out of the box and it has all the features I need. -Original Message- From: Jonathan k. Creasy [mailto:[EMAIL PROTECTED] Sent: Friday, June 23, 2006 8:03 AM To: Asterisk Users Mailing

[Asterisk-Users] troubleshooting echo on speakerphone

Hello all, I'm looking to troubleshoot some echo issues and possibly studdering while using the speakerphone with the Intellitouch ITC3002. On the hardware end, i have a 2621 setup as the router with some policy maps for qos, and a 2900XL with cos priority set to 5. I have setup created 2

Re: [Asterisk-Users] Asterisk Users Group - Portugal

Olá a todos! :) Hi all, Thanks for all of your fast replies, further now may be simpler to talk in english. But any one could write in Portguese, any way the important is to make it happen. This weekend i will post here some topics could be interesting: mailing list and a blog or php forum

RE: [Asterisk-Users] Re: Out of Office Auto Reply:

I'd second this motion, this is very very annoying. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Friday, June 23, 2006 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE:

Re: [Asterisk-Users] best hardphone for Asterisk?

The Polycom 501's or 601's are the way to go On 6/23/06, shadowym [EMAIL PROTECTED] wrote: I love my Aastra 9133i with v1.4 firmware.Pretty much everything justworks with Asterisk right out of the box and it has all the features I need. -Original Message- From: Jonathan k. Creasy

RE: [Asterisk-Users] best hardphone for Asterisk?

On Fri, 2006-06-23 at 10:39 -0700, shadowym wrote: I love my Aastra 9133i with v1.4 firmware. Pretty much everything just works with Asterisk right out of the box and it has all the features I need. If cost is important the 9112i would be better. I install all three Aastra models and the sound

RE: [Asterisk-Users] troubleshooting echo on speakerphone

ok, ill try that as well. Funny thing is i don't have these issues using the same phone on my test box at home which is a old 256M Sun Ultra5 connected to a old Netgear Cable/DSL wireless router (can you say 0 (zero) QOS). Terrelle From: Colin Anderson [EMAIL PROTECTED] Reply-To: Asterisk

[Asterisk-Users] Asterisk home on VMWare time sync issues

Hello - I am using AH on VMWARE. I have noticed that the date and time periodically loses sync with the server system time. Does anyone know why this would be, or where if I need to change a setting somewhere so it keeps time properly?Thanks! Alan. Want to be your own boss? Learn how

[Asterisk-Users] Odd SIP error message

As of late, I keep seeing a very odd error message in Asterisk and regardless of how much debugging or verbose I set I can't get more detailed info to find out what exactly is causing the error. It's every few seconds and in no regular pattern either. Jun 23 05:24:17 NOTICE[29057] chan_sip.c:

[Asterisk-Users] QueueMetrics 1.2 released today

Hello list, I am pleased to tell you that we have released a new version of QueueMetrics. The main areas of improvement were the following ones: - Unattended client monitoring: for large call centers who work for third parties, it is possible to have your clients log in directly to

RE: [Asterisk-Users] troubleshooting echo on speakerphone

Different room. Different acoustic characteristics. -Original Message- From: T. Shaw [mailto:[EMAIL PROTECTED] Sent: Friday, June 23, 2006 12:21 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] troubleshooting echo on speakerphone ok, ill try that as well. Funny

RES: [Asterisk-Users] Meetme max users

Hi, Matt: What´s your server specifications that did you use? Best Regards, Cleviton. -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nome de Matt Florell Enviada em: sexta-feira, 23 de junho de 2006 11:38 Para: Asterisk Users Mailing List - Non-Commercial

[Asterisk-Users] Voice calls sent to fax extension

I have a situation that has repeated itself a few times. Someone calls into Asterisk and is connected with a voice extension. At some point during the call, the log shows chan_zap.c: DTMF digit: f on Zap/2-1. At this point, the call is redirected to receive a fax and the Asterisk voice extension

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