similar to: Who picked up with *8?

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

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

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 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

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 need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? -- Telecomunicaciones Abiertas de M?xico

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 am using Asterisk 1.4.4 on a CentOS 5 machine for a small call center with 6 agents. I am using realtime for queues and sip and I am also trying to use realtime static to load agents.conf. The only problem I am having is that no agents are loaded when I start Asterisk. I have to manually do a "module reload chan_agent.so" so the agents get loaded from the database. Obviously

Is there an archive of security advisories for Asterisk? We recently upgraded a customer from 1.2 to 1.4 and now they are asking for documentation of all security and bug related fixes. I know the advisories get published on this list but is there an easier way to find them than trying to search the list. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de

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 just upgraded a very old Asterisk installation to the last 1.2.31 I can find in Asterisk.org site. Now for some reason my IAX clients cannot connect to the server. I can do a "iax2 show peer iaxmodem1" and I get this: * Name : iaxmodem1 Secret : <Set> Context : oficina Mailbox : Dynamic : Yes Callerid : "" <> Expire

I want to be able to send a prerecorded message to the person I am calling. I know that you can use the application map to do this. Just to test I enabled the testfeature example that is in the features.conf file. When I hit #9 during a call the other user does not hear the monkeys, they only hear a series of beeps. I have tried with different soundfiles and they all give the same problem.

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

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 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 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

I have a customer that has recently upgraded their network and now their Aastra 9133i phones are loosing their connection to the Asterisk server. They were using an external Asterisk server and now we have installed a new internal server with Asterisk 1.4.8 on a SIP/IAX implementation with no Zap cards. Only hard phones seem to be having problems keeping a connection to the server, soft phones

Sometimes I get the following error on the console: [May 31 11:14:01] ERROR[23502]: cdr_addon_mysql.c:230 mysql_log: mysql_cdr: Failed to insert into database: (1062) Duplicate entry '1180628004.3214' for key 1 -- Zap/38-1 answered UniCall/11-1 Why would the UNIQUEID be duplicated? Isn't is supposed to be UNIQUE? This worries me a bit because the customer is losing calls that

I have a problem with a Sangoma card. It worked until yesterday. Now I keep getting this error: Jul 29 17:45:17 pbxacura kernel: wanpipe1: Enable E1 CAS signalling mode! Jul 29 17:45:17 pbxacura kernel: wanpipe1:w1g1: Rx Error: No 'DeviceSelect' from target: pci fatal error! (0x000FA000) Jul 29 17:45:48 pbxacura last message repeated 30946 times It is using Wanpipe 3.5.6 and Zaptel