similar to: Accepting SIP calls from unregistered gateways

Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on

Asterisk seems to regard an unregistered phone to be busy. Is that correct? Is not an unregistered phone unavailable? It is odd to me that if someone dials an unregistered extension, then the dialplan jumps to busy and voicemail kicks in saying that the person is on the phone, when clearly they can't be if the phone hasn't registered. Any way around this?

Hi everyone, IAX2 softphone is not working with the “Unregistered” part in the asterisk (chan_iax2.c and data_pgsql.c) But with the Xlite softphone the unregistered worked properly and ast_data properly updated the IP address and port number in the database. I have seen some codes in the chan_iax2.c file: “ast_verbose(VERBOSE_PREFIX_3 "Unregistered '%s' (%s)\n", p->name,

When I do a "reload" in the Asterisk CLI I get a long list "Unregistered indication country" lines during the parsing of the features.conf file. Then, when parsing the indications.conf file, they seem to all get re-registered (lines saying "Registered indication country" are displayed). What do these lines mean and why are they unregistered and then registered?

With the exception of modesets which would switch the DPMS state of a connector from on to off, we want to make sure that we disallow all modesets which would result in enabling a new monitor or a new mode configuration on a monitor if the connector for the display in question is no longer registered. This allows us to stop userspace from trying to enable new displays on connectors for an MST

With the exception of modesets which would switch the DPMS state of a connector from on to off, we want to make sure that we disallow all modesets which would result in enabling a new monitor or a new mode configuration on a monitor if the connector for the display in question is no longer registered. This allows us to stop userspace from trying to enable new displays on connectors for an MST

On Tue, 30 Aug 2016 10:39:14 -0400 Eric Wieling <ewieling at nyigc.com> wrote: > The dialplan below cannot go to voicemail, either something else is Of course not. It's the individual extensions that have voice mail. I have a similar problem when one of those destinations is a cell phone but I know that there is no Asterisk solution for that problem. If the cell phone answers and

I have a remote user with Eyebeam on a laptop. Internet connectivity seems good, there is no packet loss to that location from the PBX. Everytime the user starts eyebeam, the application tries to register. Asterisk accepts the registration but the reply never gets to the client application, so it thinks it has not been accepted and times out. Then Asterisk unregisters the extension. --

On Mon, Oct 08, 2018 at 07:24:30PM -0400, Lyude Paul wrote: > With the exception of modesets which would switch the DPMS state of a > connector from on to off, we want to make sure that we disallow all > modesets which would result in enabling a new monitor or a new mode > configuration on a monitor if the connector for the display in question > is no longer registered. This allows

On (kubuntu) Ubuntu 11.10 amd64, gcc version 4.6.1-9ubuntu3 Using the 1.5.1 source, and the --enable-win64 option on ./configure. It seems to ./configure and make ok, I solved the dependencies early on, but during install it complains of this: Code: Error in file "/usr/share/applications/gnumeric.desktop": "zz-application/zz-winassoc-xls" is an invalid MIME type

I am using asterisk-oh323-0.7.2-pre and CVS Head of Asterisk. Oh323 Module compiled without errors. But When I try to stary Asterisk with the Oh323.so file in the modules folder, Asterisk is dying with the following error. [chan_oh323.so]Aug 2 14:08:14 NOTICE[18873]: res_musiconhold.c:490 monmp3thread: Request to schedule in the past?!?! => (InAccess Networks OpenH323 Channel Driver) ==

Hi! I'm using asterisk with a few kphone SIP-Clients. The registration process seems quite OK. But there are some problems: Calling other registered users is possible, but the rtp-stream is not reaching the right port, so you can hear nothing. In ethereal you can see, that the SIP/SDP fields addresses different ports at each client, so client A sends to port 32000 but client B listens on

Hello, i've recently configured my asterisk for internal sip calls. while testing, i noticed that 1 out of 10 calls works.. at first i thought my router dropping packets around the way as it were a bottle neck.. so i've added a switch. once i tested again same prob occurs... im using xlite as a softphone on clients pc and centos server on a dedicated machine. at times the phone call

Hello dear list. A couple of years ago, I worked with a Alcatel IP pbx and Alcatel Sip phones, and we had the opportunity to unregister user by typing *-a number and -* again, ex * 99 *, and then the phone number/sip extension was unavailable, and all of the calls to that extension was redirected to the receptionist. When the user came back and wanted to register her sip account/extension, the

Hi perhaps someone else already had the same problem. Problem description: I''m running a class B University network with approx 10k hosts attached. I would now like to stop traffic from and to hosts in my network not already registered in my DNS server. This means I''ve to handle with approx 50k rules|routes. Sure I can summarize the unalloctaed address space a little bit with

I have a question. I have two numbers on Asterisk like 902121234567 and 902123645789 and i want to divert first number's call to Trunk if second number is unregistered. Is it possible? ?f yes, how? Flow Diagram: *Two numbers are registered on Asterisk 902121234567---------------------------- registered to Asterisk

I'm proposing an asterisk configuration and considering the use of multiport SIP/FXS gateways (instead of T1 cards and channel banks). I'm looking for products similar in function to the Cisco ATA-186, but with more ports. I've seen the manufacturer's web pages for the Audiocodes MediaPack (http://www.audiocodes.com/) and the Mediatrix (http://www.mediatrix.com/) access devices.

So what are people using these days for SIP or IAX to PSTN gateways. 1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow you to use your own SIP device (phone or something like *) instead of the interface hardware they usually provide? 2. What about latency and reliability? 3. Finally, do any of the providers deliver more than one call via SIP? In otherwords, if

I have several Cisco SIP gateways sending calls to Asterisk. Because the gateways don't have user-agents, they don't authenticate with Asterisk. And because they don't authenticate, they use the default context in the sip.conf file. Is there a way to either: A) identify the inbound gateway with a variable, in channel info, or the manager interface? If there was a ${SIPDOMAIN} for

Hi, I have an incoming call that I would like answered every time by a different SIP phone (out of 50). Also, some of the phone may not be available (may be turned off and thus unregistered with Asterisk). Any way of doing this? Paulo H. Mannheimer -------------- next part -------------- An HTML attachment was scrubbed... URL: