[asterisk-users] Re: Sangoma A200D and DTMF Detection

would suggest sending an email to [EMAIL PROTECTED]. Rana Dutt Softel Solutions rdutt at softelinc dot com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE

=rfc2833callerid=John 280context=company_xmailbox=280nat=yescanreinvite=noqualify=5000We are using Asterisk 1.2.5 with standard .conf files. We are not using realtime or databases. Any help would be highly appreciated. Rana Dutt Softel Solutions [EMAIL PROTECTED

[Asterisk-Users] Jittery Linksys/Sipura meetme conference fixed

experienced a similar problem. Rana Dutt Softel Solutions www.softelinc.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] Jittery meetme conference using Linksys 942 phones

We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use

[Asterisk-Users] SIP phone failover using DNS SRV?

Has anyone successfully had a SIP phone fail over from Asterisk Server A to Server B using DNS SRV? If so, which phone worked for you? I'm assuming you set up your DNS SRV records so that the IP addresses of A and B are associated with the same name, and both servers have equal priority

Re: [Asterisk-Users] Enabling rtcachefriends prevents phones from calling each other

... Please, please add this option. If you send me a patch, I will gladly volunteer to test it thoroughly. Having both MWI working and multiple servers working is a must for us. Thanks much, Rana Dutt ___ Asterisk-Users mailing list Asterisk-Users

[Asterisk-Users] Quintum Tenor DX

that requires 8 T1s, the improved throughput and redundancy could outweigh these cons. Rana Dutt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Problem with AstTapi

'Zap/1-1' Any help would be much appreciated. Rana Dutt Softel, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[Asterisk-Users] Outlook reports internal error after using AstTapi

'Zap/1-1' Any help would be much appreciated. Rana Dutt Softel, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[Asterisk-Users] RE: multiline IP hardphone w/ FDX speakerphone?

you can get them for less than $200, I paid about $240 each. Rana Dutt -- Message: 8 Date: Mon, 6 Sep 2004 19:24:41 +0200 From: Stewart Nelson [EMAIL PROTECTED] Subject: [Asterisk-Users] multiline IP hardphone w/ FDX speakerphone? To: [EMAIL PROTECTED] Message-ID: [EMAIL

[Asterisk-Users] My Cisco 7940 is not registering with Asterisk

disallow = all allow = ulaw allow = alaw [201] type=friend username=201 secret= host=dynamic context=dialout callerid=201 dtmfmode=rfc2833 mailbox=201 Thanks in advance for any help. Rana Dutt Softel, Inc Marlboro, NJ (732) 810-6707 x200 ___ Asterisk

[Asterisk-Users] Pressing digits on SNOM phone results in letters on display

My SNOM 200 phone got into a funny mode where if I dial any digit, a letter gets displayed and sent, so dialing no longer works. For example, if I dial 9, the letter w gets displayed and sent when I press OK. How do I get it out of this mode? ___

[Asterisk-Users] Call Admission Control

Let's say you have a 256 Kbps Internet connection and you're using it for voice calls. With mu-law (G.711), each call uses about 80 kbps, so you really can't have more than 3 calls active at one time. Does Asterisk support any kind of Call Admission Control where it would prevent you from

[Asterisk-Users] SNOM II and Siptone phone on eBay

Sorry to post this here also, but the biz list doesn't seem to have much traffic yet. I have a brand new SNOM 200 IP phone and also a new Siptone II phone available on eBay, see http://tinyurl.com/2pbng They are surplus after a customer cancelled an order. Please direct all followup questions or

RE: [Asterisk-Users] Lucent Phones

Also, check out www.citel.com This company claims to have SIP adaptors for Avaya's digital PBX phones. If they work as advertised, you can keep your Avaya/Lucent phones, throw out your legacy PBX, and connect them all to Asterisk! However, I doubt they have all the display integration working

[Asterisk-Users] Connecting two branch offices using * and Mediatrix

Suppose a company has a U.S. office and a foreign office, and would like to make toll-free calls using IP between the offices. The U.S office will have an Asterisk system, but the foreign office has a large legacy PBX that they want to keep. One way to do this is to install a

[Asterisk-Users] Intermittent choppy speech using VoicePulse?

Yesterday evening, the speech on all the calls I made using VoicePulse sounded choppy from my side, although the called party said I sounded fine. Also, the voice mail messages I recorded calling in to the VoicePulse number sounded choppy. Calls I made over the PSTN line using my Zap interface

[Asterisk-Users] Cannot use # key to transfer calls

I cannot use the # key to transfer a call. I have two kinds of SIP phones, Grandstream and IpDialog, and the # key cannot be used to transfer on either one. If I press the # key during a call, I hear the touchtone for it, but Asterisk does nothing. The documentation for parking a call says that I

RE: [Asterisk-Users] gs on phone ?

