similar to: SPA3102 PSTN fallback

Displaying 20 results from an estimated 30000 matches similar to: "SPA3102 PSTN fallback"

2013 Nov 05
1
How to enable T.38 between SPA3102 PSTN Line port and ReceiveFAX app ?
Hello, I've got an analog phone which is currently receiving unsollicited FAX calls from PSTN. For learning purpose, I'm preparing an Asterisk/SPA3102 setup that would let voice calls come in and out and translate incoming FAX calls to TIF files (forwarded through email)). My target setup is : PSTN <-- analog--> SPA3102 Line Port <-- SIP --> Asterisk <-- SIP -->
2007 Jul 30
1
Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?
Hi All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone --> Asterisk --> Linksys SPA3102 --> PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use)
2011 Oct 31
1
Calls from PSTN on SPA3102
Hello list, this is my first post on this list. I have a server with Asterisk and a Linksys SPA3102 with 3 SIP phones. I have configured the SPA PSTN line as trunk to receive and send calls. I can call outside from SIP phone throw the PSTN line and all is OK, the problem is when I receive a call from the PSTN, on the out caller phone there is a demo playback. I want to redirect the call to a
2010 Mar 19
0
SPA3102 + asterisk drop call and loop (was SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?) )
Ok, I downgraded spa3102 to 3.3.6. Now when I make a call from pstn and call is established asterisk seems to drop the call. However I still hearing ringback on pstn side, call is established again, and asterisk drops the call again, like a loop. -- Executing [preat_admin at nodo:1] Playback("SIP/PSTN-08214948", "horario-atencion/our-business-hours-are") in new stack
2008 Oct 16
0
Sharing my Asterisk + SPA3102/PAP2 setup: What I've learned in 1 week.
(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to
2008 Feb 27
1
SPA3102 registration problem
Hi list, After failing to get a Sipura/Linksys SPA3000, which I've configured as a PSTN gateway, to pass on the Caller ID, I decided to try my luck with a Linksys SPA3102 after hearing some promising stories. Unfortunately, I've run into a completely new problem: it seems Asterisk won't let this device register. I went about configuring the SPA3102 in much the same way as I
2007 May 08
1
Problems witch SPA3102.
Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. -------------- next part
2010 Jan 22
0
OT - SPA3102 not detecting CID - Which settings to tune ?
Hi, I'm connecting a Linksys SPA3102 to 3 different PSTN analog lines. With only one of those, CID is shown. Beside that, everything is working OK. Lines have different providers and/or locations. All are located in France and CID Detection Method is ETSI FSK / Bell 202. If I'm connecting a TDM400-enabled Asterisk system, to one of those 2 non-working lines : it does work. The only
2012 Mar 10
1
SPA3102 asterisk signaling
Hy all, Recently a have a little problem with a Cisco device, SPA3102. I use this device with asterisk to dial out with outbound trunk. (SPA3102 has 1 FXO port) It working ok , but the device SPA3102 do this : when a call is placed for outgoing in asterisk and send to SPA3102 , this device "answer and dial the number in the same time" , in my CLI I see the channel is open , but on
2010 Mar 18
1
SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)
Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Jul 15
3
surge protector?
I lost one channel on an FXO module on a Sangoma A200 card due to a lightening zap in the area (well - it died the same night as a major thunder storm came through).... Is there a recommended/standard surge protector for phone lines I should be using? My server has 2 POTS lines. thanks Todd
2008 Nov 01
1
SPA3102 interdigit timers bug?
Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW). I have this settings on Voice/Regional: Interdigit Long Timer: 10 Interdigit Short Timer: 3 Anyway, when hooking up (without dialing anything), the timeout starts after 3 seconds. It's like the Long Timer is unused. After dialing, the Short Timer is also used to timeout. Is that normal? Am I missing something? Thanks. --
2009 Mar 04
1
faxing via linksys SPA3102 half page goes through
I'm faxing from stand alone fax machine via linksys SPA3102 but most of the time only half or quarter page goes through. Did anybody have any experience like this? -- #Joseph
2007 Jul 30
0
Questions about SPA3102.
Hello, I got a SPA3102 and everything works fine except calling from voip to phone on fxo port. The phone ring but doesn't get any sound. I connected SPA at my asterisk server and i want to call from asterisk through SPA to fxo port where i have a regular phone. Thank you for support. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Oct 10
0
linksys spa3102 for faxing
Hi, I have been considering a purchase of the linksys spa3102 for a couple hours but I would like to know from someone here, wether this device will support faxing on my local asterisk server, I have had success sending and recieving faces with an x100p, and recall that in the old documentation, they mention that if I send/recieve faxes, that it all should be done on the local server for best
2009 Dec 15
1
OT - SPA3102 - Provisioning with config file
Hello, I could successfully played with General Purpose Parameters (GPP_A, GPP_B) and a TFTP server : whenever I change a GPP value in a configuration file, my SPA3102 automatically updates the corresponding value its web server shows. My config file is : <flat-profile> <GPP_A>myid </GPP_A> <UID1>myid </UID1> </flat-profile> I though I could use this
2010 Mar 01
0
SPA3102 Firmware Upgrade via TFTP fails
Hi everyone, I'm trying to set up a VOIP mass deployment. To do so, I want to generate a configuration xml fails. I read somewhere that I had to use : http://<phone -or- device ip address>/admin/spacfg.xml but it work with an upper firmware only. My Software Version is 3.3.6(GW) The last firmware versin is 5.1.10(GW) Upgrade Enable: is set to yes on the linksys SPA3102 web interface
2013 Dec 23
0
asterisk-gui +spa3102
hello everyone, i am using asterisk 11.6 with asterisk-gui and i am stuck in setting up the linksys spa3102 and the sip trunk for it. may i have some help please? friendly, -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131223/11379631/attachment.html>
2009 Mar 17
3
SPA3102 - How to save config in a file
Hi, I've read in this mailinglist archives some notes related to Linksys SPA3102 provisioning but I couldn't find there the answer I'm looking for. Is it possible with this box (mine is unlocked) to store its config file(s) in a TFTP server, and have this(these) file(s) reloaded at boot time, for instance ? In embedded web server, there is a Provisioning tab full of settings but none
2008 Jul 29
0
Fallback on a fallback
I have two sites running Asterisk PBX. Normally the inbound calls go through a 3rd (colocated) server and are routed via IAX to the site (the site registers with the main server) I created a macro that tries to ring one location and then another. Each site explicitly Answer() the call even though it will only ring all the sip phones at the relevant location. When fall back is in effect it goes to