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...wn issues we are having with our system. Mainly dropouts and dropped calls. If you have experience in troubleshooting these issues, please contact me at email attached to this messages. Regards, Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com -- <http://www.rimmkaufman.com> <http://twitter.com/rimmkaufman> <http://www.linkedin.com/company/85385> <http://plus.google.com/104980442218952272663/posts> <http://www.facebook.com/rimmkaufman> <http://www.RKGblog.com> -------------- next part ----...

...oking for sip phones that support something other than openvpn. There are a lot of vendors in China (mainly Alibaba) that sell l2tp VPN phones. Are there any American vendors that support l2tp? Thanks, -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com -- <http://www.rimmkaufman.com> <http://twitter.com/rimmkaufman> <http://www.linkedin.com/company/85385> <http://plus.google.com/104980442218952272663/posts> <http://www.facebook.com/rimmkaufman> <http://www.RKGblog.com> -------------- next part ----...

...low=gsm allow = ulaw allow = alaw allow = g722 dtmfmode=rfc2833 ;; allows use of pushbuttoms ;dtmfmode = inband nat = no localnet = 10.0.0.0/255.0.0.0 canreinvite = no Thanks for any help. Best Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com -- <http://www.rimmkaufman.com> <http://twitter.com/rimmkaufman> <http://www.linkedin.com/company/85385> <http://plus.google.com/104980442218952272663/posts> <http://www.facebook.com/rimmkaufman> <http://www.RKGblog.com> -------------- next part ----...

...xxx (ulaw) No Tx: ACK 240 xxxxxxxxxx xx 0035bf5711b0186 (ulaw) No Rx: ACK ia.ntelos. etc. Thanks, Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com -- <http://www.rimmkaufman.com> <http://twitter.com/rimmkaufman> <http://www.linkedin.com/company/85385> <http://plus.google.com/104980442218952272663/posts> <http://www.facebook.com/rimmkaufman> <http://www.RKGblog.com> -------------- next part ----...

...above, and finally abandoned Asterisk for a commercial system? We have 167 users. I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the conference rooms. Suggestions welcome. Best Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com -- <http://www.rimmkaufman.com> <http://twitter.com/rimmkaufman> <http://www.linkedin.com/company/85385> <http://plus.google.com/104980442218952272663/posts> <http://www.facebook.com/rimmkaufman> <http://www.RKGblog.com> -------------- next part ----...

Hi All, Have completely moved off the old ESI system, and things have been going pretty good with the new server. I have one issue, which has been reported by several of our customers. I've tested it, and it does indeed seem to be a problem. When the customer is asked to dial in the first three letters of the person they are trying to reach, they will be routed to the wrong extension.

All: Yesterday I discovered something interesting. I dialed 1800ANCESTRY from the asterisk system I am testing and got the number doesn't exist message. I then dialed the same number from our old system and it went through. I realized that the "Y" in ancestry made the number too long, and went back to my dialplan. How do I ignore numbers that are too long? Obviously,

All: My company has an existing ESI IVX E-class system with 45 phones. I can add one more card, to expand it another 6 phones, but it's $8000, and then the system will have to be replaced. I have the Asterisk server up and running, with 2 sip lines from the local phone service. (Thanks to you guys, it is working great!). I'm pretty sure this is the way the company will move, and

If you do not put a context in the beginning of the sip.conf file, the default is, ta da, default in extensions.conf. Putting a context=testof idea in sip.conf got things moving: sip.conf [general] port=5060 bindaddr=0.0.0.0 ;10.8.0.34 *context=testofidea* srvlookup=yes disallow=all ;read somewhere you have to disallow everything first allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833 ;;

All, Thanks for the suggestions, but the system is a plan non-sip, non-ip, non pri setup. It's pretty much a closed box setup. And the prices for the card and support are robbery - which is why we aren't going to go with another setup like that. While it has been reliable - I don't think there has ever been an issue with it, expansion is expensive. The local company was

All: Still experimenting with the asterisk server for the company. My local phone company has given me two sip numbers to experiment with, say 444-456-1234 & 444-456-5678 Calling in and out works, and I've spread a couple of the phones out with some co-workers. My question is this: Do I have to define multiple sip lines in either the sip.conf or the extensions.conf? Now when I

All: I am using the standard voicemail in asterisk. Everything works well, except, if a users wants to record their own personal greeting, it doesn't playback. I can see the soundfile being created. I suspect it is a setting in the voicemail.conf, or an option I am over-looking on the grandstream, but if anyone can point me in the write direction, I would certainly appreciate the help.

All: Starting switching over my phone lines. I got phone line 1 switched. Everyone working. I switched the second phone line, and it worked about an hour, then I started getting errors from the cli saying the server could not register with the providing. I restarted the system, and it worked ok for about 30 minutes, and then started giving he same errors. The error is [Jul 13 11:21:14]

All: Some of the people who dial into to our system will press the pound key when entering an extension for the directory key. When waitexten gets that, I get an error messages as, for example 123# doesn't match any extension. I was going to use ${EXTEN} to just use the first three numbers, but I'm not sure how to use this with WaitExten. so I have exten =>

All: Getting ready to put the system in production. Any suggestions on "stress testing" the system? I'd like to initiate say 10 sip phone calls to make sure the provider has the bandwidth. Can you do that in CLI? I've called 4 numbers simultaneously with the hard phones I currently have and am thinking of adding 6 or so soft-phones to various pc's to make a total of

All: I am looking to install another asterisk server in an office located in a different part of the country. I think I can configure the sip and extension conf files, so that the internal phones at the two locations can call each other. My question is this, how do I properly configure the sip file for a different provider at the new location? Can I use a different register statement for

All: I've starting building an asterisk system for our company, which has about 60 users. I am new to asterisk, so thank you for your patience. I've stripped the sip.conf and the extensions.conf down to the bare minimum: Here is my extensions.conf file [globals] [general] autofallthrough=no [default] [fromprovider] exten => YYYYYYYYYY,1,Dial(SIP/151,20) [phones] exten =>

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus