similar to: Please help -- Syntax for dialing VoIP provider

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 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 please excuse if this seems obvious, but I am new to this and the SIP section in the Asterisk handbook do not give any clues nor do the SIP examples in there seem to represent real-world situations. I am using Nikotel as a VoIP provider (for now) and I would like to configure Asterisk to sign on with Nikotel so that I can use the telephones connected to Asterisk to make calls using the

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 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 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

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

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):

Is there a service like iconnect that does allow dialing out more then one concurrent connection? Asterisk works great with iConnectHere, but they only allow one call at a time. I don't want to setup an account for each concurrent call, because it will make iConnect an expensive service, and besides, all of our calls combined doesn't reach 1000 minutes per month. Any ideas?

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

Residential Long Distance. One of my biggest pushes towards a VoIP provider was cheap long distance. Now in the U.S. at least with SBC they now have a plan for Unlimited Long Distance. The price is 30.00 a month if you do not have a couple of required features on the line already like Caller ID and a feature like 3 Way or Call Waiting etc. in which case that lowers the price to 20.00 a month.

When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no ringback tone when making a call. I have tried adding the 'r' command to the dial sting with no help. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have SIP registrations working correctly for FWD and Sipphone, but it is impossible to connect to Sipcall or Nikotel, I saw that someone on the list has problems with ICH. To try and sort out the problem I tried to register to Sipcall with Linphone and sent the dialogs to tech support of the equipment provider. Here is their answer:- The reason the registration fails is because not

Could anyone please help me with that: I have an analog telephone connected to Asterisk using a Sipura 2002 ATA. When calling the extension, the caller ID presented is always the number of that extension rather than the number of the calling one. While I learned that this is the new standard behaviour (?) of Asterisk, I want to show the original caller ID. I tried the options o and f in the

Hi all, Is there a possibility to set the codecs Asterisk will choose in the dialplan ("exten=>" statements or their contexts) instead of sip.conf? My problem is that I connect my SIP phone with several providers (Nikotel, Sipgate, Stanaphone) for icoming and outgoing calls. Not all of these providers offer the same set of codecs. I'd like Asterisk to use the same codec for the

Hi Guys, I have Spikko setup as provider of DID and outbound routes and I can make calls out but no inbound calls via DID can be made. I did a sip debug which is reported below. I never receive the call though, I have a catch all in my inbound routes and it doesn't hit my context at all or not sip invite comes in: FreePBX: Trunk Name: *Spikko* Peer Detail *username=MyUsername*

Hi All, I was wondering, in my effort to show how Asterisk can replace Call Manager, if there is support for call transfers/forwarding from the users Cisco 7940 SIP phone to either another SIP client or through the AS5300 on to the PSTN. I do see some stuff in the docs but seems to be specific to a local PRI board in the PC of which I don't have. Any experiences/comments most appreciated.

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that

John Todd have started creating a document called Readme.channels that will document the syntax of extensions in all channels. I have uploaded his draft to the Wiki, so that all of you can help find the syntax, it's not so easy to grasp from reading the source. It would really be handy to have it all in one place, within the source distribution (and of course in the Wiki...)