[Asterisk-Users] time update (7905)

Hi everyone,   I'm trying to update time on all Cisco 7905 phones in my company.. is there some way to do it from asterisk?   Thanks,   Tomislav   ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] Re: Re: Cisco 7970

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

[Asterisk-Users] how to dial to hardphone by kiax

hello ,i have a problem about asterisk configuration.i have wildcard that use it connect to pstn ,now i configured it and config iax.conf ,extension.conf.i can use kiax to talk with other kiax client,but how can i talk with tradition hardphone,or cell phone.i used iax client,who can help me?

[Asterisk-Users] DNS lookup not working

Hello All, I recently upgraded my asterisk, and it seems that something has changed with regard to using domains in sip.conf host settings. Nothing seems to work unless I use the actual IP now. I have tried putting entries in hosts, but still no joy. Is anybody aware of something that needs to

Re: [Asterisk-Users] restart problem

On Tue, Mar 28, 2006 at 11:11:41AM +0800, Avneet Bansal wrote: > hi, > i m facing a problem of asterisk not starting from rc.local, i can see the > files .ctl and .pid existing in the respective directory. but asterisk > -grc doesnt work. lines are not picked up. because asterisk is already ru

RE : RE : [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6analogue lines)

This card doesn't permit to support Mark Spencer's company and project. This card has no hardware echocan and use only the X100M and S110M clones modules. This two reason are sufficient for me. -Message d'origine- De : Krzysztof Drewicz [mailto:[EMAIL PROTECTED] Envoyé : lundi 27 mars 20

[Asterisk-Users] TDM11B desperate Help wanted

Hello:After configuring FXS and FXO channels of a TDM11B card, I can make calls from the telephone attached to the TDM11B card to the outsite (the PSTN, analog line), The problem is when I try to dial from the PSTN to my asterisk box (it has to make ring the handset on the TDM card), but I got infi

RE: [Asterisk-Users] Voicemail limit?

Hi Brad, On my sip.conf I have 83 users also in my voicemail.conf but when I call the users above 70 it prompts me "User Not Found". Any idea regarding this? Thanks, Ryan At 04:12 AM 3/24/2006, Watkins, Bradley wrote: I don't think there's any kind of (significantly small, anyway) l

Re: [Asterisk-Users] Dell 2850 w/TDM2400?

Nick Hoffman wrote: On Tue March 28 2006 10:33, "Kevin P. Fleming" <[EMAIL PROTECTED]> wrote: Kerry Garrison wrote: Does anyone know if a TDM2400 will fit into a Dell 2850? It will fit, but you will need to solve the power supply problem if you intend to use FXS ports on it :-) Why is that?

[Asterisk-Users] restart problem

hi,i m facing a problem of asterisk not starting from rc.local, i can see the files .ctl and .pid existing in the respective directory. but asterisk -grc doesnt work. lines are not picked up.from the shell if execute asterisk "asterisk" it starts working fine. this is happening only on one of t

RE: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)

The iMate is availble here, as well as the O2 XDAII. I have the O2 running here without too many issues..(appart from WiFi Sucking My Battery's will to live) With SJPhone seems to be mostly stable. The ECS-IAX PDA Client is WAY too unstable at the moment...conectivity/voice quality issues... Dav

RE: [Asterisk-Users] FXO without answer supervision

> I've got both the A200d and TDM04b in the same box with fc3 and all > works very well. I'd suspect a config issue in your box of some sort. > I didn't have to do anything special. This is my first FXO/FXS card, so a config issue is likely, but like you said they're pretty straight forward. In

Re: [Asterisk-Users] FXO without answer supervision

Dan Austin wrote: Simple question that google hasn't helped much with (likely poor search terms) I just installed a Sangoma A200 with overall good results. Initial tests with both incoming and outgoing calls were very positive. Until I made a normal call that lasted more than 30 seconds. I se

Re: Re[2]: [Asterisk-Users] Re: Alarm on Unicall

Im triyed with span=1,0,0,cas,hdb3 with no results.Mi msn is [EMAIL PROTECTED]Regards and thakns.2006/3/27, Melcon Moraes < [EMAIL PROTECTED]>:Well,Did you try "span=1,1,0,cas" only? I think you can have hdb3 or ami for coding but nothing.Do you have some kind of IM?[]'sMM -Original Message

