RE: [Asterisk-Users] SIP reload configuration problem /* New subject */
New users arent being added from the sip.conf file... let me play with it and get back to you in a few weeks when I know what I am doing! Newbie remember! Dont want to send you down the wrong track and then work out that its me doing something wrong. Thanks for your help Cheers Chris -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: Friday, 9 January 2004 7:24 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP reload configuration problem /* New subject */ Christopher Raper wrote: > When creating users in the sip.conf file, they do not appear when running the "sip > show users" > command from the CLI until i restart. A reload doesnt make them appear. The appear for me after a reload. (running the latest and most dangerous CVS, many patches are added by malcolm - thank you!) While on the topic, note that some peer data based on their registration status is saved in the Asterisk database, and when doing a reload, peers already logged in will stay logged in. This is a feature, not a bug :-) Other than that, the SIP channel should be reset and new users added. Maybe there's something else that is causing you trouble with your users? /Olle ___ 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/listinfo/asterisk-users ___ 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/listinfo/asterisk-users
RE: [Asterisk-Users] Mailing list growth
Couldnt agree more with the online docos wish I could help, maybe soon! Certainly would reduce the number of basic config questions (coming from myself - I would rather read docos than annoy people!) >From sunny Oz. Chris -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: Thursday, 8 January 2004 9:25 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Mailing list growth Well, mailing list growth is not only a good thing. It's getting almost impossible to handle. As I've stated before, we need to change Asterisk.org so we can help people in a better way and avoid a lot of the repeating questions on the mailing list. There's a lot of people unsubscribing, just because of the amount of messages. Asterisk.org needs an FAQ, more documentation on line and ... I've offered to help in this, with no reply so far. I think it's getting urgent. And this is the end of yet another message from a dark and snowy Sweden ;-) /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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/listinfo/asterisk-users
RE: [Asterisk-Users] SIP reload configuration problem /* New subject */
When creating users in the sip.conf file, they do not appear when running the "sip show users" command from the CLI until i restart. A reload doesnt make them appear. As i said, I am new to the whole Asterisk thing, however have worked with IP/SIP PBX's for a few years - its most likely a user problem though! Check it out and let me know what you get. Cheers Chris PS - I would try and look at the code, but I know f^ck all about C programming, I'm a Shell man myself! -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: Thursday, 8 January 2004 6:42 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP reload configuration problem /* New subject */ Christopher Raper wrote: > I have also noticed that sip.conf doesnt get updated without a restart. > was thinking I am doing something wrong, but maybe not now.. > sip.conf doesn't get updated, but the SIP configuration is updated on a reload. Could you please describe your problem with the SIP reload a bit more detailed, so I can see if is a code, docs or user problem :-) /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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/listinfo/asterisk-users
RE: [Asterisk-Users] Re: 911 and lawsuits and redundancy
I have also noticed that sip.conf doesnt get updated without a restart. was thinking I am doing something wrong, but maybe not now.. Chris -Original Message- From: Jonathan Moore [mailto:[EMAIL PROTECTED] Sent: Thursday, 8 January 2004 8:42 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Another concern I have on this front is that it seems like some updates require an asterisk restart rather than just issueing a reload command from the * console. This that correct, or I am just not running the system correctly? For instance it seems like I couldn't get zapata.conf changes to go into effect without closing * and starting it again. It also seems like some changes in sip.conf don't go into effect either. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Cees de Groot <[EMAIL PROTECTED]>: > Jonathan Moore <[EMAIL PROTECTED]> said: > >Any ideas? > > > Wait until 2am :-). > > And of course, with well-managed security procedures around the system, > you probably will not be upgrading kernels with every bug. The last > couple of holes in the Linux kernel are locally exploitable only, and > with a * box behind a firewall that only transmits VoIP data and has > good physical security, I wouldn't rush out to upgrade and reboot the > pbx on the first bugtraq post. > > > -- > Cees de Groot http://www.tric.nl <[EMAIL PROTECTED]> > tric, the new way helpdesk/ticketing software, VoIP/CTI, > web applications, custom development > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie Question-Looking for Feedback
