search for: pbxes

Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the "domain" field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with th...

if i run let's say 1 pbx running on my main linux box and a another on my windows box if a person dial my main number and press lets say 1 are it possible to transfer the call over to my other pbx hope anyone understand -------------- next part -------------- An HTML attachment was scrubbed... URL:

...hat i think i have understood but probably have not. iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. trunking will save some bytes in flight iff one has four or more streams moving between two pbxes. but who would want to have the pbxes in the data stream anyway? reinvite rules, especially in a geographically distributed use scenario. now, i could see a network of iaxen if there was some way to negotiate call routing with costs etc. but trip looks a bit ugly and kinda far away. and it cer...

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around...

Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK, Germany, Italy, ...) ? Do you think t...

...I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the Voicebox of John Smith, please leave a message after the tone). Mit freundlichen Gr?ssen Benoit Panizzon -- I m p r o W a r e A G - System Services ____________________________________________________...

...some output on the console, see http://min.us/my-asterisk#3 I am trying to connect to my Asterisk server over UMTS; my provider, Swisscom in Switzerland, says, that they do NOT block port 5060 and/or SIP in general. This seems to be true, since I can connect just fine using this iPhone App to the pbxes.org Asterisk server and also other SIP servers that I tried. Interestingly, when I enter the account details of my (non-working?) Asterisk server in pbxes.org, it can connect to/register with my server just fine; pbxes.org is also running Asterisk. I also had some other guy from IRC trying to regis...

...thing from scratch -- better not answer at all and all that. Well, it may appear that way, but I am not a complete idiot. I know a lot about mobile switching centres, HLRs, VLRs, IN service nodes, mediation devices, billing and settlement systems etc -- I just don't know much about PSTN and PBXes. I would appreciate it if somebody could help me out with a few hints on how to set up my Asterisk box, in particular in respect of VoIP as per my last posting. thank you very much in advance kind regards bk

Hello, I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also another * box connected to A4200). These PBXes have function to assign name to extensions and display it on phone. Asterisk box is connected via PRI with euroISDN signalling (also I have tried QSIG). Is it possible to set callerid with name and display it on alcatel digital phones? With command SetCALLERID I am able only set callerid numb...

Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Thanks! __Yeha...

...zed T1" which cost a lot and did not actually include the "phone" service for any of the channels, that was additional. So the cost to go from POTS lines to something digital was extreme, so much more than I can't understand why anyone would have T1 voice interfaces, yet all the PBXes have this and it seems commonly used. I must be doing this "wrong". Okay, so I need help with: 1. Understanding terminology so I can ask for the "right thing". 2. Advice on when it is reasonable to go POTS versus something else and what that something else is. 3. Feedback...

Hi Well it is well some time that my last DUNDI peer has become unreachable. I guess too many issues with spoofed numbers etc. But I am wondering, do people, especially larger entities like telcos, still use DUNDI? I know that in some Hamradio communities, DUNDI is used to interconnect PBXes, but that is with private phone number ranges, not connected to the public. Want some DUNDI peering? DM me :-) Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29...

Does anyone have a working example of how to use the "switch" directive to peer two Asterisk PBXes? -- - Ian C. Blenke <icblenke@nks.net> (This message bound by the following: http://www.nks.net/email_disclaimer.html)

I need to deploy some quasi-virtual-PBXes, and I'd like to avoid having to be hands on for each new phone number deployed... so I would like to set up some administrative extensions that can record greetings... lets say: [admin] exten => 8(NXXNXXXXXX),1,Record($1|-greeting.gsm) [incoming] exten => _(NXXNXXXXXX),1,Playback($1|-g...

...to:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from app_parkandannounce? I'm trying to use Asterisk app_parkandannouce to build a global parking pool from within a couple of Norstar PBXes. {clip} So, the question becomes: How do I structure my extensions.conf to convince app_parkandannounce to play it's message on the incoming channel rather than using dialback? Have I simply missed a switch somewhere, or will I have to use some sort of meetme bridge configuration to short-ci...

...l - all of our outward telecom is on POTS or Centrex-enabled POTS lines. Site 1 has a Dell PowerEdge 1950 with Asterisk built from source and an AEX804E to connect to the outside world. Site 2 has an Asterisk Appliance with the 4 FXO / 4 FXS configuration, with the FXS ports currently unused. The PBXes at each site are configured to be essentially independent but with a unified dial plan so that calls can be placed or transferred across the VPN with a SIP trunk connecting the two PBXes, and canreinvite=no is set everywhere. The only other "heavy" consumer of bandwidth across the VPN is...

...got asterisk working as a PSTN gateway between two sites. The IP connection is pretty good and bandwidth exceeds 640Kbps all the time. I'm using X100P hardware on 300Mhz and 333MHz systems. One of them has a sound card, but the slower system does not have a sound card. Both are connected to PBXes and I have busydetect=yes on one end because the PBX doesn't drop the voltages at disconnect. The other has a more normal ATA for the analog port. I've searched the FAQ, mailing list archives, and documentation and I don't see this matter addressed. I'd appreciate any help you a...

...ed as a feature on *'s website, nor can I figure out how to make it happen from *'s documentation. To be clear, Asterisk is an amazing product and offers a great many features that I am looking forward to using. I am just trying to figure out when I will be able to use it to replace my PBXes. Thanks in advance. /carmi

Hi List, I was wondering if anybody had tried running Asterisk inside virtualization software such as Xen. Are there known problems doing it? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

...ost a lot and did not actually > include the "phone" service for any of the channels, that was > additional. So the cost to go from POTS lines to something > digital was extreme, so much more than I can't understand why > anyone would have T1 voice interfaces, yet all the PBXes have > this and it seems commonly used. I must be doing this "wrong". > > Okay, so I need help with: > > 1. Understanding terminology so I can ask for the "right thing". > > 2. Advice on when it is reasonable to go POTS versus something > else and wh...