Re: [Asterisk-Users] Polycom 501 Output volume

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

Re: [Asterisk-Users] Dell 2850 w/TDM2400?

no, fxs ports require you to use a 4-pin molex connector to power the pci card to generate ring voltages from the power supply. most dell/hp/whoever small form-factor units don't provide this, unless you can kludge something up from a splitter or power bus feeding a drive backplane or somethin

Re: [Asterisk-Users] definity g3 voicemail

Yeah... I am doing that one now with a merlin system... Sean C F wrote: Well, I did it using DTMF tones on analog channels, it's on the wiki. On 3/27/06, Sean Cook <[EMAIL PROTECTED]> wrote: My understanding is that the SMDI is a serial interface that passes data about the call to the syst

Re: [Asterisk-Users] definity g3 voicemail

Well, I did it using DTMF tones on analog channels, it's on the wiki. On 3/27/06, Sean Cook <[EMAIL PROTECTED]> wrote: > My understanding is that the SMDI is a serial interface that passes data > about the call to the system for voicemail and pass MWI info back to the > avaya. It is the definity

Re: [Asterisk-Users] TDM400P busy

Yes, I changed.Thank youOn 3/27/06, Infobox Peru <[EMAIL PROTECTED]> wrote: Maybe your lines use polarity reversals for hangup detection.On 3/27/06, Il Neofita <[EMAIL PROTECTED] > wrote: Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it takes the line occup

Re: [Asterisk-Users] Dell 2850 w/TDM2400?

On Tue March 28 2006 10:33, "Kevin P. Fleming" <[EMAIL PROTECTED]> wrote: > Kerry Garrison wrote: > > Does anyone know if a TDM2400 will fit into a Dell 2850? > > It will fit, but you will need to solve the power supply problem if you > intend to use FXS ports on it :-) Why is that? Do FXS ports

Re: [Asterisk-Users] definity g3 voicemail

My understanding is that the SMDI is a serial interface that passes data about the call to the system for voicemail and pass MWI info back to the avaya. It is the definity side that I am clueless on... C F wrote: On 3/27/06, Sean Cook <[EMAIL PROTECTED]> wrote: Is anyone using * to provid

Re: [Asterisk-Users] sipura spa2 + asterisk bug ?

Steve Kennedy wrote: On Tue, Mar 28, 2006 at 01:20:06AM +0300, Tofik Suleymanov wrote: How to reproduce this bug (?) : 1. register sipura spa2 with 2 lines on asterisk. 2. use first line to call somewhere. 3. while using first line try to call from second line somewhere else in 3 step i hea

[Asterisk-Users] BUG 0003710 - RE: Transfer Calls - REFER

I just realised my problem seems to be related to bug 0003710 - "0003710: [patch] Consultative transfers between asterisk servers". It's unclear from the bug info if this problem has been resolved yet. Anyone know? Doug. > -Original Message- > From: Douglas Garstang > Sent: Monday, Mar

Re: [Asterisk-Users] Dell 2850 w/TDM2400?

Kerry Garrison wrote: > Does anyone know if a TDM2400 will fit into a Dell 2850? It will fit, but you will need to solve the power supply problem if you intend to use FXS ports on it :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asteri

Re[2]: [Asterisk-Users] Re: Alarm on Unicall

Well, Did you try "span=1,1,0,cas" only? I think you can have hdb3 or ami for coding but nothing. Do you have some kind of IM? []'s MM -Original Message- From: acriollo <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Cc: Sent: Mon, 27 Mar

Re: [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)

I have a client with an installation with 3 TDM400P cards. 6 FXO, 6FXS ports. I followed the txgain/rxgain instructions and now have no echo problems. The only problem I have now is the flaky network the SIP phones are accessing asterisk with. (you should see the wiring there, ugh). It's

Re: [Asterisk-Users] Re: Alarm on Unicall

Sorry , I meanni tryed with span=1,1,0,cas ,and the results are the same.CRC4, is unavailable.The Telco says the everyting is Ok on his installation, but the problem persiste.the zttool never show the yellow alarm, just the asterisk. We made few months ago a loop test with the telco, and he says th

[Asterisk-Users] Dell 2850 w/TDM2400?

