similar to: checking dahdi channels

Displaying 20 results from an estimated 10000 matches similar to: "checking dahdi channels"

2010 May 29
6
Best way to limit outgoing calls per trunk
Hi Guys, I am looking to use System() function along with some bash scripting to determine if a Trunk is being used during certain time of the day or not. Here is what I have in mind. Please guide me if you know a better way: exten => s,1,answer exten => s,n,System(/tmp/check.sh) check.sh: check EPOCH time => do an IF for certain times => Allow mutiple calls in certain times and
2010 Jan 18
10
Dahdi/callerid issue
Hi All, Maybe someone knows this, im using dahdi in combination with a TDM400, where 2 analog PSTN lines are connected. The weird thing is tho that when someone calls the analog lines it goes perfectly fine, the line comes in and all works ok. Except: Sometimes the callerid from the caller is not the complete number, but only a few random numbers from that phonenumber, and sometimes its complete.
2010 Apr 01
2
problem compiling asterisk with cdr_odbc
"make menuconfig" does not show cdr_odbc as a selectable compile option. I have compiled and installed both unixODBC and freetds from source and have verified both successfully connect to my sql server. Both were installed to standard locations (/usr/lib). I had no problem compiling cdr_odbc on my test server(CentOS 4.6), however following the same steps on my production server (CentOS
2017 Jan 16
3
Kernel/Asterisk/DAHDI/Libpri version matrix?
I googled about a bit without success, so... Is there a version matrix available? Something that would say: for kernel version w, you can run up to version x of Asterisk, DAHDI version y, and libpri version z? For example, I have a bunch of remote hosts running kernel 2.6.26, Asterisk 11.6.0, and DAHDI 2.7.0.1. We experience occasional Asterisk crashes, so I'd like to get as up to date
2010 Apr 02
3
Asterisk send calls to SIP Trunks with Round Robin Call Distribution
Hi All, I know I can do this pretty easily with one of the SIP Proxy/Routers, I already do this using OpenSER as a load balancer. I have a special requirement that insist an Asterisk server, 1.6.1.x, is used. I will have 2 SIP trunks coming into the server and I will have to send calls to these SIP trunks with a round robin distribution pattern. I was thinking of using a group count
2017 Nov 07
4
Call preemption
Hello, Has anyone already implemented some sort of call preemption in Asterisk ? I am trying to achieve something like this : - I want to limit the number of calls on a given SIP peer to 10 - on the other hand, some calls have higher priority than others - when the ceiling of 10 calls is reached and a call with a high priority is attempted, I would like to drop a call with a lower priority
2010 Oct 16
1
DAHDI, PRI and callerid
Hi, I have just set up Asterisk to use an E1 line with a Digium card. And I can call both in and out, but my outgoing line is all ways identifying itself as the same number, and i can't even change it to another number in the same number series. Do anyone have some clue on how to fix this. I'm using Asterisk 1.6.2.13, libpri 1.4.11.4 and DAHDI 2.4.0. /etc/dahdi/system.conf:
2019 Dec 14
3
USB dahdi fxo ?
I'm moving asterisk to a laptop, so can't use the dahdi board. Is there any supported USB dahdi device ? I see the Sangoma USBfxo device, but the dahdi driver no longer supports it. Anything else ? sean
2009 Apr 14
2
dynamic menus in dialplan
I have an application that needs to vary the menu choices available based upon the availability of an external resource at a given time. What I have in mind is a system that can uplink a user to one of many different satellites. Due to the nature of orbital mechanics a satellite may be out of range at any given time. I only want to present a menu of available satellites. I can query an
2010 Dec 08
5
How to quickly move on to Dahdi channels when SIP provider fails?
Hi Everyone, There are situations when internet connection is lost, SIP provider fails, or even authentication to SIP provider fails, and we want to use the backup Dahdi channels (PSTN). As simple as it may sound but with the many different situations and error messages it seems like it's not so easy to predict all the errors. Is there any single parameter value that can be changed to send
2010 Apr 06
2
Limit Number Of Simultaneous Outbound SIP Calls
Is there a way to limit the number of simultaneous outbound SIP calls made by Asterisk? We've tried using the 'Asterisk sip call-limit' parameter but that doesn't seem to be working and one of our engineers says that parameter has been depreciated. Thanks, Deric.Page at nisc.coop -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Dec 15
2
member (In use)
Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the
2009 Aug 24
1
problem on compiling asterisk-addons-1.6.2.0-rc1
hello, I tried to compil asterisk-addons-1.6.2.0-rc1, and I have that error: [CC] res_config_mysql.c -> res_config_mysql.o res_config_mysql.c:1367: error: unknown field ?update2_func? specified in initializer res_config_mysql.c: In function ?parse_config?: res_config_mysql.c:1432: error: ?CONFIG_STATUS_FILEMISSING? undeclared (first use in this function) res_config_mysql.c:1432: error:
2010 Mar 24
2
new server install errors starting asterisk
here is the issue phones freepbx-2.7.0]# ./start_asterisk start STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. mpg123: no process killed ----------------------------------------------------- Asterisk could not start! Use 'tail
2008 Oct 10
2
Configuring Bandwidth.com SIP trunks to prevent one-way audio
Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that tells Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
2009 Aug 19
7
* 1.4 -> 1.6, zaptel -> dahdi
This has got to be an FAQ, so if someone can point me to where it is answered, I would be greatly appreciated. The documentation for all this stuff is scattered and (to me at least) either very sketchy or hard to find. What I want is a guide for how to convert from 1.4 with zaptel to 1.6 with dahdi. All the recent kernel vulnerabilities are forcing me to upgrade my home server from
2010 Jan 07
1
error compile dahdi with latest kernels.
hello, all of users: there are header files missed when you compile dahdi with kernel-2.6.29 or 2.6.33. i believe that few files are affected: wctdm.c dahdi-base.c wcb4xxp/base.c, opvxa1200.c... the errors look like these: ================================================ from?/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:61:
2009 Nov 13
2
openSuse 11.2 and dahdi-linux
OK, I know it's only just out today but this is what I get when compiling dahdi-linux. make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware' make -C /lib/modules/2.6.31.5-0.1-default/build
2009 Sep 20
1
DAHDI installation warning
Hey list, I'm getting the following warning when installing dahdi-linux-complete-2.2.0.2+2.2.0 : make[2]: Entering directory `/usr/src/kernels/2.6.18-128.1.6.el5-i686' Building modules, stage 2. MODPOST WARNING: could not find /usr/src/dahdi-linux-complete-2.2.0.2 +2.2.0/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.o.cmd for /usr/src/dahdi-linux-complete-2.2.0.2
2014 Mar 21
3
Need more meetme users -- hitting some limit
I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call