similar to: How do I make Asterisk login at/use VoIP provider?

Hi thanks to everybody who has been assisting me in solving the various problems I had to dial out from Asterisk to a PSTN number with SIP using Nikotel's VoIP service. I have drafted a mini-how-to which is available at http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf This is a first draft, I will amend this further, in particular the "verify and debug" section

Hi thanks to everybody who responded to my earlier post. I have looked at all the material and links provided and tried everything in there, but it simply won't work for me. My SIP phones register with Asterisk, but they cannot be called (everybody is busy at this time) nor can they call anything (error code 4, whatever that means) not even internal (yes I did give them appropriate

Hi list, I have some nontrivial questions. I am no telecommunication guru and I will explain it with my simple words. I hope someone can help me with these issues (with Asterisk 1.0.3): - If I call outside (with Nikotel to German Telekom) there is a remote hangup after 2 minutes. I've seen other people posting this but nothing helped. I luckily managed to get around this issue with the

Hello list! I'm using "Asterisk CVS-11/22/03-04:28:51" and try to route my normal (classic) phone calls via nikotel (www.nikotel.com). I can talk about 1 minute and get then disconnected. Here my current configuration parts which affect nikotel: register => chabrol:PASSWORD_REMOVED@nikotel/500 [nikotel] type=friend secret=PASSWORD_REMOVED username=chabrol fromuser=chabrol

Hi A few days ago, Kelly remarked that he had previously observed that Nikotel charged him for calls he did not actually complete. I have made a number of test calls to my landline without picking up the calls. I just let it ring once and hung up on the calling phone. A look at the call records on MyNikotel reveals that I was charged six seconds for every of these calls. I have raised a

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

Hi! I've read in the archives that there are problems concerning Nikotel calls being disconnected after two minutes. I had the same problem yesterday. Is there a fix? There was only a "giving up" statement after the last e-mail in the archive, I'm about to do that too. Here's my sip.conf entry for Nikotel (left out the register stuff 'cause it's working):

Hi I have just been struggling for four days to get SIP working and now as I created a voicemail box, Asterisk has become very unstable and it can't bridge SIP phone to SIP provider calls anymore. Calling internally from one SIP phone to another works fine. Calling internally from a SIP phone to an analog phone on a Zap channel and vice versa works fine. Incoming PSTN calls delivered to

I'm not sure I'm barking up the right tree here, but would I need to make use of groups to plot two separate datasets within ONE panel in xyplot? The desired end result is a single xy plot of two separate (but similar in values and ranges). Full code follows, xyplot code at bottom #########Determine Frequencies ##########coastal_slope #needs the maptools package to read ESRI grid

Hi folks, this does look like a bug to me: Asterisk replaces the @63.214.186.6 by @context which obviously leads to a failure. Any comments, do I have a configuration issue on my side that I missed? Cheers, Philipp -- Executing Dial("SIP/philipp-bd5f", "SIP/992365264680@nikotel- out|90") in new stack -- Called 99xxxxxxxxxx@nikotel-out -- Got SIP response 302

Folks, I do have a couple of problems I hope you could help to solve. I have searched the net, but had little success of finding documentation about samba 3.2 and AD. I do have Win server 2003 domain and AD in native mode. The new samba server should be a member server in the domain. The idea is to have same UIDs as the RIDs in the AD. The enviroment is as follows. Sernet packaged samba 3.2.3-37

Hello ! In order to chase after a problem I implemented the following dialplan to have an answertime of exactly one minute: exten => xxxxxxxxxxx,1,NoOp(Test wait) exten => xxxxxxxxxxx,n,Answer exten => xxxxxxxxxxx,n,NoOp(Current timestamp: ${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}) exten => xxxxxxxxxxx,n,Wait(60) exten => xxxxxxxxxxx,n,NoOp(Current timestamp:

Return-path: <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx> Envelope-to: xxxxx at xxxxxxxxx Delivery-date: xxx, xx xxx xxxx xx:xx:xx +xxxx Received: xxxx [xxx.x.x.x] (xxxx=xxxxxxxxx) xx xxxxxxxxx.xxxxxxxxxxxx.xx xxxx xxxxx (xxxx x.xx) (xxxxxxxx-xxxx <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx>) xx xxxxxx-xxxxxx-xx xxx xxxxx

Hi Samba, I hope you are doing well. I run a cifs / nfs CTDB clustered NAS solution, and I find that when I reboot any of the nodes in the cluster, I must re-export the nfs mounts so they show up properly. Perhaps this is a general linux nfs bug and I am barking up the wrong tree, but I haven't found any problem / solution mentioning this as of yet besides my own known workaround

hello, i have a FreeBSD 4.8-PRERELEASE #0 that i use as a gateway / nat box for my home. It also acts as a dns / mail server to the outside world. I'm using ipf and basically filter for bogus networks on the way in and out. I allow everything out keeping state, and allow this in: pass in proto icmp from any to any icmp-type squench group 200 pass in proto icmp from any to any icmp-type timex

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

I just discovered two directories with the same number of files and the same number of hard links but different size: # stat /home/david/linuxburg/fax.old/docq_ps.nnnn \ /hdsync/home/david/linuxburg/fax.old/docq_ps.nnnn File: `/home/david/linuxburg/fax.old/docq_ps.nnnn' Size: 8192 Blocks: 16 IO Block: 4096 directory Device: 801h/2049d Inode: 52060

GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can

I was seeing rsync errors which caused my system installations to fail repeatedly and absolutely. See the error I reported to this mailing list back in June in the Forwarded Message below. When I replaced my NIC cards (realtec chip, 8139too driver) with Linksys cards (and Dec Tulip driver for linux) the problem ceased, rsync worked like it should, and does so repeatedly. My suggestion would be

mbox messages gets header corruption caused by an extra linefeed after Content-Length Users sees their mails in Sent mbox folder without the from and to fields, without attachments and with the date of 1/1/1970 Diego. --- Here is an anonymized header: >From xxxxxxxx at xxxxxx.xxxxxx.xxxxx.xx.xx Tue Jun 03 09:14:33 2008 Message-ID: <xxxxxxxx.xxxxxxx at xxxxxx.xxxxx.xx.xx> X-UID: 3913