similar to: incoming callerid on FXO

Hi since "callerid=" in sip.conf doesn't set the Caller ID, I suppose it must be there for some other reason. Is this a not-yet-working feature for future releases of Asterisk? If not, what does it actually do? thanks regards bk

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 I thought I share this one, just in case this is an indication of some bug ... When I was trying to use music on hold at first, I didn't bother to copy any music into /var/lib/asterisk/mohmp3 since there was a sample- hold.mp3 in there which played just fine in a standalone MP3 player. But after uncommenting one of the lines in musiconhold.conf and doing reload on the console, there

Hi all, general query here --- I'm about to set up an asterisk box for use in Japan but can't figureout if it's all ISDN there or what? I have gathered so far that the two major providers, NTT and KVH both offer ISDN lines with .......INS1500 and maybe INS64 protocols? Not sure... But I'm seeing stuff about J1 vs. T1/E1 .... so does that mean I can't use a Digium card it

Hi, I would like to announce the first release of Kemari, an open-source virtual machine synchronization mechanism for fault tolerance. It offers a feasible approach to fault tolerance that does not require the use of specific hardware or modification of applications/OS. Kemari aims to keep VMs transparently running in times of hardware failures. It transfers the state of the primary VM to the

Hi. When I compiled xen-3.3-testing with DEBUG_TRACE_DUMP in xen/include/xen/lib.h turned on, I got some errors at SHADOW_PRINTK. The following patch will fix the arguments passed to SHADOW_PRINTK in xen/arch/x86/mm/shadow/common.c and xen/arch/x86/mm/shadow/multi.c. Although I haven''t tested, it is the same for xen-unstable.hg Thanks, Yoshi Signed-off-by: Yoshi Tamura

Hi, I'm thinking of introducing Asterisk on Linux for IP PBX. Now I'm using ISP that has VoIP service and I have VoIP terminal box for that ISP and a SIP account for SIP server of the ISP. Now, what I would like to do is the following. A. Setup IP PBX on Linux by using Asterisk. B. For IP-PSTN gateway, configure Asterisk to use my ISP's SIP account and connect to my ISP's IP

Hi everyone! I'm wondering if anyone on the list had the opportunity to work with an NTT INS1500 ISDN PRI service before. You see, in Japan, if you receive a call that was just forwarded by another number, the call presentation not only includes the caller (ANI) and your number (DNIS), it will also usually include the forwarding number (REDIRECT). Does anybody know how to extract this field

Hi All, I have a new asterisk setup running at home and am very happy with it. One thing that I am trying to do is to take various actions in the dialplan *if* a particular phone is registered/authenticated/connected. For example, if someone dials *me* and is shows that I am connected via my softphone, to try it instead of my deskphone (and possibly notifiy the user in advance that it is

Hello, This morning, I asked myself if WebRTC could be a viable alternative to softphone deployment. For me, main issue with Softphones is the amount of work needed for installation and configuration. Also, Softphones must be carefully choosen if Deskphone-like quality is expected. Now that WebRTC becomes ubiquitous, it might make sense to trade Softphone features (call history, BLF, ...) for

Hello guys, Is there a decent click-to-dial CTI which works well with Asterisk? We have vanilla asterisk implementation and I have tried a few (ADA, Outcall etc) but they have poor documentation and don't work very well. We are looking for an application which can allow us to dial a number from Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls with call history

Sure would like to hear experiences using various FXO to VoIP gateways with *. It seems that any thread that has anything to do with problematic FXO interfaces goes on forever with speculation about everything under the sun. Unless there is someone out there with the engineering experience to build a better one it is a waste of time, let Digium deal with it. If the TDM400P can ever be made 99.99%

I've been asked for recommendations for a small call centre, an ethernet SIP deskphone with a wireless headset. Similar approach would be a mobile phone with bluetooth head set. Either I've not looked hard enough, or there isn't much on offer. Alec Davis -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and

Hi all: Does somebody know how many meters of cable is supported between Asterisk Digium-FXO cards and the analog telephone ? Thanks, Angel

Hello Everyone, You may have seen the recent announcement regarding new OpenSSL vulnerabilities. <URL: http://www.openssl.org/news/secadv_20030930.txt > Just thought I'd drop a line to head off the usual questions. :-) Don't panic. The vulnerability is denial-of-service. OpenSSL 0.9.7c will be imported into -CURRENT and -STABLE over the next couple of days, and included

Hi all, In order to get the CallerID from PSTN (X100P) I have modified callerid.c file like that: callerid.c [line 256] from: case 3: /* Number (for Zebble) */ to /*case 3: Number (for Zebble) */ Without this modification my own number was displayed as the inoming call CallerID. Now I want to go further. I have activated CallWaiting support on the POTS line. When someone calls and I am in

Hi I have defined my SIP phones like this ... [Sip1] username=gs1 callerid= "Full name" <1001> etc etc Now, when I do this in a given extension exten => nnnn,1,NoOp(${CALLERIDNUM}) then I get "<gs1>" as callerid and not "<1001>" as defined with callerid= Sure, I could set the usernames to their respective extensions, but I don't want

Is the X100P FXO PCI Card capable of detecting a fax, answering the call, and then emailing the fax content to an email address? Thanks. Paul.

I am sure that quite a lot of people would like to have Caller ID working with their X100P and TDM400P cards outside of USA. Judging from previous threads this is just a matter of implementing this support in the software driver! So, I was thinking, if we get together and put few $(USA DOLLARS) into a basket, we could then ask Digium to actually properly implement Caller ID for non USA