similar to: announcing inbound PSTN calls

Displaying 20 results from an estimated 3000 matches similar to: "announcing inbound PSTN calls"

2006 Oct 29
4
blind transfers with IP Polycom 501
I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer => # This works for all my
2006 Nov 15
2
Grandstream GXP2000 -- What's the Catch?
We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great
2007 Mar 18
6
T1 cable for Digium T1/E1 Cards
Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance
2008 Jan 10
8
IEEE 802.1x capable sip phones
Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. Thanks in advance. ====================== Jeronimo Romero EUS Networks Email: jromero at euscorp.com Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com ====================== -------------- next part -------------- An HTML
2006 Nov 21
2
FW: CISCO 7960G & Asterisk
I was wondering if people have experienced issues with Cisco 7960G and Asterisk. Any feedback on people's experience deploying this phone in production environments would be appreciated. Thanks in advance
2007 Feb 27
5
TE110P: Error ==> Asterisk died with code 1.
Running Asterisk 1.2.9. I just installed a TE110P card and configured zaptel.conf & zapata.conf. The config files look right to me but I'm getting the following error when trying to start asterisk: Asterisk died with code 1. Automatically restarting Asterisk. Does anyone have any idea what is wrong with this configuration?? Thanks in advance!!! Here's my config files: zaptel.conf
2006 Dec 08
1
cal recording with email
I'm trying to set on-demand call recording. Here's a snippet of the pertinent dialplan. The purpose of this is to allow one user in particular to be able to receive an email recording of the call everytime he dials *91 + number. The problem is that the email is not going out or being generated when I use the ${CALLFILENAME} variable. When I use the actual file name of the gsm recording,
2006 Oct 30
1
dealing with blind transfers to invalid extensions
Running Asterisk 1.2.8 kernel 2.6.13.4-1. Everything in my dialplan seems to be working well except for one problem. When calls are blind transferred to an invalid extension I would like the call to go to the operator on ext 1000? What is the best way to do this? Thanks in advance Here's a snippet of my extensions.conf [default] exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
2007 Jan 03
3
caller id ring tones for Asterisk Phone
I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Thanks in advance!!!
2006 Dec 11
2
asterisk PLAR
Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown). The Oreilly (Asterisk: Future of Telephony) book mentions it in passing saying that all you need to enable it is to set immediate=yes in zapata.conf. Has anyone implemented this in brokerage trading environments? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Feb 05
2
asterisk server as a voicemail server for legacy PBX -- FXO or FXS???
Hey All, I'll be configuring an asterisk box to be the voicemail server to an old Merlin system which had an octel 100 voicemail server that is now dying. My question is simple: do I need to stick an FXO card in the asterisk box? My logic is that if the Merlin Magix system is actually generating electrical current, then I would need to have an fxo card. Is this correct?
2006 Nov 12
2
same extension on softphones and hardphones
Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the
2007 Feb 06
1
asterisk server as a voicemail server forlegacyPBX -- FXO or FXS???
Thanks. Is there a way I can log into the Merlin Magix to determine that? How else do I tell? ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Germann Sent: Monday, February 05, 2007 8:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users]
2007 Feb 01
8
Dell Servers
Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the
2006 Nov 13
6
Dual Wan Router with Failover
Hi List, Does anyone know of a good dual wan router that can handle SIP well and can failover between connections if there is a SIP issue on one of the lines (meaning there still is a connection however there isnt enough bandwith or sip packets arent going thru etc.) ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Apr 19
1
aastra phones with asterisk 1.2.17 - hangup after 20 seconds
Running asterisk 1.2.7 with latest zaptel on centos4.4. with Aastra 55i phones. Local outbound calling works fine, but ATT requires clients enter 7 digit code for long distance. All calls with 7 digit code are lost within 20 seconds of the call. This is the message I?m getting: Apr 19 12:38:16 WARNING[9615]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission
2006 Nov 27
1
calls hang up even after Background() message eventhough response timeout is set to 10 sec
I'm experiencing a strange problem. My inbound calls are hanging up right after Background() message even though response timeout is set to 10 sec. [voicepulseincoming] exten=>_X.,1,Answer exte=>_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(9:00-15:00|fri|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(*|*|*|*?after-business-hours,s,1)
2005 Sep 12
4
is this the correct way to use the mailing list?
don't read this. i'm a complete newbie, and i don't know how to start sending messages. if someone read this, then can answer me also how to delete a file with the console (really ashamed) y know that copy with cp, but delete??? thanks... ___________________________________________________________ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del
2011 Oct 25
1
Centos6 sealert browser doesnt appears
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, Im trying to get the sealert browser to show up on my desktop, but I cant get it to work. I have installed all setroubleshoot packages, which provides sealert and im running sealert -b from the command line over a GUI session on gnome and nothing happens. Any ideas? Jeronimo Calvo jeronimocalvop at hush.com -----BEGIN PGP
2008 Apr 11
1
Speex
<speex-dev at xiph.org> Hi all, I'm a begginer with DSP and i need your help and suggestions. I'm trying to use the Speex and a DSP DM642 to implement a solution with voice! I'm using the 1.2 beta 3 distribution and the TI's Code Composer Studio v3.1 simulator. I coosed speex_C64_test.pjt and modified the speex_C64_test.cmd to only use the DM642 external RAM memory. I'm