similar to: Autio cut off at beginning of call

Hi All, I am having trouble with receiving calls from FWD via IAX. I know this isn't a FWD support forum, but I suspect the problem is my asterisk setup. The problem is that I can dial out to fwd subscribers, even myself but they can't dial me using my FWD number. I don't know much about IAX, but it would seem to me like a registration problem, but I get no errors or warnings in

I've got a ShoreTel 210 MGCP phone drops calls. My packet capture indicates that the phone may be trying to renew its registration with *, but reports Restart Method of Disconnected (frame 2), then * seems to take that as a sign that it has lost the connection and closes things down. The phone, meanwhile, seems to think it can continue the conversation until a few ICMP "port

I'm having some trouble with placing some iax calls over an openvpn: Setup A is a 1.8GHz Celeron, T100P attached to a Zhone Zplex. Setup B is a 266MHz P2, T100P attached to a Zhone Zplex. Setup C is a 700MHz P3, T100P attached to an Adtran TA 750. Setup D is a 233MHz Pentium, with an X100P. Setups A and B are on the same physical network. IAX calls routed between them work fine. Setup D is

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following conf files 1. extensions.conf (telx-nyc) 2. iax.conf (telx-nyc) 3.

Some of you might already know that we are releasing a new phone, snom 360. To make the phone well-known and stable, we have made a soft phone version out of it and offer it for trial or private use for free (for more details, see the license conditions). There are only few limitations to the phone. First of all, the audio subsystem will work only work with an acceptable quality if you are using

I have been having problems getting PrivacyManager to work correctly. Right now I am running the 1/21/05 CVS but I have been unable to get this to work on asterisk-stable either. You can see from the debug below that the inbound call is arriving via IAX2 and both the CALLING NUMBER and CALLING NAME are both set as "Unavailable". However, PrivacyManager executes and determines that

for some reasons these calls don't really go there where they should go leading to various corruptions of the PMU state Signed-off-by: Karol Herbst <nouveau at karolherbst.de> --- drm/nouveau/nvkm/subdev/pmu/fuc/gk208.fuc5.h | 12 ++++++------ drm/nouveau/nvkm/subdev/pmu/fuc/kernel.fuc | 6 +++--- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git

On 26/02/16 17:19, Karol Herbst wrote: > for some reasons these calls don't really go there where they should go > leading to various corruptions of the PMU state I am fine with the changes but not fine at all with the commit message. it would be nice if you could understand a bit more what the problem is instead of just saying: "it works with this change (TM)" Anyway, 1-3

On Tue, Mar 1, 2016 at 4:45 PM, Martin Peres <martin.peres at free.fr> wrote: > On 26/02/16 17:19, Karol Herbst wrote: >> >> for some reasons these calls don't really go there where they should go >> leading to various corruptions of the PMU state > > > I am fine with the changes but not fine at all with the commit message. it > would be nice if you could

Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan ----- Original Message ----- From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws> To: <timebandit001@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 21, 2005 12:29

Hi, for some weeks now I have been unable to make calls via my voicepulse connect IAX account? When I attempt the console looks like this:- rt*CLI> -- Executing Dial("SIP/2008-cf55", "IAX2/NBhXXXXXX:XXXXXXN82@gwiaxt01.voicepulse.com/12124565900") in new stack -- Called NBhXXXXX:XXXXN82@gwiaxt01.voicepulse.com/12124565900 -- Call accepted by 66.234.228.160

Hi, this is a bit off-topic, but maybe you can help me: When extracting the autio track of a concert on video DVD, I noticed that the *.VOB files each contained only about 16 minutes of music, resulting in multiple files. I extracted to flac for further processing, but now I wonder how to make one continuous sound stream of those: Can I simple concatenate the flac files to make on big file

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

Hi, I need to implement trouble tracking system, we have 250 users in one premise & 3 desktop support technicians. I need to implement trouble ticket system, where user will enter their application / other issues. Mail will be sent to technician available on duty. trouble ticket will be provided to user & will be given close stat once resolved. Kindly suggest me one such application

I am trying to set up a simple Asterisk server. All it's going to do for now is to act as my voicemail box. I've got a DID from Voicepulse, and am using IAX (I'll get to SIP someday when I want to circumvent the phone company for long-distance, but for now I'd be happy to get a trial version of Asterisk running). So far, I've managed to set up voicemail.conf, extensions.conf

Hi, I plan to connect to remote Asterisk that will terminate calls to ISDN primary channel. I'd certainly like to secure this type of service, so would kindly ask for any advice on how to secure this authentication as much as reasonably possible. Since there is long IP route I guess VPN will take too much additional bandwidth... Regards, Robert.

Voicepulse has ignored four emails over the course of two weeks. Anyone have any ideas of whats wrong? - Executing Dial("IAX2/voicepulse-in-01@66.234.228.170:4569/7", "IAX2/acctname:acctpass@gwiaxt01.voicepulse.com/14109649073") in new stack -- Called acctname:acctpass@gwiaxt01.voicepulse.com/14109649073 Sep 13 22:48:25 WARNING[131080]: chan_iax2.c:5375

Good day to all-- First off let me say that I have been very pleased with the mailing list. I have learned a ton of stuff just reading other peoples questions and comments. I really enjoyed the VOIP Conference call on Friday morning. Still working on figuring out the best approach to custom voicemail emails (the reason I joined this group); however, we have more pressing issues. I

I've been using my voicepulse connect number for over a month now, but today it simply won't connect. My partner and I each have a number, both are mapped in my iax.conf and extensions.conf files. This has been working fine. Today, either number gives this message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request

G'Day All What do I type at the command line to stop and start * on a RedHat ES3 box? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041202/f9c92727/attachment.htm