similar to: GSM Sip Gateway

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

Hello I have a simple 1 channel goip gateway (http://www.voip-info.org/wiki/view/GoIP). The incoming and outgoing calls work with Asterisk except the caller ID for the outgoing calls. I think I have exhausted all possible options regarding setting a caller ID and it still doesn't work. The recipients will get "private number". The incomings caller ids are work just fine. exten

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

05.10.2015 23:24, Joshua Colp ?????: > On 15-10-05 05:22 PM, Dmitriy Serov wrote: >> Hello. Do I understand correctly that the current implementation >> res_pjsip does not support ZRTP? >> http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html > > ZRTP is not supported in Asterisk itself. > >> Nothing has changed since 2013? P.S. I greatly

Hi, I'm running into a strange problem with commas is variables. I have the following contexts: [messages] exten => _+.,1,Noop(External SMS) same => n,Set(ACTUALTO=${CUT(CUT(MESSAGE(to),@,1),:,2)}) same => n,Macro(goip_sendsms,${ACTUALTO},"${MESSAGE(body)}") same => n,Hangup() [macro-goip_sendsms] ;Call Macro(goip_sendsms,number,message) exten => s,1,Noop(SMS

Greetings all, After a long day of fighting with GTalk and having it finally working, I wanted to setup DPMA on my Digium phone. So first of all, I had to reinstall it all and reconfigure it all, since it works only on certified versions, and my installation was not from the certified branch. It took a long time of recompiling, testing, adding missing stuff, but I got it straight. Now, I

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

I have a loop that I iterate through in my view. <% coll.each do |itm| puts(itm.value) end %> How do I render the itm.value without using a <%= %> tag? Should puts not work? Joerg P.S. There is a good enough reason for me wanting to do this :-) -- Posted via http://www.ruby-forum.com/.

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

06.10.2015 1:22, Joshua Colp ?????: > On 15-10-05 05:58 PM, Dmitriy Serov wrote: >> 05.10.2015 23:24, Joshua Colp ?????: >>> On 15-10-05 05:22 PM, Dmitriy Serov wrote: >>>> Hello. Do I understand correctly that the current implementation >>>> res_pjsip does not support ZRTP? >>>>

Hello All, The problem I'm facing is the following: two machines that require analog signal. However, connection to the world is setup as follow: T1 connection to Asterisk server Any suggestions on how to convert digital signal to analog? Type of card on my Asterisk server is wct4xxp Thanks for your support! Motty -------------- next part -------------- An HTML attachment was

Thanks, Nux!. But a few hours late. I just purchased: http://www.ebay.com/itm/ZOTAC-ZBOX-NANO-Plus-Mini-PC-ZBOXNANO-AD12-PLUS-2GB-320GB-with-Power-Supply-56/382042194064?_trksid=p2045573.c100033.m2042&_trkparms=aid%3D111001%26algo%3DREC.SEED%26ao%3D1%26asc%3D41376%26meid%3Deae770f22d504a9b8366eb0c02dd20d6%26pid%3D100033%26rk%3D7%26rkt%3D8%26sd%3D152356229748 Zotac has been in the mini/nano

hi, i am trying to run an app written in visual foxpro. i am a real n00b when it comes to wine... here are the errors i get when trying to run it ---8<-----8<------ $ wine itm/itm.exe fixme:ole:CoRegisterMessageFilter stub err:ole:CoGetClassObject class {854d7ac5-bc3d-11d0-b421-00a0c90f9dc4} not registered err:ole:CoGetClassObject class {854d7ac5-bc3d-11d0-b421-00a0c90f9dc4} not registered

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

Hello. Do I understand correctly that the current implementation res_pjsip does not support ZRTP? http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html Nothing has changed since 2013? P.S. I greatly regret that moved from chan_sip to res_pjsip. Previously used very much lacking, and much of the promise failed. Dmitriy Serov. -------------- next part -------------- An HTML

Dear R users! I want to process a maximum likelihood estimation for a parametric regression survival time model with multiple events per subject. the STATA command for this survival regression is: use survreg stset failure(exercise), id(optionid) local regressors itm posret negret streg `regressors', distribution(weibull) explanation: stset declares data to be survival-time data; exercise

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

Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100<sip:100 at 108.161.145.18>;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? Thank you

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:

This patch set is intended for the next merge window. They are just enhancements of the already merged patches or ia64 porting from x86 paravirt techniques and that their quality is enough for merge. This patch set is for binary patch optimization for paravirt_ops. The binary patch optimization is important on native case because the paravirt_ops overhead can be reduced by converting indirect