similar to: *67 with Sipura 3000

If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for

Hello, I have a bunch of Windows XP Pro machines, Samba 2.2.4, and several plotters on Jetdirect cards, and several printers on Netport 10/100 Muli-port network print servers. I have successfully configured Samba 2.2.4 to allow Windows XP to join the domain controlled by Samba and automaticaly create its own computer account on Samba. I would now like to setup Samba to share all the printers

If in one time 3 IP adresses using internet. TC script: DEV=eth0 # LAN SERVER_IP=192.168.1.2 # eth0 ip address tc qdisc add dev $DEV root handle 1: htb default 255 tc class add dev $DEV parent 1: classid 1:1 htb rate 384Kbit quantum 1500 tc class add dev $DEV parent 1:1 classid 1:20 htb rate 128Kbit ceil 384Kbit prio 0 quantum 1500 tc class add dev $DEV parent 1:1 classid 1:21 htb rate 128Kbit

Hi all. Im trying to shape some traffic, and i see that the best way to do that is using TCNG. The thing is: I dont know how to shape bandwidth per IP. Exemple: 192.168.1.20 ----> 256kbit(down) 128kbit(up) 192.168.1.21 ----> 512kbit(down) 128kbit(up) 192.168.1.22 ----> 180kbit(down) 128kbit(up) 192.168.1.23 ----> 768kbit(down) 128kbit(up) . . . Does anyone has an exemple script

Anyone know what I need to do to get the FXO port on the SPA 3000 to forward calls to Asterisk? My Asterisk is running on port 5061 and I set the dial plan on the device to forward to s@asteriskip:5061 but Asterisk is not picking it up. I can see on tcpdump traces that the Invite packets do go to through to the asterisk machine on port 5061, but it's not picking them up. sip debug does not

Hi; I have ran into an issue with the upsmon tool leaking memory over a two month period. It seems to be due to one of my UPS hosts offline. System is a debian stable with the default nut package. Here's the data so far: companyfs1:/var/log# ps aux | grep [n]ut nut 2439 0.0 3.6 380800 18800 ? S Jun05 6:41 /sbin/upsmon >From my logfiles its failing on one of my UPS

I'm new to asterisk and already a fan. Please forgive me if my questions are covered by some FAQ and thanks in advance for any pointers anyone can give me. The basic problem that I'm having is that sometimes outgoing calls result in a busy signal when the outgoing line is free. I'm thinking that the channel is timing out or something but haven't figured out how to debug or gather

Hello: I have this situation: I can make calls internally, I can make inbound calls but I can't make outbound calls. Thanks in advance. These are my devices: * asterisk 11.8.1 = 192.168.1.22 * sipphone grandstream gxp2160 = 192.168.1.5 * gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4 port 1 (FXS) connected to an analog phone port 3 (FXO) connected to the PSTN These are my

Does any body know if it is possible to group SIP channels just like it is possible with Zap channels? I have a group of FXO gateways (Sipura 3000's) and I would like to treat them as a group the same as I would Zap channels. Does anyone know if this is this possible?

Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has

does anyone know a way to shape outgoing/upload traffic per ip? I have a network and i want to limit the upload with 100kbit per user. Ex: 192.168.1.20 ----> 1024kbit-DOWN / 100kbit-UP 192.168.1.21 ----> 1024kbit-DOWN / 100kbit-UP and so on....... Ive tried CBQ and HTB, but couldnt get is right. the only thing that I did in upload bases was: "tc qdisc add dev ppp0 root tbf rate

I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband

Hi! Bored by previous lhype success, I tried to boot a debootstrapped debian over nfsroot, and lo, this almost works! Commandline: # drivers/lhype/lhype_add 256m 0 ./vmlinux --tunnet=192.168.1.20 root=/dev/root nfsroot=/home/ahu/lhype-root ip=192.168.1.21:192.168.1.20:192.168.1.20 This with lhype.patch current as of this message and 2.9.20-rc4. The boot process stops after: Starting periodic

Hi! Bored by previous lhype success, I tried to boot a debootstrapped debian over nfsroot, and lo, this almost works! Commandline: # drivers/lhype/lhype_add 256m 0 ./vmlinux --tunnet=192.168.1.20 root=/dev/root nfsroot=/home/ahu/lhype-root ip=192.168.1.21:192.168.1.20:192.168.1.20 This with lhype.patch current as of this message and 2.9.20-rc4. The boot process stops after: Starting periodic

I recently upgraded from a few month old CVS version of Asterisk to v1.0.1, and dialing out through my SPA-3000 stopped working. Notice right after INVITE, in the old CVS version, it includes the number I'm trying to dial (8019596) which works fine, however in v1.0.1, it doesn't include the number and of course the dial fails. Did a config option change out from underneath me or

Hello all, I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my home PBX in such a way that whenever someone calls on my trunkline (PSTN) number, he/she will hear a customized ring tone, probably playing an MP3 file, instead of a boring standard ring tone while the extension number that is

Hi in order to keep the dial tone after pressing 9 for 'outside line' I have this in my extensions.conf [localpstn] ignorepat => 9 exten => _9[123456789]XXXXXXX,1,Dial,${PSTN}/${EXTEN:1} exten => _9[123456789]XXXXXXX,2,Congestion this is properly included in the handsets' context but the dial tone disappears after pressing 9. am I missing something? thanks in advance

Problem 1 - Outgoing: I am able to call out of the * box using the analog line attached to the sipura 3000 but when the person being called answers there is no audio from either end. * registers that the call was answered but passes no audio. Problem 2 - Incoming: When calling into the 3000 attached to * it never seems to pickup the line. The phones don't ring on the asterisk side. I used

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm 0.6.4 When making outgoing calls I don't seem to have any control over the CLI that is presented to the called party -- it can be any one of the MSNs allocated to the line, allocated on what seems to be a random basis. This is on a BT Business Highway line (which is essentially an ISDN2e line with two built-in

hi, I am new to * and setting up a test system. here my setup : - debian (from knoppix 3.3) - Asterisk 0.7.1 (from the debian package) - AVM Fritz card used with i4l - softphone I use for testing SJphone on windows - I can make great softphone - softphone calls - I can call from an outside line * and get connected to a softphone here my problem: I can not make outbound calls. I place a call