similar to: unable to create channel of type 'SIP'

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100802/4f58fea7/attachment.htm

I want to know about asterisk and openBTS If anybody made any test and experience... Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/cbd08740/attachment.htm

Hi list, I have searched through asterisk command lines but haven't found how to do this: - can I list the phones (callerid or IMSIs?) currently registered ? If I do "dialplan show" that lists the configuration I loaded, e.g [ Context 'sip-local' created by 'pbx_config' ] '2102' => 1. Macro(dialSIP|IMSI1) [pbx_config] '2103'

Several weeks ago, we posted a message seeking information about time sources.? There were many helpful and educational responses. An excerpt from one of the responses is included below.? We have been following up with regard to how SDR capabilities might be used for obtaining time using SDR dongles as well as using the time source product referenced in that response. Our SDR investigation has

Friends, After listening to Mark Summer's keynote at Astricon (hopefully soon on the Astricon web site) I think we should come back to the discussion he started. Mark talked about using Open Source in general and Asterisk in particular in third world projects as well as in emergencies in other countries. He and Inveneo help groups of people to get a better understanding of how to build

Hello, I'm working on an Opus encoder block for GNUradio (a signal processing toolkit). I was wondering if there's some formula/heuristic for estimating the packet size average case / worst case given a certain encoder setting (assuming VBR). I need to provided a reasonable estimate to the GNUradio memory allocator. --Albin

Hi list, Currently, one of my phones registers fine, and the other does not, though for me they have the same config... Can somebody help debug/understand why? The logs in asterisk say: [Feb 24 13:48:09] NOTICE[20626]: chan_sip.c:15642 handle_request_register: Registration from 'IMSI208300618462231 <sip:IMSI20830061xxxx at 127.0.0.1>' failed for '127.0.0.1' - No matching

Thanks for the input! --Albin On Sat, May 12, 2018 at 6:00 PM, Orestes Zoupanos <oresteszoupanos at hotmail.com> wrote: > Hi Albin! > > There may be some details at: https://tools.ietf.org/html/rfc7845#section-6 > > Otherwise, I hope someone else on this list might be able to give a better > formula for estimating packet size. > > Regards, > > Orestes >

Hi All, I am using Asterisk for one of my projects in OpenBTS. I am having the age old problem of "extension not found" when try to make a call from one registered SIP phone to other registered SIP phone (two mobile phones connected to Asterisk via OpenBTS). The exact error thrown on Asterisk CLI is *"chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to

Bugzilla has received a request to create a user account using your email address (zfs-crypto-discuss at opensolaris.org). To continue creating an account using this email address, visit the following link by February 28th, 2008 at 03:48 PST: http://defect.opensolaris.org/bz/token.cgi?t=F9zVJ7vXDC&a=request_new_account If you did not receive this email before February 28th, 2008 at 03:48

I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being

Note also that the packet size you give the encoder also acts as an absolute max on the bitrate. For example, if you ask for 32 kb/s VBR but give a max packet size of 120 bytes, then you're absolutely certain the bitrate will never go over 48 kb/s. Jean-Marc On 05/12/2018 12:42 PM, Albin Stigö wrote: > Just a follow up... I guess I was a bit confused about the VBR > setting. I realise

Hi I met a problem in asterisk, please see message in the following, the detail debug log is in the attached file. can someone help to point out where to correctly configure asterisk, thanks a lot ! BR/Scott -------> -- Executing [8690 at phones:1] Dial("SIP/IMSI466990004244439-00000014", "SIP/IMSI466974104638690") in new stack Really destroying SIP dialog '

Guys, OK, now that we have the new core contributors squared away, we can go back to finishing the contributors list. There were some people who got added and then there is existing contributors list. I think Darren''s suggestion to wait to add new core contributors is fine (let the new constitution settle down and we can in the meanwhile have a more inclusive look at both rather than

Anyone on this list have extensive experience with BTS? http://deancollinsblog.blogspot.com/2009/03/open-bts.html Please email me, particularly if you have experience in deploying over multiple cells covering large geographical areas (200k's sq). Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York

Now that FLAC has officionally joined the Xiph family, there are three different free audio codecs, which can be stored/transported using the ogg container. Would it be possible to create an API and/or a library that would cover them all? If my understanding of the situation is correct, if an application wants to support vorbis, speex and flac, it has to use three different interfaces.

Now that FLAC has officionally joined the Xiph family, there are three different free audio codecs, which can be stored/transported using the ogg container. Would it be possible to create an API and/or a library that would cover them all? If my understanding of the situation is correct, if an application wants to support vorbis, speex and flac, it has to use three different interfaces.

Now that FLAC has officionally joined the Xiph family, there are three different free audio codecs, which can be stored/transported using the ogg container. Would it be possible to create an API and/or a library that would cover them all? If my understanding of the situation is correct, if an application wants to support vorbis, speex and flac, it has to use three different interfaces.

Hi list, I have a problem with installing Asterisk (under https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages): sudo apt-get install asterisk-1.8 Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting 'asterisk' instead of 'asterisk-1.8' Some packages could not be installed. This may mean that you have requested an