similar to: Asterisk and SER and Asterisks Queues

Waldo, How do you let your customers manage 'their' PBX. I too have a setup like you. However, I installed a * server for each customer, via vserver. I'd like to now what kind of software/webbased package you use for this. I also have SER installed as a front-end server for the * servers. But, as I'm still not very into SER, don't know exactly how this fits in. Should I use

lqbal, I do plan on having alot of users. Two markets I'm trying to get some volume users from are: residential consumers and business users. Residential consumers should get basic line services such as their own DID, voicemail, caller-id, call-waiting, three-way calling, and basically, all the standard features you get from companies like Vonage, etc. This particular market base

I'm wondering if anyone can recommend one over the other. I'm mostly interested in running open source solutions, so I would prefer if your recommendations are within the open source arena. Basically, I contemplated the idea of using SER as a NAT Helper and possibly as a SIP server for a portion of our user base. We prefer to have Asterisk in the mix because of the additional

I was playing today with the different queueing strategies in queues.conf when I noticed the following behavior. I have 4 agents defined in a queue in queues.conf. These agents login using AgentCallbackLogin. The strategy in the queue is set to leastrecent. I place four calls into the queue and * sends only one call to the least recently used agent. If that agent does not pick up, the

I've been reading a bit about Enum and DUNDi and still have something not very clear to me. This is a HYPOTHETICAL scenario: I have 4 asterisk servers. All of them are handling registrations of both SIP and IAX2 UAs. SIP agents are being load balanced by something like SER. I have another server in charge of load balancing IAX2 UAs registration (some sort of dynamic firewall telling

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in

Hi all, Assuming 1.0.7 is the latest stable version, how/where can I find out the different CVS revisions available and a description of what has been patched/updated in each CVS revision so I can decide whether to leave my 1.0.7 installation as is, or if I need (or think I need) to patch it with a CVS version? Thanks, Waldo

Hey gang, I'm trying to work out all possible scenarios using SER & Asterisk in our upcomming deployment. The example scenario is 50 different customers, all with different numbers of SIP UAs. All UAs would register with SER; This will help keep any inter-office conversations off our bandwidth since SER doesn't handle the RTP stream. Calls from PSTN to UA are easy to handle.

I'm looking for an inexpensive way to connect 20 analog phones to asterisk. I could get a bunch of Linksys or Sipura boxes but was wondering if there is a more cost effective way? I came across the Mediatrix 1104 and even the Mediatrix 1124 but that comes out to be almost $100/port. I might as well buy inexpensive IP phone. Does anyone have any suggestions? Thanks, Waldo

Hello i am trying to use this exmple with SER-0.9.3 but still NATED Clients are not working any other requirement http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper ----------------------------------------------------------- # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters

Hi, we a have a SER (OpenSER) in front of 2 real-time Asterisk. SER simply forward SIP messages to 1 of the Asterisks: UA --> SER --> Asterisk We have a problem with REGISTERs: Asterisk answers with 200 OK, but changes the Contact header, inserting the IP of SER instead of the original IP (the IP of the UA). It seems that performs a sort of NAT-traversal, but all the elements are on

Hi guys, I'm testing the sending of a URL to an XLite softphone when a call is in queue. See the output of the CLI below: -- Executing Queue("Zap/69-1", "q_sample|tT|http:// www.google.com/") in new stack -- Started music on hold, class 'default', on Zap/69-1 -- outgoing agentcall, to agent '1000', on 'Local/ 1000@agents-1b94,1'

Hi list (again), I have another question which I have not been able to resolve from neither the wiki nor Google. I've been able to configure a multi-tenant setup of asterisk for 2 small call centers with no problem, by simply playing with contexts (which I guess is how everyone else is doing it). The problem I have is that I've only been able to configure one global agents.conf

I have two asterisk machines. One of them has a Digium board (server A) and the other is simply using ztdummy (server B). Server A is running on Debian and Server B is running Gentoo. Server A is running Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running Asterisk 1.0.7. The problem I have is that when I try to transfer a call into a meetme room in server B, it simply hangs

I was reading on the wiki that Asterisk runs very solid on NetBSD. Can anyone comment? What is the definition of solid? Who is running Asterisk on NetBSD and which version of Asterisk are you running? Also, I know there is limited support for Digium cards on NetBSD, but is there any support at all? Would a TE410P work in NetBSD? I want to build a very simple VoIP to TDM gateway. My idea

I was reading on the wiki different possibilities of automatically restarting asterisk every so often. In some places, people mention they restart it once a day other on shorter or longer intervals. I believe the main reason people are doing this is because of possible memory leaks. I'm running a system for IVR services. It's not a heavily loaded box, but there is almost always

Hi all, Hope somebody can help-I really am stumped as to why this won't work. I realise that this isnt an Asterisk problem (Please dont bash me on the list) and I have emailed the SER list but I havent received a reply and maybe someone on this list can help...Once this problem is solved I am going to use Asterisk for voicemail etc with SER (I have it set up) I currently have SER set up and

We use SER to front several Asterisk systems. Phones register on SER, which also acts as a load balancing and failover proxy for the Asterisks. Phone account details are held in MySQL, which Asterisk could access but does not currently do so. At present, call routing is done on the Asterisks using a FastAGI program which does the database access. We've been asked to implement busy lamps.

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've