similar to: Recommendation for one chip GSM gateway --> Yeastar vs. Dinstar

2014-11-12 2:45 GMT-02:00 Luis Eduardo Cortes <luedcortes at gmail.com>: > Hello: > > I'm newbie in asterisk, please help me. > > My context is as follows: > > 192.168.4.2 --> Asterisk 11.13.1 complied from source > > 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway > > When I call from a GSM cell phone, my TG100 GSM gateway answers and > dials

Hello: I'm newbie in asterisk, please help me. My context is as follows: 192.168.4.2 --> Asterisk 11.13.1 complied from source 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway When I call from a GSM cell phone, my TG100 GSM gateway answers and dials extension 7777 (configured as a hotline on TG100) to asterisk server, but asterisk server sends me "SIP/2.0 401

Hello, I have a weird problem between Asterisk 13.3 and a Yeastar U200 pbx over IAX trunk. Should I call from Yeastar to my asterisk 13.3 the call goes through without issues. Should I call from asterisk 13.3 to Yeastar I can hear a ring tone however the yeastar does not show any activities. On the yeastar I initiated a debug command iax2 set debug peer "my trunk name" While I

I am in the process of buying a TDM800 card from Yeastar ( http://www.yeastar.com/products.asp?TypeName=TDM800%20PCI%20Card&cTypeName=1 ) Any one has tested this cards? How reliable are them? I am specially interested in the FXO/FXS module. -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ ------------ It's most certainly GNU/Linux,

Hello: I have this situation: I can make calls internally, I can make inbound calls but I can't make outbound calls. Thanks in advance. These are my devices: * asterisk 11.8.1 = 192.168.1.22 * sipphone grandstream gxp2160 = 192.168.1.5 * gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4 port 1 (FXS) connected to an analog phone port 3 (FXO) connected to the PSTN These are my

Hello: Is that possible? Do I have to upgrade first to firmware version 6.4? Does firmware version 6.4 exist? Thanks in advance. Regards. -- Usuario Linux Registrado # 342019 --> http://linuxcounter.net/ <-- skype --> luedcortes gtalk --> luedcortes at gmail.com msn --> luedcortes at gmail.com

Dear all, I've read some news about Sisky (http://www.yeastar.com/Products/SiSkyEE.asp), a service to interconnect Skype clients with SIP clients. Does anybody test Sisky and can tell me about his experience ??? (Sisky runs on Windows because Skype and its API are more stable on this OS). Regards, Alejandro

Hi list, I have a request to integrate Iridium in a Asterisk system. A quick search didn't return much: I expected to find products similar to GSM gateways, but this does not seem to exist. so I'd be very interested about possible solutions. Has it be done already, how? Thanks, -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise

Thanks for reply, but this is irrelevant, I'm looking for an *Iridium* gateway. Regards, -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise https://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.797.527 Le 03/04/2018 ? 16:05, albert zhang a ?crit?: > http://www.dinstar.cn/en/index.php/GSM/ > > 2018-04-04 10:01 GMT+08:00 Jean-Denis

People, I'm trying to force dahdi to load the modules I need to get spans working. I have two cards, an Digium TE210E (PCI-e) and a Yeastar TDM1600 FXO (PCI) Actually it is loading just the first of them, it is (wtc4xxp) for TE210E, but doesn't load the second module as specified at /etc/dahdi/modules: wct4xxp ystdm16xx I have to load it manually to get spans working. modprobe

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially.

Has anybody ever integrated Skype with Asterisk? If you have, which software would you recommend to accomplish such a task? ChanSkype? And how reliable are the calls? Did the DTMF tones work? Thanks in advance. _________________________________________________________________ Discover sweet stuff waiting for you at the Messenger Cafe.? Claim your treat today!

Panasonic PBX KX-TDA600 it doesn't support SIP protocol for VoIP technology it only support H323 Trunk through 4 or 16 channels gateway card and TDM technology with ISDN BRI and PRI card. Mc GRATH Ricardo

Hello all! Does anyone have a good howto to setup Asterisk and Skype. Thanks Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080526/831b3824/attachment.htm

Am 24.06.2020 05:05, schrieb Michael Maier: Hi > Your basic architecture looks good to me - now you have to start the Nice to hear it... > analysis of the problem with pcapsipdump and wireshark as I wrote > before to get an idea what actually happens at > all. You most probably won't come any further without doing any > analyzing. You have to learn it. It will take some, or

What would be the best way to solve this problem? Anyone else that have got the same problem with Android’s native SIP client, especially on Samsung phones? I do not know if the bug is in Android native SIP, or Samsung’s build of the SIP client, or if the bug is even with the OpenVPN client, or where the bug actually is. The ACK might even be sent for real, but have the incorrect source IP so

I know it's easy to write a simple loop to do this, but in the spirit of lapply, I thought I would ask if there is a builtin to filter or take a subset of a list based on a predicate in a similar way to the Erlang lists:filter/2 function: http://www.erlang.org/doc/man/lists.html#filter-2 filter(Pred, List1) -> List2 Types: Pred = fun(Elem) -> bool() Elem = term() List1 = List2 =