similar to: Agents.conf from realtime static

I have a small call center running with Asterisk 1.4.4 and Unicall. Everything seems to be working but twice now we had to reset the server because all lines stopped working. You can see users dialing in and reaching the queue but the agents never get the call and the lines are not released. I saw that there is a new Zaptel driver which fixes a racing condition with a TE110P card which is

Hi iam using 1.2.17 does any one have information meetme in realtime and store in mysql i dont see any document could some one help me is this possible ? ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070606/36d236c2/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am trying to set the CDR(userfield) to a certain vaule using the application map of features.conf but I am not able to do it. When I receive a call I would like to tag it with a client code (3 digit numeric) so I can referenci it later from the CDR. I have edited features.conf with something like: code => #111,self,SET(CDR(userfield(111)) or

I have several installations of Asterisk (several versions) where we have our own web interface that uses Mysql and Realtime. When we do modifications to Mysql we use a Manager connection in order to reload the configuration (we use Realtime static for extensions) sometimes Asterisk will crash. Not every time and not every X times we reload. Sometimes it takes ten reloads and other just one

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come over the other lines. -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director

I am trying to set up Astunicall 1.4.16 with a link from Alestra in Mexico City. I have done everything I usually do for other links in Mexico but this one simply will not send or receive calls. I just get Protocol error. Anyone has any experience with R2 and Alestra? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001

Does anyone know if you can use softphones on thin clients? I have a new customer that wants to use Eyebeam (about 10 users) on a thin client platform. Each user has a little box on their desk that has a USB port, mic and headphone jacks and monitor. I am worried about conflicts when running 10 softphones on the same server since they will all try to use por 5060. -- Telecomunicaciones

Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not

We have a setup running Asterisk interconnected to a Panasonic TDA200. The Asterisk server has a two port E1 card, one connected to the phone company and the other to the Panasonic. Everything is running fine and we can send and receive calls from the Panasonic and phone company. We are using MFC/R2 for both links on Asterisk 1.4.4 and Zaptel 1.4.3. The only detail we have is that we cannot

I am having a bit of a problem implementing the pickup command in my dial plan. I have setup this rule: exten => _*8XXX,1,Pickup(${EXTEN:2}) This works as expected when someone dials an extensions number and I can get the call. The problem I have is that when a call enters my welcome menu and does not press anything there is a timeout that sends them to the recepcionist. The rule is:

I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we

Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

I have a customer that has a Trendnet TEW-435BRM router which has the bad habit of rewriting all external connections so the Asterisk server only sees the IP address of the router itself. Up to today this has not been a problem since all extensions are on the local network but now they want to have a couple external IP phones (SIP). I opened up the ports on the router and my phone can register.

I have a customer that needs an Asterisk server to sell minutes for cell phones in Mexico. I do not see a problem with that since he will get the calls by SIP and then use GSM adapters to get the calls into the GSM network. My problem is that his customers only want to be identified by IP and not by a username and password. Is there a way to authenticate just by using an IP address? --

I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static configuration for extensions.conf will not load. All other realtime configs work (SIP, IAX2, Voicemail). I cannot find any reference or documentation about the structure of the realtime static database for 1.6.1.x but I have used the same table structure since 1.4.x. CREATE TABLE `ast_config` ( `id` int(11) NOT NULL

Hi, Is there a way to know if a member of a queue is currently engaged on a call? Or if a queue can return a busy status if all members are currently engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the scenario only falls into TIMEOUT, and has to finish the assigned number of seconds into the QUEUE CMD before it falls back to the next routine on the dialplan. Any ideas?

I have searched for some time but I have not found an asnwer on how to fix the CDR when a call is transferred. The problem is that if someone dials a cell phone and then transfers the call to another extensi?n the CDR for the cell call stops and there is no way to track that the call was transferred so we can bill correctly. Many people have asked this question but there is no answer, only a

On Mon, Sep 12, 2016 at 3:01 PM, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 9/12/16 3:39 PM, George Joseph wrote: > > > > On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjoseph at digium.com> wrote: > >> >> >> On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cursor at telecomabmex.com> >> wrote: >> >>> Has

I have a big headache. I have an Asterisk server connected to an Avaya PBX. Everything is working between those two. The problem is that I have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the Internet to the Asterisk server through a Fortinet firewall. When calling from a PAP2T I get one way audio, the remote site can hear me but I cannot hear them. If I do an "rtp

Starting with Asterisk 1.2 I have always used realtime static to load my extensions.conf into Asterisk. It worked perfectly up to version 1.6.0.X but starting from 1.6.1.X and upwards it simply does nothing. I can see that the extensions.conf file is mapped to the database: == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': ==