Does anyone know if a TDM2400 will fit into a Dell 2850?  Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com   ___ --Bandwidth

Re: [Asterisk-Users] Re: Alarm on Unicall

It looks like sliping frames also. Steve has a divine rod to check if the frames are sliping at the very moment this happens. Putting all the jokes aside, how do you know that hdb3 isn't supported? Maybe you need some sort of CRC4, so try "span=1,1,0,cas,hdb3,crc4" What does your telco says ab

[Asterisk-Users] Re: Alarm on Unicall

Sorry , I meanni tryed with span=1,1,0,cas ,and the results are the same.2006/3/27, acriollo <[EMAIL PROTECTED]>: Thanks all for four feedback ,i tryed with span=1,1,0,cas ,hdb3 is not supported, and the results are the same.Some one toldme that this can be a grounding problem.  There is a posibili

Re: [Asterisk-Users] definity g3 voicemail

On 3/27/06, Sean Cook <[EMAIL PROTECTED]> wrote: > Is anyone using * to provide voicemail to a definity system? I > understand with the new SMDI functionality in trunk that this will be > easier to provide some of the integration features. I have done this on other Avaya systems, you got any idea

Re: [Asterisk-Users] txfax problem

I am using Asterfax but I also tried the following with the same results:   [macro-testfax]exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/QWWWxx1.tif))exten => s,2,txfax(${FAXFILE}|caller)   [default]exten => 7001,1,Answerexten => 7001,n,Dial(ZAP/97/94805175098|30|gM(testfax))exten => 700

Re: [Asterisk-Users] [1.2.5] DTMF not being set correctly (RESEND)

Rich Adamson wrote: I am having trouble getting DTMF mode to be set to inband on incoming calls. I have the following set, and for some reason the connection is still negotiated with rfc2833. [outbound] type=friend secret=XXX username=XXX authuser=XXX host=XXX.XXX.XXX.XXX context=in

[Asterisk-Users] definity g3 voicemail

Is anyone using * to provide voicemail to a definity system? I understand with the new SMDI functionality in trunk that this will be easier to provide some of the integration features. Looking for some hints on the definity setup and anything on the SMDI side. Anyone with a working solutions

[Asterisk-Users] Transfer Calls - REFER

I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107. IP addresses have been changed to protect the innocent. Here's the REFER that the phone at 2944093 sends directly to Asterisk: U 216.186.128.68:5060 -> 216.186.142.203:5060 REFER sip:[EMAIL PROTECTED] SIP/2.0. Via:

Re: [Asterisk-Users] Polycom SIP 1.6.5 and Bootrom 3.1.3 - Download link

Hi guys,     I got a copy of both of these.  I've uploaded them to one of my servers.  Here is the download link:   Bootrom:  http://support.gafana.com/polycom/bootrom_313.zip SIP: http://support.gafana.com/polycom/SoundPoint_IP_SIP_1_6_5.zip   - Gabe     - Original Message - Fro

Re: [Asterisk-Users] sipura spa2 + asterisk bug ?

On Tue, Mar 28, 2006 at 01:20:06AM +0300, Tofik Suleymanov wrote: > How to reproduce this bug (?) : > 1. register sipura spa2 with 2 lines on asterisk. > 2. use first line to call somewhere. > 3. while using first line try to call from second line somewhere else > in 3 step i hear fast busy tones

RE: [Asterisk-Users] txfax problem

James,   Which fax application are you using?   MD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James HawksSent: Monday, March 27, 2006 5:29 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] txfax problem I can not get txfax to work with a multiple page ti

[Asterisk-Users] SER inside of Asterisk is SCALABLE ?

Hi, I read somewhere if you need an scalable Gateway for huge number of agent it is better to use SER ( OpenSER ) inside of your asterisk. My question is that is it really true ? and why ? Also is there any way to integrate SER with asterisk in this way that the SER itself won't route any call

Re: [Asterisk-Users] How to disable event_log?

