similar to: GSM to SIP Adapter

Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet

Hello, a local prison contacted us regarding some calling card solution. they need 4 E1s to serve 120 rooms in that prison. we are planning on using 4 servers to serve the calls and one for the database servers' specifications are: 2.8 Dual Core Proccessors 2 GB Ram 160 Sata Drive each server will be provided with 1 E1 card Questions are: 1- will those servers be able to handle that ammount

Greetings List,Trying to interconnect with a new provider.. the require a?compliance?with?RFC 3261 ?so knowing less than needed about RFC documentations.. i would like to know if Asterisk is actually in compliance with?RFC 3261 or not..?Can any one help with this? Regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308

Greetings, Because of the heavy load and the high expectations of an asterisk server offered as a solution integrated with our CRM software.. we were looking into other possibilities than software Licenses for G729 and G723 codecs.. to lower the pressure on the processor giving it more space to do more work. We heard of a hardware (PCI CARDS) can be used with Asterisk that does the work. And we

Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM "Customer Relations Management" with an SIP dialers built in.the question is the following.. is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will

Greetings List. I'm facing a strange issue with one of my providers.. after sending an INVITE request my server places the call on hold.. until the call is answered.. this is happening only with this provide although i have 3 other providers i route calls through.. can anyone explain what is going on? -- Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562 2308

Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _________________________________________________________________ The

Hello Everyone, We have an order for SMS messaging. Can you gents and ladies be kind enough to disclose if SMS is possible using Asterisk? What is a quick way to test a `Hello World` to my cell. Finally, do all service providers support SMS messaging? Kind Regards, Jayson. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a way skip / ignore the member whose status is busy in the Queue. I have two channel member in queue and i have set the peer limit 2 for these members. I want to skip those member who are currently on the call (answered to calls) and now their status is busy, if Queue see the busy status caller will not enter in the Queue and go to the next priority. [test-queue] strategy = rrmemory

Hi, Looking for some reliable and quality providers of USA DIDs. Any pointers ? Thx Sans -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100623/816aecdd/attachment.htm

Is anybody intimately familiar with the OpenVox G400P card, or the Quectel M20 RF modules fitted to it? I am having a strange network connectivity issue with just such a card, as follows: The card was previously used with four O2 SIMs, and -- once I mastered creating message PDUs! -- worked beautifully, save for the fact that O2's definition of "unlimited" as in text messages

This is the problem: Call coming into a queue in ringall strategy, if a member (SIP) of the queue is busy when entering the queue, and this member comes free after a little time, the member never rings.. How to solve this? I changed all parameters of the queue with no results... Wath i need: If one member of the queue is busy when a new call come in to the queue, this member can hangup and

Hello; I am looking for SIM adaptor to be connected with Asterisk to be able to send and receive calls from the mobile operator and if possible the same adapter to be used for SMS "sending and receiving". But what if anyone called this SIM card that is connected to this adapter and no one relied his call, how this miss call can reach for the use at the asterisk PBX? Regards Bilal

Hello; I need to be able to send SMS messages for campaign or for specific users, also I need to be able to receive SMS messages and do automatic reply. Do I have to use dongle or extra channel? What is the difference? Also, I read that there is SMS through sip, how this work and what is the difference between the sip SMS and gsm sip? If I need to send sip SMS, how destination will receive it?

Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? If I create a queue and make certain SIP-users member of the queue, do these SIP-users always need to log in to the queue to be able to receive calls that are in the queue ?? Can't a member be just available when the phone is registered to the Asterisk-server ? In stead of also having to

Hello, I have added a fade-in/fade-out floating layer to display infos on Ajax request. I have used Position to display it on a corner. Is there a simple way to make this floating layer displayed near the mouse and make it follow the mouse movments Tarek -- Tarek Ziadé | www.afpy.org Python - why settle for snake oil when you can have the whole snake? (Mark Jackson)

Hello, I have a small class where i want to use attributes of the instance when an ajax call returns, so I wrote: *var* MyClass = Class.create(); MyClass.prototype = { initialize: *function*(form_id, rendered_id) { this.form_id = form_id; *var* edit_form = $(form_id); *var* rendered_node = $(rendered_id); *if* (!edit_form || !rendered_node) { this.enabled = *false*;

Dear List, i have been thinking of building a calling cards solution based on Asterisk and a2billing.. i have a few questions regarding this solution and was hoping you may have the answers and could be generous enough to offer them. the servers i'm thinking of are with the following Specs: Processor: Intel X3210 Ram: 8Gb HDD: 2x500 GB Sata Internet Link: 100mbps Dedicated was thinking of

Hi All, I am newbie in this asterisk and a2billing technology . i had successfully installed asterisk in my server fedora -8 [server behind NAT/STUN] i after installation i can able to create users and tested the call features with X-Lite . the was working fine . after i installed the A2Billing in my same server with follow the steps from a2billing installation guide. but u cant access the

Hello, As i check the call logs, some of my clients seem to make successful calls but, in logfiles, Source field seems empty..Still I can see who is the source from Channel tab as SIP/XXXX, and the called number and the time etc but.. nothing on Source and the Called ID tab. Just some clients has this problem. But as i check nothing special in their settings. What might cause this problem. Using