similar to: Possibly I don't understand sip peers

I'm using realtime sip peers and I need to enable a range of IP addresses for a peer. I have: deny = 0.0.0.0/0.0.0.0 permit = xxx.yyy.zzz.0/255.255.255.0 mask = 255.255.255.0 defaultIP = xxx.yyy.zzz.112 host = xxx.yyy.zzz.112 Addresses other than .112 are being denied. Can someone offer assistance? Am I doing something wrong? Bruce

Hello list, this is the communication between an Aastra 5000 PBX and Asterisk, where the Aastra makes a call to Asterisk. For some reason, Asterisk responds with 401-Unauthorized and I don't know why. Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with this Aastra. A1.A1.A1.A1 = IP-address Asterisk PBX AS.AS.AS.AS = IP-address Aastra PBX Aastra PBX makes a call

I'm still trying to work this out. I've got this in my sip.conf [1003polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.215 context=default mailbox=1003 subscribecontext=phonestatus [1004polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.214 context=default mailbox=1004 subscribecontext=phonestatus And this in my extensions.conf [phonestatus] exten =>

I'm trying to make an AMI call. I want to call a number, play an announcement when the call is answered, then call a second number and connect the two when the second call is answered. I an able to make a simple call to two numbers and connect them using the manager API but playing the announcement has me beat. Suggestions anyone? Bruce Ferrell

It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql> describe

Hi All Total noob on the list so all help appreciated.... I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones.... The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't

Hi, we had an attack on a server and we don't understand how it was possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL, network 188.161.128.0/18 Hacked account had following setup: [111] type=friend username=111 context=from-111 host=11.22.33.44 dtmfmode=auto qualify=yes nat=yes canreinvite=no defaultip=11.22.33.44 port=35060 disallow=all allow=ulaw,alaw call-limit=2

Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan,

I know over time SIP OPTIONS message handling has changed and I've seen some write ups that seem to indicate that an s extension in the default context is needed now to get them to work. It's probably my error in any case. So, what am I doing wrong or what do I need to do to get the sip ping to work? Bruce Ferrell Just for fun, I created a sip peer called ping at a fixed address

Scenario: Both boxes are behind firewall, port udp 4569 is open. If I don't want the username and password in dialing string do I have to use register statement in IAX.CONF. Can anybody post some working samples; I have a hard time making it to work with the samples posted on wiki. -- #Joseph

On Thu, 8 Jun 2017, Bruce Ferrell wrote: > Yes, 7 does track upstream. upstream 6 uses systemd also and Scientific > Linux 6 does not. I would say that indicates a solution. Upstream 6 uses systemd? jh

I have a cisco 7940 with the following sip.conf config: [Desk1.1] type=friend secret=****** defaultip=192.168.1.14 insecure=no mailbox=102 callerid="Desk1.1" qualify=500 canreinvite=no context=extensions host=dynamic group=2 I do not get message waiting indicator (mwi) on this phone. Is the another .conf file invilved in configuring this function other than the mailbox=xxx in the

Hello, I have a few entries for sip peers in sip.conf with different name and username, like [TestSIPUser] type=peer host=dynamic username=testuser secret=1234 context=test_context [TestNewUser] type=peer host=dynamic username=newsipuser secret=3456 context=test_context When a call is made from any of these peers I want to get the username of the peer. for eg:- If a call is being made from

so far all the mirrors I've checked have 3.9 in the directory for 3.x Can anyone tell me how to get back versions? I'm looking for 3.4 or 3.5 Thanks in advance Bruce Ferrell

Hmmm.... Can't place calls... Can't access website... Neither of the 3 nameservers answer anything... Anyone heard/know something to explain all this?

Hi I have a new question. Here the situation : I use softphone on 2 computers (soft1 and soft2) located on the same subnetwork. When I register on asterisk server using soft1 with one user (e.g JOHN) which I declared in sip.conf I can register again with this same user using soft2. Is it normal ? I notice that I can pass some call from both but incoming call for JOHN user only arrive on the last

Hi all, I'm looking for pointer for setting up VNC so that access to the system is via gdm/kdm. Yes, I know about vino, and /etc/sysconfig/vncservers but what I'm looking for is a sertup that allows me to see the *dm login screen instead of being dropped direct into a desktop. Thanks in advance

On 09/29/2020 10:01 PM, Bruce Ferrell wrote: > On 9/29/20 6:50 PM, H wrote: >> On 09/29/2020 09:09 PM, John Pierce wrote: >>> do other USB 3 (XHCI) devices work on this system ? >>> >>> On Tue, Sep 29, 2020 at 5:27 PM H <agents at meddatainc.com> wrote: >>> >>>> On 09/29/2020 02:11 PM, Bruce Ferrell wrote: >>>>> Try

Hi, I've gone through the source tree and I don't see a MIB description file for the SNMP agent in asterisk. can someone point me to it. Thanks, Bruce ferrell