Greetings all. I am new to the Asterisk world! Found it very impressive so far! In relation to the below. I have worked with Alcatel PBX's for the last 3 years. Alcatel OxE supports SIP and H323 as well. As far as SIP goes I have also found the Xlite to be good for soft phones. I am using one now. Check out www.xten.com Xlite is free and easy to use. I also have been given a Pingtel SIP to play with. http://www.pingtel.com/ As far as H323 terminals go I have not played with all that many, however the simple Microsoft netmeeting works for testing purpose anyway. Now a question to all you experts out there, and this may seem VERY stupid, but I have configured the sip phone and have it logged in and can dial 500 to get to the sample messages etc. However i cannot work out how to give the sip termainal a number that can be dialled. I would assume that it needs to be in the dialplan, so I have added it in via the extensions.conf file, however I am sure that I have stuffed the config somewhere. Can someone please point me in the right direction. Would be much appreciated. Also, do i need hardware to make a SIP to SIP call... eg. Compressors etc. Cheers Chris -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: Thursday, 8 January 2004 8:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie Question-Looking for Feedback M. Matt Colgin wrote: >I've been looking at Asterisk for a replacement for our phone system and I'm >hoping someone can help validate my assumptions. > I'll try.. :) > >We have 4 analog lines coming into the building. These lines are simple POT >lines and we have them in a "hunt group" with Verizon so that when a single >phone number is dialed, the first line is rang, if that line is busy it will >ring the second line, and so on. > >I would like to put together an Asterisk system to handle these lines and >allow us to do VoIP, call queuing, voice menuing, etc. In looking at the >product offerings of Digium, it appears that I need 4 Wildcard X100P's and 1 >Wildcard TDM400P 4-Port. For VoIP work, I'm looking for any recommendations >that can be made. My first priority is to support a user in New Zealand >talking to our phone system in the US, but there could be another 2 I'd like >to support in the US (all on cable modems with the typical capped 30KB/s >upload). I'd like for it to work very well with the Asterisk PBX and be as >simple as plugging in a ethernet cable or even support 802.11b/g with little >to no configuration. > >In addition, I'm curious on other people's experience with software based >VoIP phone. Specifically, it appears that a good amount of amount could be >saved by using software based phones inside the building, thereby negating >the need to purchase 3 hard VoIP phones and the Wildcard TDM400P. Can anyone >recommend a good software package, that is fairly idiot proof and would work >well for a small call-center with temp/minwage employees? > > >To Summarize: >- Can and does it make sense to purchase 4 Wildcard X100P's? >- Can and does it make sense to purchase 1 Wildcard TDM400P (4-Port)? > I will answer these together, the recomendation is typically not to go above 3 cards in a system which means that you could give 5 cards a go but chances are you are not going to have a happy time with it.. My suggestion would be to either use a channelbank and a T100P or the simpler solution convert your 4 analog lines to 2 ISDB BRI lines and then get a 2 port AVM or Eicon ISDN card.. >- What VoIP hard phone works best with Asterisk? Are there WiFi ones that >are less than $100? > My personal favorite in terms of both cost and performance would have to be the Snom 200.. Other options are the Grandstream (cheapest there is), the Cisco(a little pricey), the Snom 105 and no doubt a few others.. A Grandstream costs about $75 and AFAIK its still the cheapest so I would have to say No, you will not likely get a WiFi VoIP phone for under $100.. >- How much bandwidth does VoIP require? Will cable modem users with a max >30KB/s upload ok? > The bandwidth requirement is dependent on the codec but 30KB/s should hande any codek no problem.. the bigger problem you will have between NZ and the US is latency which is really annoying when trying to transfer realtime data.. >- What VoIP soft phone works best with Asterisk? > I have found X-Lite or X-Pro to be the best.. > >Also: >- What kind of uptime are people experiencing? > I have over 100 days continuous, with reboots to apply patches.. Others on the list have said thay have over a years uptime.. >- How much system load will be needed for 4 concurrent VoIP conversations? > I have a P2 400 development server and have done 4 concurrent ( thats all I have ) VoIP sessions.. My production server is more powerful and i have not really looked at the number of concurrent sessions but its never really broken a sweat.. >- What kind of gotcha's have people had that would be good for a newb
