similar to: IAXtel + NAT

Hey all, I was playing around with IAXTEL last nite and have outgoing calls working a treat. I'm sure I woke a few people up in the US with my annoying test calls. :) Anywayz, incoming calls are a different matter. I have a NAT firewall my * box is sitting behind and the server 'appears' to have registered correctly with IAXTEL. Thing is, when I try and call my 1700 number

I need to avoid MOH on my asterisk box, so i need to have a ringing tone when attendant transfer is made, or a call is on hold.. Is there any way to do that. I did not see a simple way to do that. Regards

I''ve been trying to migrate a win nt 4 machine to a xen domu for the past few months with no success. However, on my current attempt, the original hardware no longer boots, so I''m trying to resolve the issues with xen properly, or else take a long holiday... Anyway, the physical machine had a 9G drive (OS drive), a 147 G drive (not in use) and a 300G drive (all SCSI Ultra320 on

Hi all, I have call recording enabled via the Monitor command and it seems, the call stops being recorded after the call is transferred. Is this normal behavior? If so how can I continue recording of calls after they have been transferred....

Just wondering if there is a reason PPP support is compiled into zapata by *default*: # Uncomment for Generic PPP support (i.e. ZapRAS) # KFLAGS+=-DCONFIG_ZAPATA_PPP Especially since the comments imply that it should be commented out by default... The main reason I ask is because I usually try to re-compile the kernel to only include the bits that I need, and so I don't include PPP...

Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600.... Is there any additional debug apart from "show hints" to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =-

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

Hi, I was just wondering if it is possible to cause a script to run (configured in the domu.cfg file) each time "xm create domu.cfg" is run, but before the machine is actually started? ie, I''d like to "setup" the disks for the VM before xen/qemu tries to use the devices and allocate to the domu. I''m using xen 4.1.3 on Debian testing, and using the xm

you can do that. But are u installing qmail and * on same box. i wont recommend that. i use qmail and *. qmail is strictly for internet email. * is on separate server not exposed to Internet. * box also has sendmail. i hv configured sendmail to use smart host (qmail server). This way its safe and secure. HTH, -B ----- Original Message ----- From: "Ing Isianto Istiadi"

I would just like to get a refresh of the situation for Australian users. It would seem that the TE400P is currently available and is likely to acheive approval for use in Australia within the next 2 months ? (when is the end of summer?). Once this is done, it will certainly suit larger installations, but it still leaves a number of 'gaps'. Anyone with analog phone lines will need

I have a peculiar situation, and am hoping someone on the list can offer assistance. I am running CVS HEAD, and am using ITSPs for DIDs. The server has no Zap hardware, but is configured to use ztdummy. All incoming calls are via IAX2. Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also able to transfer calls among my SIP devices, voice mail, IVR, etc. All of my SIP

Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? Thanks, Waldo

Hi, I get fascinated when I dial someone and get an IVR play " for accounts department press 1, for sales, press 2 or hold the line for an operator" and then have MOH play before the line is picked up at the desired extesion. Please, permit me as I know this will be one of the dumbest questions to ask in a group like this. I'll apprecaite any specific guide/instruction. Thanks

Hi, Is it possible to configure asterisk so it doesn't play MoH from certain phones? Regards, Jan

On Thursday 31 August 2017 at 18:15:54, Joseph Smith wrote: > I was hoping Asterisk would handle more than 4k simultaneous calls. I know from experience that Asterisk can handle more than 4k simultaneous calls, however it's an extreme case to have all of them playing music on hold. I think that if you tested 4k simultaneous calls with standard media streams on the majority of them, you

Hello, does anybody know about an existing skill-based routing solution for asterisk? I found only some theoretical documents on voip-info.org. I would like to have finer control over who can get which call in which order. Example: Several operators with several topics. Each operator may have a given knowledge-base for given topic. Topics may be weighted in question of complexity as well. Some

Hello I did not mention it but of course the MOH directory is listed in /etc/asterisk/musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh [myfolder_1] mode=files directory=/var/lib/asterisk/moh/myfolder/1 sort=alpha [myfolder_2] mode=files directory=/var/lib/asterisk/moh/myfolder/2 sort=alpha [myfolder_3] mode=files directory=/var/lib/asterisk/moh/myfolder/3 sort=alpha

Le 23/03/2017 ? 20:17, Jonas Kellens a ?crit : > Hello > > > is there any more information on how to reload/read musiconhold files ? CLI> module reload res_musiconhold -- Daniel > On 07-03-17 10:46, Jonas Kellens wrote: >> Hello >> >> I did not mention it but of course the MOH directory is listed in >> /etc/asterisk/musiconhold.conf : >> >>

Good morning, i've noticed many times that there are IVRs that play a ring tone just before bridging me to an agent. My asterisk does not behave like this but i've always wanted to. I'm now playing with 1.6.2.9 and i've read in queue's doc: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue R ? stops moh and rings once an agent is ringing (Asterisk Trunk) (in

Hello list, I have defined a new MoH-class in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; *[106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes* In sip.conf I have this commented out : ;mohinterpret=default ;mohsuggest=default Asterisk sees these moh-classes and files : vps2301*CLI> moh show classes Class: default Mode: files