similar to: CISCO 7905 Phone Weirdness

Displaying 20 results from an estimated 3000 matches similar to: "CISCO 7905 Phone Weirdness"

2005 Mar 16
3
Cisco gateways and hairpinning
Hello: Has anyone on this list had to configure hairpinning on a Cisco gateway running IOS 12.2 or 12.3 and using a PRI for connectivity to the PSTN? If so could you tell me how it is done? I'm told this is the source of my call transfer problems and yet I cannot find clear instructions for how the configuration is done. Thanks,Steve -- ISC Network Engineering The University of
2005 Feb 08
3
Looking for FXS device - CISCO ATA 186
I was looking for something to connect a couple of POTS handsets to my asterisk server and found this on ebay http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&category=162&item=5162868118 &rd=1 The documentation says that it does SIP - therefore will it work in an asterisk environment. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus
2006 Mar 14
2
OT - force Cisco phones to reboot
Hello to all Does someone knows how to force the Cisco IP phones (7940 and 7960) to reboot weekly or monthly? I think this would be useful because sometimes we change the configuration settings in the TFTP, but the phone just check the TFTP when he restarts... Thanks Jo?o
2009 Jul 16
1
H323 situation
Hi all, I have this installation: Asterisk 1.6.1.1 with h323 support, pwlib_v1_10_3 and openh323_v1_18_0. I have a problem that is, when a call comes from H323 and goes to a Sip phone the asterisk sends two rtp streams to the sip. I checked this with tcpdump, save the payload (voice is in G711u), one is the ringing indication and the other is the voice coming from the user in h323 side. And
2004 Sep 11
1
call park question
I can part a call (dial #700 it is parked on 701) but if I dial 701 I am told it is not a valid extension? I have include => parkedcalls in my local extension context. I have Ttr on all extensions and the incoming pots line. It parks, plays MOH but I can't retrieve it. --john -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Jan 28
1
Command to light MWI on 7940 /7960
We have several agents on queues, and want to indicate to them that they are logged in or logged out. We have tried several different ways, from changing the screen to presenting different service menus, but cannot get anything to be "in their face" (their words, not mine). One of our team has suggested, as the agents do not have voicemail, is to use the MWI on the 7940 phones to
2005 Feb 14
1
Uptime/reliability with SER, Asterisk
Could anyone shed any light on how SER and/or Asterisk (stable branch) has held up for them in that last while? Are you using SER and/or * in a production environment? Do you ever restart the software or reboot the system? How many users are utilizing the system? How many calls per day/concurrently? I read some uptimes and such on the mailing list from long ago, so I was wondering what some more
2005 Sep 12
1
Other Voicemail systems
Since I can't seem to get anything figured out for the Comedian system, are there any other systems out there that we can hook asterisk into? Sherwood McGowan ViaTalk Level 2 Support VOIP System Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/32ae9d25/attachment.htm
2006 Jan 06
1
server recommendations
OK all. I need some help. Looking to deploy asterisk servers and want to get a recommendation on what server to buy. I love Dell's, but from what I see on the list they seem to have some issues. I would like to stay with one brand and need systems for small offices (20 users), medium (50 users) and large (100 users) systems. Thanks for the help. Keith
2014 Nov 25
0
asterisk prompt
Hello, I am trying to set up color prompt. In the documentation I have found this: %Cn[;n] Change terminal foreground (and optional background) color to specified. A full list of colors may be found in include/asterisk/term.h* But nowhere could I find what format the color code should be. I have tried all possible permutations, none of them works: ASTERISK_PROMPT="%Cn[COLOR_BLUE]
2005 Feb 10
1
SER Asterisk Voicemail
Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java
2019 Oct 23
2
issues with Asterisk CLI
Hello, I have Asterisk 16.2 on Debian. In the Asterisk CLI, I would like to change 2 things: 1) change the keybindings for commandline editing (what in bash is called "readline" editing of the command line) The CLI is missing some very useful keybindings, and even worse, has misconfigured others, For instance, "ctrl"+"w" should delete the last word backward. But
2017 Jun 06
4
asterisk server - no sound
Thanks Anthony. I did it on the server, according to https://www.voip-info.org/wiki/view/port+forwarding However after doing it, when running Asterisk I get the following message sudo asterisk -vvvvvvr No ethernet interface found for seeding global EID. You will have to set it manually. Unable to access the running directory (No such file or directory). Changing to '/' for compatibility.
2005 Jun 16
3
SER and Asterisk question
Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and
2005 Feb 14
2
FW: SER Asterisk Voicemail
Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW:
2005 Feb 04
4
HP ProLiant server for Asterisk
I'm looking at ordering a server from HP. I checked around on Google and found in the Wiki that the ProLiant DL380 is supposed to be known to work with *. I'm going to get a price quote on the following setup: HP ProLiant DL380 G4 Server w/ the following options: Intel Xeon 3.20GHz/1MB 2GB REG PC2-3200 (2 X 1GB) HP ProLiant Battery Backed Write Cache Enabler for SA6i RAID 1 drive set HP
2005 Feb 16
4
DTMF inband detection improvement
Hi all, I have some probleem detecting DTMF send by a GSM phone, I'm using SIP with ulaw. do you know what are the options to improve the detection ? I'm using asterisk 1.05, is the CVS HEAD version had some improvement about DTMF detection? Florian.
2005 Feb 03
5
Cisco 7960G phone crashes during SIP upgrade
Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image
2010 Aug 25
1
Asterisk 1.6.1.17 ACK/BYE question
We're running Asterisk 1.6.1.17 for our campus voicemail server and Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are diverted to voicemail using a 302 redirect when the called party doesn't answer. In this case the caller is able to hear the greetings and begin to leave a message only to have Asterisk terminate the call mid-recording. We're uncertain why
2006 Jan 06
5
3RD REQUEST - Any Help Is Appreciated
Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has