similar to: Another IP address to block

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

-----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

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

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

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 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

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 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.

This has come up before on the list and archives but I don't seem to find a solution for this. On just a few nodes we have this situation where we see the IP disappear from the CLI iax2 show peers list but the status shows OK: 3012/3012 (Unspecified) (D) 255.255.255.255 0 OK (89 ms) How can the status be OK a few milliseconds ago and have no IP ?? The strange thing is

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

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

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

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? --

Hello List, i need to be able to record the call transferred from iax extension to sip extension when i call the sip extension from the IAX extension i can record the call without any issue but when i receive a call from customer in IAX and i transfer this call to SIP client the conversation between customer and IAX client is recorded but the conversation between customer and sip extension is

Hi Specs in subject line: CentOS 6.X all latest patches), iptables 1.47, Apache2.2 I use the Geolite legacy databases together with iptables 1.47 to filter traffic for a variety of ports and only allow .AU traffic to have access. Maxmind (https://dev.maxmind.com/geoip/geoip2/geolite2/) changed the default DB to the latest version which is GeoLite2, this leaves all users in need of the old

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 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 running Asterisk 1.2.13, Zaptel 1.2.11 with a TE110P, a TDM04B and an Astribank-32. They have been complaining that there is echo on calls even when they are IP to IP on the same network. There are 18 Aastra 9133i phones and 30 analog phones connected to the Astribank. I can understand there being a bit of echo on the analog phones, but I do not understand why there would be