[asterisk-users] Polycom Speakerphone

Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? Thanks, Eric

Re: [asterisk-users] Anyone use the Linksys phones?

Interesting comment on the speakerphone. Have you found a reasonably priced desk set with a good speakerphone? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Monday, September 24, 2007 2:45 AM To: Asterisk Users Mailing List -

Re: [asterisk-users] Skype + Asterisk

by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Jacksch http://www.jacksch.com ___ Sign up now for AstriCon 2007! September 25

Re: [asterisk-users] Follow-me challenge

Sorry, didn¹t realize you were sending the call out on a Zap channel. Yes, as soon as the call goes out a Zap channel it is ³answered² as far as Asterisk is concerned. I send out all my findme traffic via SIP. On 2006-12-19 21:19, Chris Johnson [EMAIL PROTECTED] wrote: On 12/18/06, Eric

Re: [asterisk-users] Inform callers on recorded/monitored number.

such scripts. Thanks Angel - Original Message From: Eric Jacksch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 19, 2006 1:02:34 PM Subject: Re: [asterisk-users] Inform

Re: [asterisk-users] Follow-me challenge

___ --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 -- Eric Jacksch +1 613 860-0964 Ottawa +1 647 722-3544 Toronto +1 514 907

Re: [asterisk-users] Inform callers on recorded/monitored number.

the best spam protection around http://mail.yahoo.com ___ --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 -- Eric

[asterisk-users] SIP Quality Metrics

I've been reading about ${RTPAUDIOQOS}, which supposedly contains call quality metrics, but I can't seem to convince Asterisk to grab it for me after the call. Anybody have any luck with it? Anybody know how to get Asterisk to executive a command after the call is over? Thanks, Eric

[asterisk-users] SIP Quality Metrics

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[asterisk-users] Findme problem

Greetings all, I've been working on having Asterisk put a call through to two different numbers, and give the call to the first one that acknowledges by pressing the 1 key. I found an example on the wiki, but I can't get it working. When I answer the call I hear the message telling me to

[Asterisk-Users] MixMonitor and filenames

A client wants to record all calls to a specific extension. MixMonitor seems to do the job, but is there a way to get it to append something to the filename for each call? Right now it overwrites the file every time a call comes in. I realize there is an append option, but I'd prefer a separate

[Asterisk-Users] MWI Problem

When I register to a remote Asterisk system using IAX2, I can see it notifying my Asterisk box that I have voicemail waiting. How can I get Asterisk to use that information and send WMI to one or more of my SIP phones? Thanks, Eric ___ --Bandwidth and

[Asterisk-Users] Interrupting a call

Greetings all, I've tried out chanspy, but what I'm really looking for is the ability to interrupt a call (i.e. barge in for emergency purposes). Has anyone found a way to do that with Asterisk? Regards, Eric___ --Bandwidth and Colocation

RE: [Asterisk-Users] Interrupting a call

: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Mendoza Sent: April 3, 2006 19:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Interrupting a call FOP ? Jorge Eric Jacksch wrote: Greetings all, I've tried out chanspy, but what I'm

Re: [Asterisk-Users] Can some bady help me ???

When you say command not found, is your system not finding make or is make running and complaining that it can't find a Makefile ? If the former, you may not have make (usually nmake) installed, or it is not in your path. If the latter, you're probably in the wrong directory. Make looks for

[Asterisk-Users] Forwarding inbound calls right back out

I have calls coming in via SIP (a DID) and I want to forward them right back out to my cell. If I do it in one step, (as if 2125551212 was the DID, and 202111 was my cell number) exten = 2125551212,1,Dial(SIP/${PROVIDER}/1202111,60) The call comes in via sip, my system sends the invite

[Asterisk-Users] German Termination and DIDs

Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Problem with two simultaneous calls

Greetings all, I have asterisk running on a high speed connection, sending and receiving calls from a service provider. I can receive calls from the PSTN into voicemail, etc., and it works fine. However, when I try to redirect the call back out to the PSTN (for example sending it to my cell

[Asterisk-Users] Codecs for fax traffic

Are there any codecs that are particularly good for fax traffic? Any to avoid? --- Eric Jacksch [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] Wildcards and variable number of digits

of digits works. Any ideas? [globals] P1=Phone/phone0 P2=Phone/phone1 EXTPHONE=6138600964 EXTNAME=Eric Jacksch VPC_ID=x VPC_PW=x ; Free world dialup info FWDNUMBER=483835 FWDCIDNAME=Eric Jacksch FWDPASSWORD= FWDRINGS=${P1} FWDVMBOX=2201 ; extension

Re: [Asterisk-Users] Wildcards and variable number of digits

these. Lyle - Original Message - From: Eric Jacksch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 05, 2004 7:48 AM Subject: Re: [Asterisk-Users] Wildcards and variable number of digits

Re: [Asterisk-Users] Wildcards and variable number of digits

,Macro(dialwfd,${EXTEN:1},60) Try these. Lyle - Original Message - From: Eric Jacksch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 05, 2004 7:48 AM Subject: Re: [Asterisk-Users] Wildcards and variable

Re: [Asterisk-Users] Wildcards and variable number of digits

in extensions.conf. Those will fix the problem with not enough time to dial the complete number. Lyle - Original Message - From: Eric Jacksch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 05, 2004 9:31 AM Subject: Re

Re: [Asterisk-Users] Wildcards and variable number of digits

Thanks, I'm really stuck...perhaps it's something specific to the quicknet phonejack card. No matter what I do, I can't configure a dialing pattern for a phone plugged into the card unless the pattern has a fixed number of digits. Everyone tells me that exten = _9X.,1, ... Should get anything

[Asterisk-Users] Number of digits

Perhaps this will help... I have a phone connected to a QuickNet PhoneJack card. When I pick it up, I get a dial tone. When I dial a certain number of digits, the call is processed by Asterisk. The question: How does Asterisk determine how many numbers to let me dial? I'm banging my head

Re: [Asterisk-Users] Wildcards and variable number of digits

with a mode buffered which is otherwise the same as dialtone Eric Jacksch wrote: Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten = _7., How do I get Asterisk to wait until the user is finished dialing

Re: [Asterisk-Users] Wildcards and variable number of digits

until the user presses the pound (#) key to send the phone number to the pbx. The features can be toggled on/off any time by dialing *1# or *0# or in the config file with a mode buffered which is otherwise the same as dialtone Eric Jacksch wrote: Greetings, I'm having

Re: [Asterisk-Users] Wildcards and variable number of digits

] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Jacksch Sent: Sunday, September 05, 2004 2:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Wildcards and variable number of digits Not sure I understand..does that help my problem

RE: [Asterisk-Users] Wildcards and variable number of digits

an extension defined within the current context wins over the included wildcard context. S= bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Jacksch Sent: Sunday, September 05, 2004 2:50 PM To: Asterisk Users Mailing List - Non

Re: [Asterisk-Users] Wildcards and variable number of digits

. This way an extension defined within the current context wins over the included wildcard context. S= bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Jacksch Sent: Sunday, September 05, 2004 2:50 PM To: Asterisk Users

Re: [Asterisk-Users] Wildcards and variable number of digits

I don't think so, but I'm very new to Asterisk - is there an easy way to check? On 2004-09-05 20:56, Craig Guy [EMAIL PROTECTED] wrote: Do you have early dial enabled at all? Craig - Original Message - From: Eric Jacksch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non

[Asterisk-Users] Wildcards and variable number of digits

Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten = _7., How do I get Asterisk to wait until the user is finished dialing instead of trying as soon as it gets the second digit? I can use _7XXX, and dial the FWD 3-digit