similar to: Phone Giptel G100 with Asterisk?

Hi All, I have installed a number of Digium G100 devices in many countries like South Korea, Japan, Singapore and Australia. I have just installed two in New Zealand and both sites are having a problem with Caller ID. Incoming calls are dropping the first digit 0 from the caller ID. I was previously using DAHDI and a TE121 device which may have been adding the 0, I'm not too sure about

Hi, We recently dumped a Xorcom box that was no end of trouble and replaced with a Digium G100.  PRI came right up, and we have been using it fairly flawlessly for several months now, with one caveat.  Calls that arrive from the PRI are sent to the asterisk instance (13.23.1, chan_sip), then routed by the dialplan to various other gateways or upstream providers.  When the call finally lands

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

Message: 1 Date: Mon, 14 Feb 2005 09:53:36 +1100 From: "PHP Mechanic" <oliver.bode@phpmechanic.com> Subject: [Asterisk-Users] Who makes these phones? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <08d401c5121e$dbea4750$0200a8c0@oliver> Content-Type: text/plain; format=flowed;

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, I am trying to color the data points on my boxplot but I only get the default black. Not sure why. Any ideas? Thank you, Mark Here is my code. > dat.g100 <- dat.sg[100,] # selected gene (row) from a dataframe of 37 columns > d1 <- as.matrix(dat.g100[,1:18]) # subset 1 > d2 <- as.matrix(dat.g100[,19:37]) # subset 2 # boxplots > par(mfrow=c(2,1)) > boxplot(d1,

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

Funny - I downloaded the latest Asterisk CVS, and it's pretty much working. Will report when I have some more success. PaulH -----Original Message----- From: Philipp von Klitzing [mailto:klitzing@pool.informatik.rwth-aachen.de] Sent: Tuesday, 28 September 2004 9:46 PM To: Paul Hales Subject: Re: [Asterisk-Users] Leader IP10S Hi! > I have been lent a Leader IP10S phone (SIP) for

Hi there, I have serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542

I'm trying to sort a DATAFRAME by a column "ID" that contains alphanumeric data. Specifically,"ID" contains integers all preceeded by the character "g" as in: g1, g6, g3, g19, g100, g2, g39 I am using the following code: DATAFRAME=DATAFRAME[order(DATAFRAME1$ID),] and was hoping it would sort the dataframe by ID in the following manner g1, g2, g3, g6, g19,

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 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, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protect the server from unauthorized users or is there

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure because anyone can bypass the SER and register themselves as a peer with the asterisk. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of

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

I'm not sure that the problem lies in the NAT because the phone is talking to Asterisk. I'm hoping this is a simple config thing I've overlooked but I've tried all kinds of combos inside the [] in my mgcp.cfg file. The phone's IP is 192.168.1.116 (my comp is .110). The router to which the phone and my comp is plugged into has a WAN IP of 10.0.0.28. All the other comps (and SIP

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