similar to: Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?

Displaying 20 results from an estimated 4000 matches similar to: "Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?"

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 -->
2008 Aug 20
2
Linksys SPA3102-NA firmware upgrade on Linux
Does anybody know if the process of upgrading firmware on "Linksys SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org http://www.voip-info.org/wiki/view/Sipura -- #Joseph GPG KeyID: ED0E1FB7
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
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
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
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 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 Mar 30
0
SPA3102 PSTN fallback
Hi - I got a SPA3102. I've set it up without to many problems. If the unit looses power, calls to the PSTN are bridged which is nice. However, if the Asterisk server is unavailable (I turned it off to test), calls out are not bridged to the PSTN. I've rebooted the SPA3102 with the asterisk server off, but still it gives me no dial- tone. Under the configuration, Auto PSTN
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
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 Nov 04
2
Cisco SPA3102 Thoughts & Other Recommendations
I'm looking to build a VoIP solution for 100+ service vehicles that have WiFi hot spots installed (with cellular uplinks). Currently we are trying out Skype wireless handselts and Majick Jack. I'd also like to consider an Open Source solution that can bring the calls back to our data center [possibly integrated without our existing BCM 3.x VoIP PBX]. For hardware someone on the IRC
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:
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. --
2008 Jul 11
1
Sipura 3000 replacement ---> SPA3102 how reliable is it?
I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? -- #Joseph GPG KeyID: ED0E1FB7
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 Mar 05
1
Linksys SPA devices and CID
Hi list, After successfully configuring Linksys SPA3000 and SPA3102 devices as Asterisk PSTN gateways, the only thing I can't get working is the PSTN Caller ID. The analog and SIP phones I've used can both display CIDs for internal calls, while the analog model also displays CIDs correctly when attached directly to the PSTN line. However, when PSTN calls come in via the SPA
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
2007 Dec 30
2
asterisk callerid
I'm missing something simple I think: I have an spa3102 for which I want asterisk to use the incoming pstn callerid when it sends the call to a local extension (207). callerid works fine for the internal phones (between each other) The spa3102 is picking up the PSTN callerid and displays it in its own status pages Asterisk however, doesnt see the callerid at all. The spa3102 is set to:
2007 Aug 07
2
Outbound dialing
Hello all. I am just getting back into Asterisk and I am setting up my Linksys SPA3102. I have incoming calls working fine, as is the phone plugged into the unit. My problem is I cannot get the SPA3102 to dial a phone number automatically. I can call the extention of the PSTN and I get a second dialtone, and I can then manually dial. I'd like to be able to have Asterisk pass the