similar to: SIM adaptor (huwewi or other)

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?

Hi all, Anyone ever used GoIP GSM SIP Gateways ? If yes, what was your experience with those ? I'm looking at this: http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBX&hash=item415d37377c If anyone has any (good) experience with another brand, I'll take the names and models. Thanks

Hi List; Is there an Digium cards support GSM SIM cards so we can fix an SIM card to be used for calls within mobiles as it is less rate? Or I have to use an FXS to SIM adaptor? If yes, then anyone advise a models and prices? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo!

Hi list, I write on the list looking for help, buy a openvox gw gsm for four channels and I'm a little disappointed with the support openvox, for some reason , The call doesn?t get trough support tells me it was my asterisk server, but does not really work me and my internal calls are working perfectly, I tested with another sangoma FXO gateway and works perfectly. the problem is that

Hi All; How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution? Regards Bilal

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

Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW?

Hello, I have a question about "directmedia" or "canreinvite", I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from "sip show settings" that my "directmedia" configuration is applied. Thanks

Hello; I need two scenarios: 1) If someone sent SMS message, then we need to query information from the database based on information sent by the SMS (like the name or the mobile number), after querying from the database, we need to reply by the SMS. Can asterisk do this? To which level? 2) I have vtiger CRM, and it is possible to send SMS (there is a button: send SMS), the question is: what is

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:

Hi All; I need to use a gateway that converts from SIM to E1 to I can send and receive calls via the GSM, so did any one use a good gateway for this and reliable and stable and costly effective, so he can advise us to use it? Also, it will be a separate product if we need also to use it for SMS (send and receive), also we need a reliable and constly effective product for this. Thanks in

Hi list, I need your help ,I have an incoming call x the ivr and the operator takes the call. ext "101" , If a second call reenters and the operator is talking, I want to send to the extension 102 I use the Variable DIALSTATUS , but not working check IVR [IVRINMA] exten => s,1,Wait(1) exten => s,n,Set(CHANNEL(language)=es) same=> n,Set(TIMEOUT(digit)=4) same=>

Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel (one SIM card). any suggestions? Tarek Sawah -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131011/794f5a49/attachment.html>

Hi, I used Asterisk 1.8 and I have a gsm modem with 8 port. When I called target number, gsm modem and asterisk show me one of these ports active. after hangup, the actived port is going to idl status and ready to use. but after some call from extension, when I want to call another number, asterisk gives me Busy status, however all ports are idle and ready to use. I think asterisk have to

I am really a beginner of PRI ISDN board, I am wondering if there is a "quick start" chan_dahdi.conf configuration I could use. I tried to install two "FreePBX" boxes follow the instructions from "http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html" connected them between PRIs, It worked. And now if I refer the FreePBX "chan_dahdi.conf" it looks

Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device

Hi I just bought some digium analog cards and I would like to build an IVR system for my customers. However I am googling and googling , I didn't find any blog and instruction for beginners like me. So I come here for help. Any tips or blogs will help. Regards, Hua Jie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Converting this customer from a MiTel system to asterisk. Discovered that the inbound calls from the T1 are going to extension 366. (This was mapped in the MiTel for some arcane purpose.) The dial plan I am currently using is shown below. When loading the dial plan, I get this warning: WARNING[5004]: pbx_config.c:1561 pbx_load_config: The use of '_.' for an extension is strongly

> Can anyone comment on using SMS in conjunction with VoIP service using > one of these three VoIP providers: voip.ms, vitelity.com, > flowroute.com? Are some SMS services more compatible with Asterisk > (i.e. SMS over SIP works perfectly or not)? Is it best to use a > different data channel for SMS messages (i.e. SMS via HTTP, SMS via > XMPP) instead of Asterisk's built

I recently received a GoIP-32 for a client project -- primarily outbound calling. How should a GoIP be configured for Asterisk? No fancy shmancy Elastix or FPBX GUI -- just using the configuration files. Single Server Mode, Config By Line, and Trunk Gateway Mode all seem likely suspects. How did you configure your GoIP and why? What do your relevant sip.conf section(s) look like? What does