On 3/27/2006, "Roger Schreiter" <[EMAIL PROTECTED]> wrote: >Hi, > >how can I disable event_log in order to reduce >hard disk activity? > >I can't find any hints in conf files. Roger. check logger.conf under the general heading. ; This determines whether or not we log generic events to a file ; (

RE: [Asterisk-Users] Bluetooth headset in handsfree modewith SJPhoneorX-lite

I am able to pickup, hangup, and flash, using the buttons on the phone with all of the soft clients and both of the headsets I mentioned below. I don't believe I had to change anything on the client side, just had to get the ctp (cordless telephone profile) working in the bluetooth stack, which was

Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)

"Peter Bowyer" <[EMAIL PROTECTED]> wrote: > How many companies have deployed DECT in their buildings? Thousands, perhaps tens of thousands. DECT technology is very big in Europe. It's not so widespread in North America, primarily because the FCC allocated radio frequencies are different from th

[Asterisk-Users] Re: Alarm on Unicall

Thanks all for four feedback ,i tryed with span=1,1,0,cas ,hdb3 is not supported, and the results are the same.Some one toldme that this can be a grounding problem.  There is a posibility ?The messages are peteated all the time, then i lost some calls and some times we can not access to the PSTN. R

Re: [Asterisk-Users] TDM400P busy

Maybe your lines use polarity reversals for hangup detection.On 3/27/06, Il Neofita <[EMAIL PROTECTED] > wrote:Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it takes the line occupied. ___--Bandwidth and Colocati

Re: [Asterisk-Users] sipura spa2 + asterisk bug ?

That ATA cannot do 2 simultaneous calls with g729. The second call is probably trying to use ulaw, alaw or g723. Are you sure any of them are enabled for that extension? Julian. On 3/27/06, Tofik Suleymanov <[EMAIL PROTECTED]> wrote: > Hello, > > How to reproduce this bug (?) : > > 1. register si

[Asterisk-Users] ANNOUNCE: WIST - Web Interface for SIP Trace

Hi Folks, I'm glad to announce WIST for SIP debug/trace dialogs. This software born as a prof concept of the idea to capture SIP traffic from a remote host (SIP Proxy, Gateway, etc) and show up alive SIP messages about an specific dialog (filtered by From SIP user) to help our tech support team t

RE: [Asterisk-Users] oh323 signal update support

Hello We have detected that the newer Cisco IOS versions include a SignalUpdate message after each alphanumeric UserInputIndication. Did the oh323 asterisk module support SignalUpdate? Has anybody know something? Thanks Jsalas ___ --Bandwidth and C

Re: [Asterisk-Users] Authorization by ip

Sorry, don't know where my mind was... with host=specific.ip.address, the client doesn't even need to register with * for * to be able to _try_ to send calls to it. Heck I try to be quiet and learn, but sometimes my fingers send the emails when I'm sleeping... Eric "ManxPower" Wieling wrot

[Asterisk-Users] How to disable event_log?

Hi, how can I disable event_log in order to reduce hard disk activity? I can't find any hints in conf files. Must I hack the source code or even use brutal methods like creating a dir called event_log in the log dir, in order to prevent asterisk from creating an event_log file? (Just chmod a-w

Re: [Asterisk-Users] Receptionist Phones

The 360 has an expansion unit. It adds 42 extensions. Darrell S. Long BestWeb Corporation Daniel Hazelbaker wrote: Hmm, which phone from Snom are you using for this? I've looked around their website and I can only find 3 VoIP phones, the 300, 320 and 360. The 360 by the looks of it on

Re: [Asterisk-Users] Receptionist Phones

You can get an extension module that adds another 42 buttons. Paul Hales Technical Manager AsteriskIT - Original Message - From: "Daniel Hazelbaker" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, March 28, 2006 8:22 AM Subject: Re: [Aste

[Asterisk-Users] txfax problem

I can not get txfax to work with a multiple page tiff image. It works fine with a single page tiff image. Has anyone had this problem before and if so is there a fix?     Thank You,James HawksCTO CustomerFunding.com Inc.(877) 784-1030 x206 ___ --Ba

Re: [Asterisk-Users] Alarm on Unicall

Hi, You will notice the reports are of alarms being cleared. If this just happens at start up, it may just be reports of the E1 initially setting down after the link is established. If you keep getting alarm reports during normal operation, your link is probably unreliable. Steve acriollo

