search for: nikotel

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 host=calamar0.nikotel.com qualify...

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 following workaround: The provider section should only contain disallow=all and then only allow=gsm. If I add allow=alaw in th...

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 Nikotel service. Checking the preferences in Nikotel's softphone to get a clue for what the settings are, here is what I...

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....

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 which is blank right now. here is a plain text summary:...

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...

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 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 "Moved Temporarily "new sip:joesmith@63.214.186.6"" back from 63.214.186.6 -- Now forwarding SIP/philipp-bd5f to 'joesmith@from-sip' (thanks to SIP/nikotel-out-c286) May...

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.

...username 120 Registered sip debug also confirms successful registration. I wonder what the syntax is to dial a number via a VoIP provider. This appears to be documented NOWHERE. I tried this: ; International long distance through VoIP service ; exten => _00N.,1,Dial,SIP/${EXTEN:2}@calamar0.nikotel.com,tr exten => _00N.,2,Congestion and sip debug tells me that the account doesn't match the one on record, whatever that means. I tried this: ; International long distance through VoIP service ; exten => _00N.,1,Dial,SIP/myusername@calamar0.nikotel.com/${EXTEN:2},tr exten => _00N....

...ernally from a SIP phone to an analog phone on a Zap channel and vice versa works fine. Incoming PSTN calls delivered to a SIP phone also works fine. Dialing out from an analog phone on a Zap channel using a SIP provider works fine as well. HOWEVER, when dialing out using a SIP provider (both Nikotel and iConnect) Asterisk cannot bridge the two legs of the call and all I get is silence. here is what the console shows: -- Executing Dial("SIP/Sip1-1862", "SIP/442071231234@nikotel|60|r") in new stack -- Called 442071231234@nikotel -- SIP/nikotel-4815 is ringi...

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:...

...300 ;trustrpid = no ;progressinband=no useragent=Asterisk nat=no externip=XXX.XXX.XXX.XXX localnet=192.168.2.0/255.255.255.0 promiscredir = no register => 7304502:my_sipgate_pass@sipgate.de/7304502 register => 31557110304:my_budgetphone_pass@budgetphone.nl/557110304 register => mvanbaak:my_nikotel_pass@calamar0.nikotel.com [7304502] type=friend context=from-sipgate host=sipgate.de username=7304502 secret=my_sipgate_pass nat=yes canreinvite=no insecure=very [31557110304] type=friend context=from-budgetphone host=sip.budgetphone.nl username=31557110304 secret=my_budgetphone_pass qualify=yes...

Hello all. Has anyone managed to get SIPCALL.co.uk's service working with the [*] box? I've managed to register with other SIP providers but not SIPcall. The debug just show's [*] attempting to register. But receiving a 401 error everytime. Cheers Matt

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 all of the...

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?

...n => _0.,1,Dial(CAPI/@XXXXXX:${EXTEN:1},60,r) > exten => _0.,2,Congestion > exten => _0.,3,Hangup() > > exten => _90.,1,Dial(SIP/${EXTEN:2}@sipgate.de,60,Ttr) > exten => _90.,2,Congestion > exten => _90.,3,Hangup() > > exten => _91.,1,Dial(SIP/${EXTEN:2}@nikotel.com,60,Ttr) > exten => _91.,2,Congestion > exten => _91.,3,Hangup() > > ISDN/CAPI, sipgate.de and nikotel.com. I have seperate contexts for > each incoming service. > > I would like to add the prefix for dialling out to the callerid of > incoming calls, so I can use...

...standard behaviour (?) of Asterisk, I want to show the original caller ID. I tried the options o and f in the dial command - e.g. exten => 1002,4,dial(sip/2999,20,o) no avail. The phone rings and shows 2999 instead of the calling party! The SIPURA seems to be ok: when I connect to Sipgate/Nikotel etc. directly, everything is ok What's wrong? My Asterisk Version is 1.2.1 Conrad

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 provider side as well as of the device side, to prevent codec translation. Unfortunately, Asterisk seems to negotiate the codec for the...

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...)