similar to: Transfer problem

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

Hi. There is not much info on the FWD site about this. What codecs do they use? When I try to connect with X-Lite, it works with GSM. When I try to call out with *, it wants G729. I have disallow=all and allow=gsm in the sip.conf. I end up getting errors: Unable to find a path from G729A to GSM Unable to find a path from GSM to G729A What's up with that? I was able to make a call once

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user

Hello, I was using the doublehash.patch that Iain Stevenson had created back in August to change the transfer key from a single hash "#" to a double-hash "#". It always patches properly, but when I went from CVS 2004-01-12 to Asterisk 0.7.1 it doesn't seem to work anymore. I've attached the patch to this email and I use the following command to patch it: patch -p1

I have a Cisco 7960 & IpDialogs that I am not able to use the transfer button on it. What happens is that it puts the call on hold and then it gives you a dial tone. You can dial but it will not transfer the call. What we are trying to do is transfer to extension 700 for parking so another person can pick up the line. We can not use the # key to do this due to we have several IVR's

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't

Hi folks, I'm wondering if it is currently possible to configure Asterisk to forward the conversations from 2 channels into a streaming daemon, such as Icecast, so that other people connected to the Internet can listen. The concept is similar to a radio talk-show. The show host would connect to Asterisk via an X100P or through VOIP. His or her voice would then be sent to the streaming

Hi Folks, I''m using a 2.4.x kernel and TC from the iproute2 package so that I can limit traffic through my gateway. I''m using this to mark packets when they leave the LAN: /sbin/ipchains -A forward -j MASQ -i eth0 -s 192.168.1.0/24 -d 0.0.0.0/0 -m 1 When the packets return, I need to have them marked again so that the ingress filter will limit the bandwidth in the opposite

Hello, I recently tried an upgrade of CVS on my test server today and found that the res/res_parking.c file is completely gone. This is where I had to go into the code every time I do an upgrade and change the code to allow for doublehash transfers instead of single hash transfers: That means that you need to hit the pound key twice to initiate a transfer instead of once. Because of our inbound

Attached is the log. I'm building with: clang version 10.0.0 (https://github.com/llvm/llvm-project/ 3a6da1122b990386edeba0987d0d1fdc9c8dc53d) Target: x86_64-unknown-linux-gnu Thread model: posix On some Ubuntu-like distribution. I also ran with ASAN once and it found a bunch of leaks in bin/tco. Best, -- Mehdi On Tue, Apr 7, 2020 at 4:36 AM Richard Barton <Richard.Barton at

Hi all F18 merging has finished so commit access should be back to normal. Thanks Rich > -----Original Message----- > From: llvm-dev <llvm-dev-bounces at lists.llvm.org> On Behalf Of Richard > Barton via llvm-dev > Sent: 9 April, 2020 16:08 > To: llvm-dev at lists.llvm.org > Subject: [llvm-dev] F18 upstreaming Now! > > Hi all > > We are about to merge F18

I don't remember the details of the particular case where we encountered this, but I think the loop started with the condition check and ended with an unconditional branch back to the beginning. -- Krzysztof Parzyszek  kparzysz at quicinc.com   LLVM compiler development -----Original Message----- From: Philip Reames <listmail at philipreames.com> Sent: Thursday, May 30, 2019 3:00 PM

Has anyone else had problems registering with Sipphone over the last few weeks? Previously, this had worked fine. I contacted Sipphone technical support, but they're not much help. register => 17471234567:password@northamerica.sipphone.com/123

Hi, I had been using 4 X100P cards in my Asterisk box, but 2 of them were sharing an interrupt. Therefore, periodically I would hear beeps and clicks that I had assumed were a result of this. So, I ordered a TDM400P with 4 FXO modules and installed it in the box last night. Today, we've had nothing but problems with it dropping calls. I installed the latest CVS of everything, and we've

Theoraplugins updated In-Reply-To: <20040618162546.GA23083@real.com> Message-ID: <00d601c45552$fb9f3210$02efa8c0@guyute> vorbis-bounces@xiph.org wrote: > The RealPlayer 10 plugins for Vorbis and Theora have been > updated. This > release fixes bugs related to SMIL, HTTP playback of static files, > and a few Icecast related issues. I know this is probably in a FAQ

Hello All, I am looking for an R library/function that allows the specification of a custom link function in a linear mixed model. I've been using glm() in library MASS to fit fixed-effect models with a custom link but my study design demands mixed models. Any suggestions on the best R library/function to achieve this would be greatly appreciated. I have tried, to no avail, to

Here's a patch that changes the behaviour of # transfers in asterisk. A single # is transferred to the remote phone/system. Two # in quick succession will trigger a transfer. This is very useful for users who have basic analogue phones connected to an ATA 186. For example, when calling a remote conference or IVR system you often want a single # to be sent to the remote system - not to

Hi Mehdi, I can't replicate those failures at my end, could you let me know what OS, compiler and CMake flags you're using so I can try and reproduce? Thanks! David Truby ________________________________ From: llvm-dev <llvm-dev-bounces at lists.llvm.org> on behalf of Mehdi AMINI via llvm-dev <llvm-dev at lists.llvm.org> Sent: 07 April 2020 06:44 To: Richard Barton

Hi llvm-dev We have not been able to complete all the work we need to do before merging F18 into LLVM as Flang so we will not be dong that today as previously announced. We propose to slip this back a week to let us finish off the last bits of work. All code changes are in review as of Friday. If you want more detail, you can see the exact status here:

On Jun 2, 2020, at 1:57 PM, Kit Barton <kit.barton at gmail.com> wrote: > A few comments on the document: > 1. It seems the current document has settled on using threads on > llvm-dev, however there are still two reference to the LLVM Proposal > Reviews category on Discourse: last paragraph of Proposed Solution > section, first paragraph of the Review Discussion Template