Re: [Asterisk-Users] Receptionist Phones

Hmm, which phone from Snom are you using for this? I've looked around their website and I can only find 3 VoIP phones, the 300, 320 and 360. The 360 by the looks of it only has 12 buttons you can assign to different extensions; am I missing something or is that the phone and you just do 1

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

Hi, Are you able to answer calls by pressing the answer/hangup button on your headset or are you using the computer to answer the calls with a SIP phone. I can get the head set to work fine with the PC and sip phones what I cannot do is get the answer/hangup button on the head set to actually

[Asterisk-Users] sipura spa2 + asterisk bug ?

Hello, How to reproduce this bug (?) : 1. register sipura spa2 with 2 lines on asterisk. 2. use first line to call somewhere. 3. while using first line try to call from second line somewhere else in 3 step i hear fast busy tones on second line and asterisk console gives me this short error:

[Asterisk-Users] TDM400P busy

Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it takes the line occupied. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://li

Re: [Asterisk-Users] Re: Alarm on Unicall

Besides the messages, can you make/receive calls normally? You should use "span=1,1,0,cas,hdb3", since you're connected to the PSTN and they will provide you the timing sync source. Do you have zttool compiled? What does it say? []'s MM acriollo wrote: This is my unicall.conf [channels] lan

[Asterisk-Users] Intel Compilation Questions - Asterisk 1.2.5/6

Hi, I use Centos 4.2 (kernel 2.6) on an Intel P4 at my house and i plan to use that machine to build asterisk. However the target system (a small test server) is a latest Intel Celeron 2.x Ghz but with Centos 4.2 kernel 2.6.   I don't build things a lot (thanks for the RPMs) so i better ask before

Re: [Asterisk-Users] Receptionist Phones

Yes - set up about 10 of them at a business last year. Monitoring is fine - picking up calls is a bit iffy at the best of times. (that is, picking up a ringing call by pushing the extension button. *8 works fine) Paul Hales Technical Manager AsteriskIT - Original Message - From: "Darrel

RE: [Asterisk-Users] Linksys and sip

Are they compatable and if so how to configure the pap2 or the r41p2 thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aster

Re: [Asterisk-Users] Receptionist Phones

Do you have experience with the Snom phones? We have not had much success getting them to work under Asterisk as a receptionist phone. Specifically, the ability to monitor and pick up calls ringing on other extensions has been a problem. Darrell S. Long BestWeb Corporation [EMAIL PROTE

[Asterisk-Users] Question about Polycom 601 and expansion module.

96 handle_uc_event: > Unicall/9 Alarm No Alarm raised, Yellow Alarm cleared > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: > Unicall/10 event Alarm > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: > Unicall/10 Alarm masks 0x 0x0004 > Ma

Re: [Asterisk-Users] Receptionist Phones

Using a Snom phone, you can monitor a lot more extensions, so I figure it's got to be a Polycom issue. Paul Hales Technical Manager AsteriskIT - Original Message - From: "Daniel Hazelbaker" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday,

[Asterisk-Users] Re: Alarm on Unicall

This is my unicall.conf[channels]language=escontext=incoming    ; Contexto pod defaultusecallerid=yeshidecallerid=no;restrictcid=nocallwaitingcallerid=no threewaycalling=notransfer=yescancallforward=yescallreturn=noechocancel=yesechocancelwhenbridged=yese

RE: [Asterisk-Users] Master.csv Shell Script

Thanks that is exactly what I was looking for! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan & Company, LLC Sent: Monday, March 27, 2006 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Ma

RE: [Asterisk-Users] Config File Management

How does Fast AGI help? -Original Message-From: Giovanni Miano [mailto:[EMAIL PROTECTED]Sent: Monday, March 27, 2006 1:00 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Config File ManagementYou can use FastAGISee http://www.ast

Re: [Asterisk-Users] Authorization by ip

Does this work the same with IAX? - Original Message - From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, March 27, 2006 12:44 PM Subject: Re: [Asterisk-Users] Authorization by ip meaning, when you

