similar to: DID Trunk Lines and Caller ID

Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

Not sure if this is the best place but does any one have any used IAXy's they are interested in selling? I am looking to pick one up cheap for a proof of concept before going all out on them. Also does any one have any real life practical experience with how well (or not so well) that these devices have worked for them? you can reply to me off list at asterisk@matraex.com Thanks Michael

Hello Everyone, I am new to asterisk and linux too, I managed to installed asterisk on redhat8 with the help of mailing list archives and Handbook guide. I configured 2 SIP phones (grandstream) and it is working fine internally. We have T1 Line coming in with block of 200 DID Numbers. I want to assign DID Numbers to each of SIP phones as an extensions and able to call any PSTN line.

Anybody ever run into a case where the Sipura Dial Plan will not work with the S0 option to immediately connect? My Dial plan reads (*xx|[3469]11S0|0|00|[2-9]xxxxxxS0|1xxx[2-9]xxxxxxS0) and I can dial ONLY then numbers in the dial plan so I know that it works. For some reason when I dial 5551212 1212121212 It does not dial for a while and then it dials 555 1212 Anyone have any ideas?

Does anyone know of a resource for extensions in which the server (whether asterisk or custom scripts) can trigger the phone to be answered? So for example an operator can have a headset and when a call comes through the call is automatically (through a script) connected to the headset instead of the operator having to manually answer the call. Any responses, help or ideas of a type of supplier

I have benn having trouble with the Monitor Command. Basically any time that I send a call into a MeetMe room I am only able to monitor half of the conversation. File-in is recorded with the incoming voice but file-out does NOT record anything. I have tried this with both the b and m option as well as without any options to the MeetMe command. Also the Monitor correctly records both sides of the

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

Hi, Is there IAX client in Applet JAVA which can be embeded in a web page ? Best regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031208/c388ef61/attachment.htm

Hello all, I am building a software based on asterisk to handle incoming answering service calls. I have one problem that I have not been able to figure out a reasonably priced solution to: The goal of this software is to allow the agent to be able to do their entire job from the desktop. The only thing that seems to be a problem is getting the operator (agents) headset logged on to the

I currently have Asterisk deployed in my office with a TE411P. On the first port of this card is a T1 from the telco setup for D4 AMI. Unfortunately, I'm not receiving caller ID on inbound calls from this line. The caller ID information is arriving in the form *ANI*DNIS*. In zapata.conf, I have signalling set to em_w. The DNIS always arrives correctly, but I'm never receiving the ANI

Hi, I am trying to find remote call forwarding feature in asterisk. I don't know is it possible or any one had already done it. SBC (local Telco) provide such feature. I can call into my voicemail number, and set the remote-call-forward to my cell or another number. It is like person can remotely manage to set the call-forward or DND to his/her extension. Can this be doable in asterisk?

I am trying to receive both CID and DNIS from the telco through a non-pri T1. Currently I have the T1 setup and operational both outbound and inbound calls are completed as should be expected. The calls came in and were placed in the context specified in zapata.conf on exten => s,1. I have requested that the telco provide callerid (they call it ANI) along with 10 digit dnis for my 800

I am an Asterisk newby, and I cannot seem to get Caller ID information from our T1 line. When calls appear at the phones, they say the call came from "asterisk" and unknown number. I know how Caller ID information is passed on an analog phone line (between the rings) but with a T1 line, I don't know technically how it is done. I don't see the caller's number in the

I am writing a call center application. I do not want to use Queues to manage my incoming calls and connect them to the operators for a few reasons which I wont go into here. The option I come up with is to create a context that the call goes to which runs background() and just loops to play it again and again forever. The background() will have options to dial 1 to leave voicemail Then, when

Currenly we have our Xapian-based search engine search photos from our collection only with the image's name and tags. Thus, I'm wondering that except for these text contents attached to the image, whether Xapian can actually support any kind of image search? Thanks for any suggestion! Best, Shen DISCLAIMER: This message contains confidential information and is intended only for the

Folks, I found free-nx in the repositories and tried to install/use it I could not figure out how to get it working. I tried following directions on the wiki But I still could not get it working Feel free to contact me at stephen.bovy at teradata.com I am currenly using the official ns server on sles 11 sp3 Is their any direction on how to install free nx on sles ??

I have a number of inbound analog lines connecting through Digium cards to an Asterisk box. Asterisk then bridges the calls over to the internal extensions which are all SIP phones. Currenly the inbound lines do not have callerid on them so callerid=no in my zapata.conf file. What happens on inbound calls is that the SIP extensions are dialed but their callerid shows

hello list, i need to setup an asterisk system with 5 ISDN trunks. i found C4 cards but they are very expensive. i found that if i use 5 AVM Fritz! cards it would be very cheap. i want to use 2 boxes. 3 in boxA +2 in boxB =5 isdn. and i want, this two boxs to work as a single box so that one box can share ISDN hardware from other box. this system will be serving a call center. currenly we are

Hello, Currenly I've ASterik@Home 1.5 running on DMZ. I can register SJphone there, good audio on 8200 (webmeet me calls) and i also can dial Zapata extensions. When I dial sip phone extensions nothing happens if the client that i'm calling is registred, if the client has voicemail it goes to voicemail. IMPORTANT: I get this error message on my Check Point Firewall: "sip

Help needed Currenly i'm running samba on a RedHat 7.3 system. The primary domain contreller in the network in a windows 2000 server machine. I'm using the windows 2000 server to authenticate my users in the domain. Each time a new user is created in windows 2000, i have to create the same user in linux inorder to allow the user to have access to the samba shares. Is the re a script or