similar to: Passing PIN Numbers

I am looking for a good Asterisk GUI to manage my server. Any Suggestions? Regards, Michael DiMartino Director of MIS The telx Group, Inc. 17 State St, 33rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/96bf8fea/attachment.htm

To All; My current issues is a 5 second delay for call that is being transferred from the Norstar units to the Asterisk servers VIA a PRI. Is their anything that can be done to speed up the transfer on the Norstar. Below is my current phone config. < Norstar1 >----PRI----< Asterisk-1 >----IP-WAN----< Asterisk-2 >---PRI---< Norstar2> The Norstars are MICS 0x32 4.1

Hi, I haven't seen any mention of this on the list. I'm curious if anyone has tried it and can share some opinions on it? http://www.thirdlane.com/screenshots.htm http://www.thirdlane.com/opensource.htm#manager Defaults Manager - initial PBX configuration Device Manager - management of devices (phones) Mailbox Manager - configuration of user mailboxes Extensions Manager - dialplan

I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP

I use SIP over VPN with RR from TWC no problem, connect via WiFi. According to http://www.speakeasy.net/speedtest/ I am getting 3.5Mbps down and 353Kbps up at this time (6:15pm Saturday). My laptop currently has an X-Lite (free version) softphone with GN Netcom USB professional contact center headsets (GN8110 USB XP adapter). We have found that the headset makes a major difference in the quality

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following conf files 1. extensions.conf (telx-nyc) 2. iax.conf (telx-nyc) 3.

Hallo all do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel? thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

To All I am using a SNOM 190 w/Asterisk server. Here is my sip.conf [7501] type=friend context=external username=7501 callerid="Telx 7501" <7501> mailbox=7501@telx.com host=dynamic dtmfmode=rfc2833 My question is this. With above settings in my sip.conf specially "dtmfmode=rfc2833" What should my "DTMF Payload Type:" be set to on my SNOM 190 phone.

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

Hii, I want to forward calls from an asterisk server to a local gsm network. I have read the wiki pages on various forums. But the thing i want is to receive the call(Voip) from an asterisk server then it should be forwarded to a gsm network & again to either a gsm/ PSTN from the gsm network itself. Please post a help. Thanx in advance. -- Day by Day in Every Way I'm Getting Better

Does anyone know of any companies where I can interconnect with for SMS? ? .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office

Hello, I'm trying YAACID ( http://www.shatterit.com/opensource/yaacid/ ) for incomming call notification on PC (and open url with callerid), but it does not display/pop anything :-( my config is very simple... (yaacid is successfully registered as manager in asterisk) thanks PJ * dialplan: '953' => 1. NoOp(${CALLERID}) [pbx_config]

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

Hi, I currently use? Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig of memory When i use i4l on any call, the called party ( on the telco operator side ) ear me with a delay of 1 sec after 1 minutes , 2 sec after 3 minutes and so on... After a quart hour, the delay make the conversation just

Hello all, Recently I migrated all telephony in my house to asterisk thanks to the Asterisk, QuadBRI which works wonderfully well. Some small tweaks to make but that's on the long list. On the short list is the ability to reliable send and receive SMS. For SMS I already built a script email2sms, but sometimes the SMS doesn't get send from some reason, the sms log then reports something

For canreinvite=yes to work, I think I need to remove the t argument in the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways stay in the middle. I don't want that, so I removed the 't' argument. That works. Now, when two UA are calling, Asterisk gets out of the RTP stream. However, when removing the 't' argument, the Music On Hold doesn't work anymore

HooDaHek 0.6 has been released. So soon, you say? Well, the best laid plans of mice and men... Steven BerkHolz is a pretty sharp stick and said to me, "Why don't you have HooDaHek change the CallerID when it looks up the name in the database on an incoming call?" Much head smacketh ensued, and as I made that change for Steven, I noticed that I had the way wrong version of

Hello, i wonder why i didn't find a solution for this problem yet, because it seems very common: I have an asterisk server with an AVM (Fritz) ISDN-Card (BRI), and some SIP-Softphones which i can call from outside by calling the phonenumber of the Asterisk-Server and then dialing the number of the SIP-Phone. If I make a call from a SIP-Phone into PSTN, only the MSN of the asterisk-server is

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can