similar to: Yet another faxing issue..

Displaying 20 results from an estimated 2000 matches similar to: "Yet another faxing issue.."

2008 Apr 03
1
Hearing "transfer" during call
Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:
2008 Jan 03
5
GSM Gateway behind SIP ATA?
I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how
2004 Jul 01
2
Grandstream HT286 1.0.4.63 & Meetme
Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,
2004 Oct 07
2
Asterisk ---- SER ----- GAteway and Reinvite
Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use
2005 Jan 21
1
sip.conf configuration for internal calls
Hello all, I'm a newbie in * and i want to start by making internall calls between ip phones (Grandstream BT100, and HT286), if someone can help me with an ewample of sip.conf file specially with the "register" field in [general] defintion. Thanks D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail sur
2010 Mar 12
1
t38 ATA
Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I've tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex -------------- next part -------------- An HTML
2010 Mar 30
2
Dropped Calls
I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference
2005 May 28
1
Fax and SIP Device
A DID number was dedicated to receive fax, but i have the problem when getting fax call, which call will become a normal phone call and no fax was printed. When fax is detected, the fax extension is executed and dial the extension of the HT486 device (firmware 1.0.5.22). Somehow sending fax out working well. In the mailing lists, i notice some are using HT286 and it work. Could someone share
2007 Mar 23
3
Semi-OT: Use T.38 ATAs to Extend fax lines
Greetings. I have a scenario I would like some advice on. I have a 100,000 square foot building that we will be moving some work crews into. It has offices on each end of the building and a fiber line between them. I currently have an asterisk 1.2 system in place and about 30 phones. My problem is they want a few fax machines out in the warehouse area where I currently have no wiring for
2006 Mar 28
0
codec translation problem???
2005 Sep 23
0
Problem with outbound calls
Hi everybody, I have some problems making calls from a sip user (HT286) to the pstn trough Digium Wildcard TE110P, i allways have an error : SIP 403 INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79 To: <sip:0170708959@192.168.1.4;user=phone>
2018 Apr 11
2
Possible to resize a Windows guest's disk while online?
I'm looking for a way to online resize a Windows disk -- i.e., be able to resize the disk without shutting down, rebooting, or detaching the disk. Is this at all possible? Or am I just barking up the wrong tree? I'm not finding a way to do this and even Amazon has a weird workaround, in which the user must write data to the newly resized-drive in order to recognize the new size
2007 Aug 02
1
A simple IVR extension problem
Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf ---- context=incoming signalling=fxs_ks channel => 4 context=internal signalling=fxo_ks channel => 1 ----- extensions.conf: ---- [office] exten => s,1,Dial(Zap/1,30) [home] exten =>
2007 Mar 23
7
Doorphone vs. Grandstream BT101
I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while
2008 Dec 10
1
SendImage() to Polycom ip550 or ip670
I tried really quickly the other day to send an image to these phones from the dialplan like this: exten => 2821,n,SendImage(/var/lib/asterisk/images/asterisk-intro) or exten => 2821,n,SendImage(asterisk-intro) It didn't work for me. Should this work? Is anyone else using this with Polycom Phones? Bob
2008 Sep 10
1
Color coding plotted numbers
Hi all, I have run a pca and want to plot the samples multivariate space using the sample numbers. In addition, I would like to color code the sample numbers by group, but can't find a way to color-code the sample numbers by group. So, I have to do it in two separate steps: plot(bulbil.pca, col=as.numeric(group.bulbil[,3]), pch=as.numeric(group.bulbil[,3])) # group.bulbil[,3] is grouping
2007 Jan 02
3
yet another faxing issue (outbound only, via ATA)
2 Asterisk servers 1.2.12.1 Connected via IAX2, same switch, GigE, no packet loss, etc 1 with a Sangoma A101 for a PRI to the PSTN Ulaw QoS enabled NAT for the registered ATA boxes, no nat between the * servers Faxing inbound: Call from PRI hits the first Asterisk server Then talks to the 2nd via IAX2 NVFaxDetect receives the fax, converts to PDF and emails it out Works great!
2005 May 08
3
Grandstream firmware 1.0.6.2
Grandstream owners, I just noticed that there is a new firmware release, for those that are interested: http://www.grandstream.com/BETATEST/ Doug
2020 Aug 11
2
[PATCH] x86/paravirt: Add missing noinstr to arch_local*() helpers
On 11.08.20 10:12, Peter Zijlstra wrote: > On Tue, Aug 11, 2020 at 09:57:55AM +0200, J?rgen Gro? wrote: >> On 11.08.20 09:41, Peter Zijlstra wrote: >>> On Fri, Aug 07, 2020 at 05:19:03PM +0200, Marco Elver wrote: >>> >>>> My hypothesis here is simply that kvm_wait() may be called in a place >>>> where we get the same case I mentioned to Peter,
2020 Aug 11
2
[PATCH] x86/paravirt: Add missing noinstr to arch_local*() helpers
On 11.08.20 10:12, Peter Zijlstra wrote: > On Tue, Aug 11, 2020 at 09:57:55AM +0200, J?rgen Gro? wrote: >> On 11.08.20 09:41, Peter Zijlstra wrote: >>> On Fri, Aug 07, 2020 at 05:19:03PM +0200, Marco Elver wrote: >>> >>>> My hypothesis here is simply that kvm_wait() may be called in a place >>>> where we get the same case I mentioned to Peter,