similar to: Adding digits to incoming callids depending on context?

I am calling from my laptop to an asterisk box which answers the call and I can hear the voicemail prompts, but the problem is that after so many seconds, MSN Messenger drops the call because it thinks it hasn't been answered by the remote machine. I'm not sure if this is an asterisk problem, or if it is Messenger not knowing the call was answered. Has anyone else run into this sort of

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 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! 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 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 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

Hi, Is it possible to send calls that are forwarded from a Cisco 7900 phone using the Call Forward All feature out using a different service provider or group like another SIP trunk? I don't want to tie up our incoming lines that are ZAP so I was thinking about getting a secondary service for just forwarded calls. Thanks, Calvin -- S i x F e e t U p | "Nowhere to go but

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

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

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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

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

I have a * setup with Grandstream SIP phones dialing out through Nikotel via SIP. When i dial out and the other side picks up, the Grandstream keeps ringing for another seconds and two and the sound coming from the other side is lost. After these two seconds the call is connected find and works flawless. Calling the same path the other way to the Grandstream phone works fine without the cut-off.

Any got experience with these? I couldn't fint anything in any postings... it seems they have a h.323 on voip01.go2call.com and a sip on sip01.go2call.com I have tried to register with some of the same as I use for nikotel, but Asterisk does not want to register. I've tried to use both the user name (ingvald) and the PIN code 440.... as authentication. ---from sip.conf----

(I posted this note on http://www.voip-info.org/wiki-ZyXEL+P2000W+configuration too) I tried to put together comments that were asked for on the P2000W. These configs seem to work fine for a ZyXEL P2000W, thanks to Giles Scott for getting me started with it. DTMF keys work fine and are "read" correctly by Asterisk. It's important that you upgrade your phone to a modern version