similar to: Reliable European SIP/IAX Providers?

I have been trying to make Broadvoice match incoming Caller ID and do specific things based on the number received, but due to Broadvoice requiring the "s" to start off an incoming extension, I cannot get this to work. Has anyone been able to do this? Here are some examples of my setup: from sip.conf: context=broadvoice-incoming from extensions.conf [broadvoice-incoming] exten =>

I currently have my astcc databases card lenghts at 7 digits long. I would like to expand this to 10 digits now though. Will I screw things up if I leave the old 7 digit long pins in there and start using/generating 10 digit pins? Thanks

I've been having a lot of problems with Broadvoice lately. Anyone else been without service for extended periods of time this week?

We are using the following to record conversations. exten => _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP}) exten => _1XXX.,2,Monitor(wav,${CALLFILENAME},m) exten => _1XXX.,3,Dial(IAX2/4506:zj5S3A5a@nl.voipgate.nl/${EXTEN:1}) exten => _1XXX.,4,Congestion exten => _1XXX.,104,Congestion This was working previously to record both sides of the conversation but now

Hi every all, since a few weeks I came back to asterisk and tried to install version 1.6. The installation went fine so I decided to buy new dids on Voxbone. I have added the sip peers of Voxbone Belgium1 like this in the sip.conf [81.201.82.39] host=dynamic type=friend insecure=very context=your_context canreinvite=no qualify=no deny=0.0.0.0/0.0.0.0 permit=81.201.82.39/255.255.255.255 but

Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound

Hi, We have been using Voxbone for some time for origination, and they now offer E911 services.? We are trying to set this up and having trouble meeting their authentication requirements. I setup a peer as I normally would, with user/pass as they supplied ("lacoursj", "pass"), but my calls are rejected.? Their support is asking that I follow this auth mechanism: 1st step

Hi List, I recently signed up with Voxbone to get some International DIDs. I was just about to purchase a DID this morning... but when I went to get it.... voxbone wanted the end user's address information. So I started to put it in... unfortunately... the end-user is in the U.S....but the only options are for a few select cities in GERMANY! I don't understand. Is there some

I *am* doing that, as I assumed it would be required just for the 911 mapping we have provided, but that doesn't change the SIP header. Cheers, j On 05/08/2018 02:41 PM, Khalil Khamlichi wrote: > try setting the callerid with > > same => n,Set(CALLERID(all)=17864089672 <17864089672>) > > ofcourse for each customer you will need to provide his own did. > > >

Signed-off-by: Tiger Yang <tiger.yang at oracle.com> --- Config.make.in | 1 + configure.in | 6 ++++++ fs/ocfs2/Makefile | 4 ++++ kapi-compat/include/pipe_buf_operations.h | 10 ++++++++++ 4 files changed, 21 insertions(+), 0 deletions(-) create mode 100644 kapi-compat/include/pipe_buf_operations.h

I posted the same question on asterisk-biz mailing list but didn't have much response. So I am posting it here now. I need a good, reliable and stable DID provider for USA, Canada and Europe. I prefer to have fixed monthly rates for incoming and outgoing calls and not per minute charges. Features I need to get with DIDs are: 1. my own caller ID and caller name on outbound calls 2. multiple

Found 15 patches (out of 162) that appeared relevant to ocfs2 1.4. Please review. Sunil

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. I recognize that some action might be required by another provider which is outside Vitelity's control, but it seems that they should at least be trying to help resolve the problem by helping me determine

After months of BroadVoice ignoring my trouble tickets for dropped calls, delayed termination, etc., I'm throwing in the towel. While they have credited $19.95 to my account, they refuse to credit anything more, despite ALL of the problems I've had. I feel the least they could do is credit the remaining $8.61 to my account, yet they won't. I haven't really been following up on

All, I'm in Dallas, TX, US and am looking for inbound-only DID service with 10+ channels on a SIP trunk. Is anyone on this list doing something similar and have any recommendations for a provider? Of course I'll be routing either SIP/IAX to an asterisk server that will be hosted in a Dallas colocation facility. I found VoxBone.com but they only want to deal with customers that can

Hi. i am running asterisk 11 and i would like to have features access codes in my system such as call waiting(all types) (enable/disable), call forward (enable/disable) and DND. my dialplan is pretty simple and it is the following [DefaultPlan]exten => _XXXXXXXXXX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten => _XXXXXXXXXX,1,Busy() exten => _4XX,2,Answer()exten =>

Dear, Currently, I'm looking for a UPS that can support NUT. I found an APC Smart-UPS 5000VA 230V SRT, but I'm not 100% that is supported. *APC Smart-UPS 5000VA 230V SRT* http://www.apc.com/shop/nl/nl/products/APC-Smart-UPS-SRT-5000VA-230V/P-SRT5KXLI I searched in http://networkupstools.org/stable-hcl.html and I found that you support "*APC Smart*" I would just to confirm

We currently use KLIBCKERNELSRC and KLIBCKERNELOBJ variables which used to be required to point to a kernel source and object directories. However the kernel headers were reorganised starting in Linux 3.7 such that that doesn't work properly. This series removes KLIBCKERNELOBJ and changes the use and documentation of KLIBCKERNELSRC to be consistent. Ben. Ben Hutchings (4): [klibc]

That is the general idea. But how do i make it work? is there somewhere ready? On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkiniston at gmail.com> wrote: > Define your *72 and *73 extensions in your internal context, Have them set > a value in the ASTDB that you then check when dialing your handsets. > > The same can be done for call forwarding, store a number in the