search for: 221a

hello everybody, I'm planning a quick upgrade from 2.2.1a to 2.2.2, and I want to know which files from $PRIVATEDIR (and others) I need to keep... I have it like this: -rw-r--r-- 1 root root 42 Jul 28 16:40 MACHINE.SID -rw------- 1 root root 8192 Nov 2 12:55 secrets.tdb -rw------- 1 root root 10274 Nov 2 12:56 smbpasswd I'm sure

Hey geeks, I have been reading several faqs and howtos now and haven't come very far with what I really wanted to accomplish: I have installed samba from the RH 7.2 distro rpm. I have set it up so I can access my windows shares on the linux mashine. I can "ping athlon" (which is the netbios-name for the win2k mashine) I can "ping linux" (which is obviously my

...tivity 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 Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

...eave a message and is cut-off after about 10 seconds. In an ngrep trace we see Asterisk receive an ACK from the SBC and it immediately responds with a BYE message for that call. Has anyone else experienced this type of issue? --- ISC Networking & Telecommunications 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 215-573-8396 215-898-9348 (fax) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100825/b3c86e07/attachment.htm

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:

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

...he Octel. The problem is I still need an SMDI channel that is integrated with the message processing part of the gateway. Has anyone look worked or developed any working models that might help? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

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

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

...Comedian mail and the spool directory where it looks for managing messages. Has anyone implemented voicemail in this way? If so could you share some thoughts on how the integration is accomplished? Thanks -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

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

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

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

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

...gt; <p class=3DMsoNormal>---<o:p></o:p></p> <p class=3DMsoNormal><o:p>&nbsp;</o:p></p> <p class=3DMsoNormal>ISC Networking &amp; Telecommunications<o:p></o:p></p> <p class=3DMsoNormal>3401 Walnut Street, Suite 221A<o:p></o:p></p> <p class=3DMsoNormal>Philadelphia, PA 19104<o:p></o:p></p> <p class=3DMsoNormal>215-573-8396<o:p></o:p></p> <p class=3DMsoNormal>215-898-9348 (fax)<o:p></o:p></p> <p class=3DMsoNormal>...

It seems on my phone, which is hooked up to a large pbx network powered by an asterisk server, that it will randomly start ringing with a callerid# of 2013 which is its username for that phone. I have looked and been watching on the asterisk command line with the -vvvvvvr switch and nothing has been seen that indicates a reason for this random ringing. This leads me to think that this trouble

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

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:

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

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.