similar to: PAP2-NA Authentication Issues

hi all, just wondering what's happening here: i have a pap2 and an spa941. everytime i call my spa from my pap2 i can see it being added dynamically on the regcontext: [Oct 7 11:59:08] -- Saved useragent "Linksys/SPA942-5.2.8" for peer 100100 [Oct 7 11:59:08] -- Added extension '100100' priority 1 to sipregcontext but from spa to pap2 i dont see it, i looked

Does anyone know how to set the distinctive ring on the Linksys SPA941? I want to be able to dial one extension and have the phone ring with a certain tone and then dial another and have the phone ring with a different tone. I have tried the following ------------------------------------------------------------------- exten => 802,1,SIPAddHeader(call_info=Classic-4) exten =>

I have been given a Patton SmartNode 4114 and asked to get it working as POPS gateway for our asterisk box. The 4114 has 4 FXO ports. It's got firmware 3.21 on it. I currently have a single POPS line plugged into port 0. I can not seem to get the two to talk together. I am running asterisk 1.2.21.1. I am seeing the following repeatedly in * Jan 24 16:23:40 NOTICE[17063]: chan_sip.c:11291

Dear Group, I have just purchased two Linksys SPA941 and flashed these to the latest firmware. Everything works well except for the Hold button? Has anyone else experienced the same issue? What was the solution? Kind Regards Shad Mortazavi

Hi all, I'm new to this list, so forgive me if I'm not supposed to ask this: I currently own a Linksys SPA941 SIP phone with 5.1.8 firmware. Is there any way to use TLS with this phone<--->asterisk (v 1.6.0.9)? It is said that is supports TLS/SRTP but I don't see any of these options in the configuration file or the admin (advanced) SIP conf panel. Am I missing something? Thnx

I have an a bunch of SPA941 Linksys phones for users in and out of the office. When the phones are in the office (and on the same network as the asterisk server) the WMI goes on when it should and off when it should. But when the phone is on another network and natted it fails to do so. The light always stays off. Has anybody had a similar problem (and hopefully a resolve)?

Hi I have a big problems on my Asterisk systems : I have one Asterisk Server with static IP (no nat) and 6 Linksys SPA941. All SPA are after a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router * SPA-3, we have a pat 5062 => SPA-3 * SPA-4, we have a pat 5063 =>

Has anyone fed a Nortel BCM from Asterisk? I'm interested in switching our company over, but don't want to replace all the handsets in one fell swoop. I imagine some of the PRI cards can "emulate" a switch? I'd still like to pass CallerID into the Nortel, etc but all the external traffic would be VOIP, not TDM.

Hello list, I got a couple of those "wouldn't it be great questions", as following: 1. Is it possible, with asterisk to hold a central phonebook directory of callers?, so that when this party calls a "textual" caller ID will be displayed on the phone display. 2. How can this be configured with Trixbox, I've looked at the configuration options - I assume it plays no

Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)

My server is being attached all day and fail2ban is not stopping the attack. I updated stamstamp to match fail2ban requirements. [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" <sip:7002 at x.x.x.x>' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830

Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941" <sip:4941 at public_ip>' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941"

Hello after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in MySQL DB) register anymore. [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '<sip:testacc77 at 178.19.90.240>' failed for '78.119.140.190:5076' - Wrong password [Aug 15 22:04:13] NOTICE[30098]:

Here is some info that may allow some "locked" PAP2 and PAP2-NA units to be used with Asterisk: I have a PAP2-NA (from a provider other than Vonage) for which I did not know the admin password, though the "user" pages were accessible to me. The provider had set it up to fetch at startup, its configuration file by HTTP from a numeric IP. It was running 2.0.10(LSc). A search

Hi, Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone? I've tried http://[ip address]/admin/spacfg.xml however that file doesn't appear to exist. Thanks.

Hi. We have an odd problem with incoming SIP calls. I have attached a SIP debug log, with some asterisk verbosity as well, demonstrating the problem, below. Is this a known bug? Vital stats: - Asterisk 1.2.3 - Sipura SPA-841, SPA-941 phones - Fedora core 3 The problem manifests itself with these symptoms: - an internal SIP extension receives a call from our PRI - the SIP phone answers the

I'm starting work on some provisioning tools to simplify plugging in and configuring hard SIP handsets and conference bridges (maybe eventually MPEG-4 PoE video cameras that speak SIP as well). Issue is that I'd like to glean as much information out of the configuration files... but don't want to write a whole new parser to do it (especially not one that understands templates and

Hi, I'm experiencing the same problem. We have 2 office locations and the Asterisk server is at one of them. At the other location, all SPA941 access the Asterisk server over an Internet link. All phones are set to "nat=yes" at the remote location. So my problem is that the MWI doesn't work at the remote location. The Sipsak messages are sent properly, but it's sent to the