similar to: Adit 600 over MGCP.

Has anyone ever gotten an Adit 600 to work with Asterisk1.4 via MGCP. Asterisk keeps giving me the following error in the LOGs: [Jun 30 08:32:59] NOTICE[26785]: chan_mgcp.c:1726 find_subchannel_and_lock: Gateway 10.0.0.245' (and thus its endpoint '*') does not exist MGCP Config: [AFSWestAdit600] host = dynamic context = default canreinvite = no threewaycalling = yes

Hi all, I try setup LVS and samba server and i found one problem when I registered computer to domain. I drawed picture and attach it. client(195.146.40.2) | | LVS VIP(195.146.40.1) DIP(192.168.30.1) | | samba(192.168.30.10) For clearly understand : I have two networks, external and internal. When nmbd receive announcment from client send answer to client. Answer packet contain ipadress of

Anyone using the G729 codec to create a h.323 trunk between an Avaya Communication manager and Asterisk Freepbx System and works? I don't have the G729 codec installed on the Asterisk and running G711MU on avaya and getting invalid codec when calling from Avaya to Asterisk. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! There's any way to set up a call using G726 (sip peer) receive it on Asterisk convert it to G711Mu to send it to PSTN broadband termination? I've put the following in sip.conf: disalow=all allow=gsm allow=g726 (my TAs use G726 32K) best regards, Helder -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi folks! Our company are going to buy an E1 line with Euro ISDN and 30 lines (channels). This is how it will be configured: 3 Lines, of the total of 30, is going to be for the company phones, and share one phonenumber (eg. 555-12340). 1 Line will be dedicated to a specific unique phonenumber (Fax) (eg. 555-54321). The rest of the lines/channels (26) will be used by (by, not for) our customers,

AgentCallBackLogin is going to be deprecated, so I've decided to emulate chan agent using AQM and RQM funcions and Local channel. I use asterisk 1.2.13 and latest 1.2.x. zapata. I used example 2 from http://www.voip-info.org/wiki/view/Agents+without+agent+channel and example from queues-with-callback-members.txt from asterisk 1.4 doc directory. My dialplan is very similar to Digium's

> we point them to somedom.contoso.com which is then > resolved to both and the client can pick I want to avoid Round-Robin DNS > You cannot use CTDB with AD DCs You can. /etc/ctdb/nodes 192.168.30.11 192.168.30.12 /etc/ctdb/public_addresses 192.168.30.10/24 ens18 /etc/ctdb/events/legacy/20.samba-ad.script #!/usr/bin/env bash function monitor_ldap() { ldbsearch -H

Dear list, I am trying to configure the nagios plugin called check_sip. I just read the README file included with the plugin. I follow the readme steps to configure the plugin, without success. In the nagios web interface I can see (No output!) In the status information column. If I run the chech_sip plugin from a linux console, I get /usr/local/nagios/libexec# ./check_sip -u

I have a Cisco 2600 which understands the "gsmfr" codec, which appears to be what Asterisk calls "gsm" -- at least it ends up using it. I also have a PSTN gateway which is speaking ulaw. When the 2600 calls through Asterisk to the PSTN, it negotiates the g711ulaw codec, but when the PSTN calls through Asterisk to the 2600, it seems that Asterisk is doing translation, and it

Hi Gurus, We have configured asterisk to trunk with avaya with ooh323 channel driver. The sip phone registered on asterisk can dial the extensions registered on avaya via this trunk , and vice versa works too. Even we can make the avaya branch to dial asterisk?s extension and then this extension dial back to another avaya?s extension. But if we dial the external DID number via this trunk from

Hi team, I am trying to find solution to hangup b-party call after 1 min with out disconnecting the call of a-party but following dial plan which is disconnect both the calls. Please suggest me the solution. [TB] exten => _X.,1,Wait(${INCOMING_WAIT}) exten =>_X.,2,Verbose(TB) exten =>_X.,3,Answer() exten => _X.,4,Set(mainLoop=0) exten =>

Hello all, I am having an issue with a BT-101 and * . When dialing a number from the BT-101, upon the remote side answering, the call is established but no audio is passed in either direction. I have tcpdump'd this session and found this: (192.168.30.1 is * - 192.168.30.32 is BT-101) 22:41:47.899462 IP 192.168.30.32 > 192.168.30.1: icmp 36: 192.168.30.32 udp port 5004 unreachable

Hi all - This isn't really directly Asterisk related, but has anyone successfully set up a Polycom phone to register two lines on two different Asterisk boxes? I can get the first line to register, but the second one does not. I can still place calls from that second line, which indicates to me the server, user, and secret are correct. I'm running the newest 2.6 series firmware with the

Hi, below are some test results from implementing a prio qdisc ''that is also below''. The qdisc is attacted to a vlan interface for my external network. Both tests were run at the same time. The links are policed at 6.0M ''by our provider''. 192.168.70.1 --> 192.168.30.1 My question is: If using a prio qdisc should''nt the iperf run with a tos of b8 have

Hello everyone! I'm trying to get tinc running, but couldn't make it until now... My current situation is: all tincs are connected, but there ist (almost) no traffic: Only two PCs are able to ping to another, but not to itself. Let me explain, what I'm planning to do: I've got four networks: PROM: 10.69.0.0/16 PAN: 192.168.30.0/24 (I'm afraid, I can't change PAN to

We have recently install an asterisk solution with about 60 physical extensions. While the system is running it runs reasonably well (Still have a few teething problems) but twice now they have experienced a degradation in voice quality and dropped calls and then finally asterisk completely crashes out. Restarting asterisk will work for a little while and it will crash again, each time less time

(refers to posting "Samba 3.0 + ADS, winbind problem" from August, 28th) Setup: client: SuSE8.2 professional (kernel 2.4.20-4GB) with openldap2 2.1.12 and heimdal kerberos 0.4e from the SuSE CDs and Samba 3.0.0RC3 compiled from source with flags "--with-ads --with-pam --with-acl-support". server: Windows 2003 Server as Active Directory Controller (configured as pure

What server hardware would you recommend for an Asterisk system which are really critical? The additional hardware will probably be two digium TE110P cards, and an Adit 600 platform. If it's possible to run on -48VDC, It would be great! Are there any experiences with any HP or FujitsuSiemens systems? Or other "complete server" systems? Thanks! BR Daniel Nystr?m

Maybe this is a stupid idea, but what we are doing here is using just the domain name for all sorts of services, i.e. LDAP. So instead of pointing a client to dc1.somedom.contoso.com or dc2.somedom.contoso.com, we point them to somedom.contoso.com which is then resolved to both and the client can pick. Best regards Johannes Am Mo., 23. Nov. 2020 um 17:02 Uhr schrieb Rowland penny via samba <

On 10.07.20 13:01, Rowland penny via samba wrote: > On 10/07/2020 11:45, basti via samba wrote: >> >> On 10.07.20 12:39, Rowland penny via samba wrote: >> >>>>> What OS and version is this ? >>>> debian 10 >>>>> What Samba version ? >>>> 2:4.9.5+dfsg-5+deb10u1 >>>> >>>> only winbind installed.