Re: [Asterisk-Users] Authorization by ip

It does? All this time I thought that permit= and deny= is what limited access! Check the docs, host= is for OUTGOING, permit/deny is for INCOMING. Mojo with Horan & Company, LLC wrote: meaning, when you put host=dynamic in sip.conf, it doesn't matter what ip the client comes from. if you put

Re: [Asterisk-Users] Authorization by ip

meaning, when you put host=dynamic in sip.conf, it doesn't matter what ip the client comes from. if you put instead: host=www.xxx.yyy.zzz then it _does_ matter where the client comes from, it must be that IP. Giovanni Miano wrote: You can use in sip.conf tag "host" host=192.168.1.1

Re: [Asterisk-Users] Call Waiting Issues

Brad Glonka wrote: I have two call waiting problems. I have a POTS line into and FXO port and telephones on an FXS port 1) I can't seem to use the flash button(on the phone) to answer a call waiting call. I see the callerid coming though and here the call waiting tone, but I just can't s

Re: [Asterisk-Users] Caller ID length

Please be aware, when connecting some versions of Avaya Legen/magix system to Asterisk using a PRI, and the CIDname is longer than 15 characters, then the Avaya will reset the dchannel and all calls in progress will be dropped, I learned this the hard way. There are some other things that will make

Re: [Asterisk-Users] Call Waiting Issues

You have to flash the FXO not your phone, use features.conf to accomplish that. On 3/27/06, Brad Glonka <[EMAIL PROTECTED]> wrote: > I have two call waiting problems. > > I have a POTS line into and FXO port > and telephones on an FXS port > > 1) I can't seem to use the flash button(on the phone)

[Asterisk-Users] FXO without answer supervision

Simple question that google hasn't helped much with (likely poor search terms) I just installed a Sangoma A200 with overall good results. Initial tests with both incoming and outgoing calls were very positive. Until I made a normal call that lasted more than 30 seconds. I setup the FXO with kew

Re: [Asterisk-Users] Receptionist Phones

Yes, I keep reading on the mailing list archives and the wikis that (wether or not it is indeed a Asterisk issue) Polycom keeps saying that an issue with Asterisk prevents you from monitoring more than 7 total (not per sidecar) extensions. Daniel On Mar 27, 2006, at 12:08 PM, Justin Moore

Re: [Asterisk-Users] On site installtion Tech. wanted

I am able to provide local installation, configuration, and troubleshooting in the northwest region (Portland to Seattle and surrounding).   Please feel free to inquire for further details.   Richard Amerman 7 Tech NW 360-931-2721  On 3/25/06, Bart Fisher <[EMAIL PROTECTED]> wrote: Maybe I could

Re: [Asterisk-Users] Master.csv Shell Script

If you've got PHP installed, here's one I made for our office: http://horanappraisals.com/asterisk/total_account_codes/ Run it with no parameters to check Master.csv in the current directory, or pass the filename to parse as the first parameter. # ./total_account_codes /var/log/asterisk/cdr-c

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

On 3/27/06, Daniel Hazelbaker <[EMAIL PROTECTED]> wrote: > I have seen that the polycom setup (601+sidecar) works but only for up to 7 > phones From what I've seen, each sidecar supports up to 14 additional stations. Three of those along with the 5 buttons on the 601 comes up to 47 on my calculat

Re: [Asterisk-Users] Alarm on Unicall

What about some unicall.conf and zaptel.conf lines? []'s MM acriollo wrote: Hi all, any body can tell me why i am receiving this message in my sever ? I have running * with 10 Digital Lines, but i am receiving a lot of times this message . Is a software issue or is a hardware issue ? Regard

Re: [Asterisk-Users] Call Simulator

You can use dialout filehttp://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+auto-dial+out&preview=20 2006/3/27, Steve Totaro <[EMAIL PROTECTED]>: SIPPS is one, I would like to hear of others.   Of course you could create a dialplan that loops calls in and out.   Thanks,

Re: [Asterisk-Users] Authorization by ip

You can use in sip.conf tag "host"host=192.168.1.12006/3/27, Sam Tam <[EMAIL PROTECTED] >:Can somebody send me a config of how to authorize SIP client by IP?Sam ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSC

RE: [Asterisk-Users] Ability to put call on hold via manager?