In your extensions.conf, the b and u are reversed. Use u${EXTEN} for priority 2 and b${EXTEN} for priority 102. -Ron -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Clifton Sent: Tuesday, March 02, 2004 10:29 PM To: [EMAIL PROTECTED]

[Asterisk-Users] Dialing out after caller leaves message

I want Asterisk to call my cell phone after someone leaves me a voice mail message. How do I do this? I cannot use Dial after the Voicemail application, e.g., [Step 1] exten = 100, 1, Dial( SIP/100, 15 ) [Step 2] exten = 100, 2, Voicemail( u100 ) [Step 3] exten = 100, 3, Dial(

RE: [Asterisk-Users] Dialing out after caller leaves message

The following suggested sequence does not work: exten = 100, 1, Dial(SIP/100, 15) exten = 100, 2, Voicemail(u100) exten = h,1,Dial(Zap/g1/CELL_PHONE) The Dial command in the 3rd step will fail because the channel it uses to dial out on has hung up already! For example, if extension 101 leaves a

RE: [Asterisk-Users] Message waiting light not coming on

. Iain --On Thursday, February 26, 2004 12:55 am -0500 Rana Dutt [EMAIL PROTECTED] wrote: I cannot get the Message Waiting Light (MWL) on my Grandstream phone to turn on when I leave a new voice mail message for that phone. I have specified the correct mailbox in my sip.conf as follows: Have you

RE: [Asterisk-Users] Conference and transfer

Thanks for the info. Which phones support consultation transfers? The Grandstream and IpDialog phones most certainly do not. Also, I find it disconcerting that there's a Conference button on the Grandstream phone, but when it's pressed, nothing happens. If this sends out some sort of switch-hook

[Asterisk-Users] Message waiting light not coming on

I cannot get the Message Waiting Light (MWL) on my Grandstream phone to turn on when I leave a new voice mail message for that phone. I have specified the correct mailbox in my sip.conf as follows: [200] type=friend username=200 host=dynamic context=dialout callerid=200 dtmfmode=rfc2833

[Asterisk-Users] Could voice mail problem be related to RAM?

I wrote to the list a couple weeks back about my voice mail messages sounding garbled. This happens no matter what phone I use to record the message, Ive tried IpDialog, Grandstream and SJ-Phone. Obviously, no one else is having this problem, since Ive never seen it discussed here. I

[Asterisk-Users] Conference and transfer

Two newbie questions: 1) Does Asterisk support a consulting transfer? E.g., call comes in, Mary answers, Mary presses Transfer and dials Joe, verifies that Joe answers and informs him who is calling, and then presses Transfer to complete the transfer? 2) How does one set up a 3-party conference?

[Asterisk-Users] Voice mail sound distortion has everyone laughing

I wrote earlier about how garbled my voice mail messages sound, so I thought I'd attach an example of a message I recorded earlier today. Just click on the attached WAV file for a good laugh. -Original Message- From: Asterisk PBX [mailto:[EMAIL PROTECTED] Sent: Friday, February 20,

RE: [Asterisk-Users] softphone configs?

What does your sip.conf look like? Please include it in your next message in its entirety. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Wednesday, February 18, 2004 1:08 PM To:

[Asterisk-Users] Slow, distorted speech on voice mail messages

When I play back my voice mail messages, the words sound very s-l-o-w-e-d down and distorted. It's like I'm speaking in slow motion. I've tried recording using an IpDialog SipTone phone, a Grandstream phone and SJ-Phone, and the problem always happens. So it's not the phone. When I talk from one

[Asterisk-Users] Double digits seen using Grandstream phones

My attempts to use voice mail from my Grandstream Budgetone 101 phone always fail because Asterisk is seeing either double digits or dropped digits, no matter what dtmfmode setting I try. Here is what happens for each mode: dtmfmode=info, phone set to send INFO 101: every digit is seen double,

RE: [Asterisk-Users] Re: Double digits seen using Grandstream phones

PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James H. Cloos Jr. Sent: Tuesday, February 17, 2004 11:51 AM To: [EMAIL PROTECTED] Subject:[Asterisk-Users] Re: Double digits seen using Grandstream phones Rana == Rana Dutt [EMAIL PROTECTED] writes: Rana My attempts to use voice mail

RE: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water

I was able to solve the audio quality problem by going to www.grandstream.com/BETATEST and downloading the latest beta firmware, version 1.0.4.46. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Tuesday, February 17, 2004

RE: [Asterisk-Users] Buzzing on Grandstream phones

Try downloading the latest firmware, version 1.0.4.46 from www.grandstream.com/BETATEST I used to have bad audio problems on my Budgetone 101's until I upgraded their firmware. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd Wallace Sent:

RE: [Asterisk-Users] Howto apply a patch, diff file

Type the command patch file.c fix.diff where fix.diff is the patch to apply to file.c. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Larsen Sent: Tuesday, February 17, 2004 9:03 PM To: [EMAIL PROTECTED] Subject:[Asterisk-Users]

[Asterisk-Users] Speech between Grandstream phones sounds like talking under water

When I make a simple phone call from one Budgetone 101 to another, the speech sounds slurred and slow, sort of like the person is talking under water. Both phones and the Asterisk server are on the same subnet. Both phones are configured to use the PCMU (ulaw) codec as first choice, and the Voice