similar to: Asterisk GOIP Outgoing Callerid not working

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

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, Is there any function to replace colProds that finds column-wise products of a matrix? Or is there any other function that would give a better solution this problem ? I have a matrix and an array, example: > a GSM1.CEL GSM2.CEL GSM1.CEL 10000_at 1 3 1 10001_at 3 3 3 > b 10000_at

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

Hi everyone, I would realy appreciate your help and guidense with a problem I have. Im trying to set up a SMS Gateway as a alert system for my network. Although the Aircard is picked up as a usb device and has a driver in the kernel, it does not have a /dev/XXX device name. So my question is how do I create this /dev/XXX device and make it persistent. I have installed CentOS as my base with

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. 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

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

I'ld like to test SMSes between Asterisk and an analog phone on a GS-ATA-286 (SIP). I have spent many hours trying any exemple on voip-info or mailing-list, but no message got sent from the analog phone. My goal, at now, is to send SMSes locally from and to Asterisk. Does anyone have a working configuration ?

hi is it possible to use asterisk as an sms central to send SMSes directly to clients on PSTN instead of just communicating with a central? the telco to which we're currently connected doesn't have a central.... roy

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 all, i today have searched on the internet about a solution to let asterisk act as a SMSC (Short MessageServiceCentre) to deliver SMSes directly to GSM Phones. I only have found some cases with use of an extern SMSC (i.e. by the Mobile Net Provider) Is there a possibillity to do that, or ist asterisk only able to send SMS to analog phones? Do you have any idea or hint for me? Thanks

Hello *, I recently started playing around with the "SMS" application. Several of my SIP clients are FritzBoxes, with SMS capable DECT phones connected via ISDN or analog line. So far, SMSing works great: I defined extension 0193010[01] to receive SMSes, which works well with the default settings for German phones. Incoming SMS in the morx queue will be parsed by a cronjob and

Interesting approach. What we've done is to write an app that runs on a separate machine that simply does some asterisk -rx calls to the running Asterisk instance via an SSH library and then evaluate the string returned. For example, to monitor our registered SIP service providers, we compare the output for the sip show registry CLI command when the system is in the correct state, to the

My recent migration from Courier-IMAP to Dovecot has been marred by these errors: Oct 24 12:12:40 server dovecot: IMAP(juha): Fixed index file ~user/Maildir/.Geekzone/dovecot.index: first_recent_uid_lowwater 431 -> 430 Oct 24 12:12:40 server dovecot: IMAP(juha): Corrupted index file ~user/Maildir/.Geekzone/dovecot.index: Duplicate header extension keywords Oct 24 12:12:40 server dovecot: child

Hey, all. I'm trying to set my CID such that, internally, I see a four-digit extension (which is also handy when checking VM), but externally, I see the full 10-digit number. So I plugged these lines into my extensions.conf: exten => _XXXXXXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2) exten => _XXXXXXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1}) exten =>

> On Sun, 15 Nov 1998, Jan Kratochvil wrote: > Hmmm... are you saying the Ethernet driver was broken on the Windows > machine, or under Linux? I assume the DOS/Win machine.... I may look into No - Linux. :-) Becker's tulip.c was the one distributed with Linux kernel (not the one from Becker's web) and it was dropping packets massively probably due to missing PCI kludge

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this

Hi list, why I cannot hear voice when I call from a SIP telephone (Budgetone and others) to a H323 telephone (several models)? could anybody please give any idea to solve this issue? Please, let me know. Thanks in Advance. N.B. The configuration for "extensions.conf", "sip.conf" and "h323.conf" files are: ***************************************