I was thinking about that as an option. Basically I am integrating a CRM call center app with * and want the agents to be able to click a radio button to put callers on hold. They only have analog headsets with on-hook and off-hook. It seems like parking and un-parking the call would be pr

Re: [Asterisk-Users] Config File Management

You can use FastAGISee http://www.asteriskjava.org2006/3/27, David Gomillion <[EMAIL PROTECTED] >:Sorry for thread breaking... I'm on digest.>> I'm curious (ok, well I admit it - it's for perosnal gain) what >> methods people are using to manage asterisk config files when they>> have multiple aster

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

Hi, You need to have completely replaced the Microsoft driver, because it doesn't support the headset or ctp Bluetooth profiles. This gave me fits! I followed the instructions at http://www.windowsdevcenter.com/pub/a/windows/2005/07/05/bluetooth.html and it works with both a Plantronics and a Mo

[Asterisk-Users] queue caveats

According to http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue, under the "Notes" section: "Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option above). SIP transfers result in the Agent remaining affiliated with the cal

Re: RE : [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6analogue lines)

[EMAIL PROTECTED] wrote: > Hi, > > Jump to a TDM2402E for 6 POTS lines with hardware echocan. > Only one IRQ used, and easy future extensions by adding modules. > Have anyone here used a clone i.e. A1200P-01 (A1200P + 1 FXO100 module) ? ___ --Bandwi

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

Forgoe the patch, just upgrade to 1.2.6. The changelog lists it as a fix from 1.2.5 to 1.2.6. >I'm a bit newbie, could you tell me how to i apply the patch? > >Thanks in advance >Marco Mouta > >On 3/27/06, Benoit Panizzon <[EMAIL PROTECTED]> wrote: >> On Friday 24 March 2006 16:05, Benoit Panizzo

[Asterisk-Users] TE 205P/A102 fit in hp dc7600?

Hi   I would like to know if the TE 205 fit in a hp dc7600? what about the A 102 from Sangoma?   Thanks   Jose Manuel Cortes___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Ability to put call on hold via manager?

You could park it to parking extensiones. Does it help you? Steve Totaro escribió: Does anyone know if there is built in ability to put call on hold via the manager interface? Thanks, Steve Totaro http://www.asteriskhelpdesk.com ___ --Bandwid

[Asterisk-Users] Master.csv Shell Script

Im not looking for anything super detailed, just something to run through the master.csv file and give total time per account code. . . .does anyone out there have a script like this I could work from? ___ --Bandwidth and Colocation provided by Easynews.

[Asterisk-Users] Wanted: Cd-bootable Fedora+Asterisk

I'm in search someone who would be interested in developing a Fedora-baed Asterisk system that is bootable from a CD or possible flash. I am aware of the various commercial and free solutions out there, but none I have found suit our needs...mainly because they are not easily extensible and/or upg

[Asterisk-Users] Call Waiting Issues

I have two call waiting problems. I have a POTS line into and FXO port and telephones on an FXS port 1) I can't seem to use the flash button(on the phone) to answer a call waiting call. I see the callerid coming though and here the call waiting tone, but I just can't seem to answer it. The

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

We may end up using a software solution, but there are two main issues with a software solution (for us at least): 1) For us in particular, our receptionists have ALWAYS (for the past 15 years at least) used a physical switchboard style for "routing" and seeing availability. From past hard

[Asterisk-Users] Ability to put call on hold via manager?

Does anyone know if there is built in ability to put call on hold via the manager interface? Thanks, Steve Totaro http://www.asteriskhelpdesk.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

[Asterisk-Users] Alarm on Unicall

Hi all,any body can tell me why i am receiving this message in my sever ?I have running * with 10 Digital Lines, but i am receiving a lot of times this message .Is a software issue or is a hardware issue ? Regards.Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: Unicall/5 event A

Re: [Asterisk-Users] Searchable forums

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

[Asterisk-Users] Asterisk 1.2.6 and Zaptel 1.2.5 Released

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

[Asterisk-Users] Asterisk 1.2.6 and Zaptel 1.2.5 Released

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

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

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

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

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

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