similar to: Thomson ST022 - External Call problems

Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem. For this device (when receiving calls) I get periodic "chan_sip.c set_destination: can't find address for host" I have set the "canreinvite=no" in the sip.conf. Does anyone have a sample entry from sip.conf for the Lynksys SPA 942

Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Pls suggest. cheers /ag -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110212/ccd9d985/attachment.htm>

Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: -- SIP read from x.x.x.x:1024: REGISTER sip:mysipserver.com SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-839856dc From: "SPA922" <sip:5403@mysipserver.com>;tag=685bbad1fae3325do0 To:

Nearly got an SPA922 phone working behind a NAT, the phone registers, and I can dial out and have two way speech, on an incoming call the SPA922 rings I answer and the SPA922 shows "Anwsering" but never does and the far end continues ringing until the voicemail answers, this then show as a disconnected call on the SPA922 I'm on the lastest firmware 6.1.5(a) Thanks in advance

Hi all, I've just bought some SPA922. First time with this hardware for me. I see no LAN tab in its web GUI where I can setup NAT for PC conected to its LAN ethernet port. However, when I connect a PC to that port, SPA922 works as bridge. Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist such LAN tab for setting up parameters as port forwarding? (by the way,

Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do

Hi all, I'm testing Linksys SPA922 phone and I have strange issue. when call is finished on the phone I see "CallEnded" and normal silence for cca. 5 seconds and then I get fast busy for cca. 20 sec. So, this isn't automatic hangup as on other phones I have tried (Cisco 7940, grandstream, XLite,... ) and I have to manually hangup handset to finish a call. Is this normal behavior

Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty

Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short term demo with the customer, but the demo systems has

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11

I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN

I just bought an Aastra 480i CT for a client who needed cordless capabilities in their office. I'm trying to set up the base station and cordless handset in my office first. I'm able to connect the phone to my Asterisk box and make outgoing calls from either the base station or the handset - to extensions within my office as well as numbers outside the network. But I can't

Hello! Sorry for the OT-thread, but i don't know where else too ask... Has anyone done http-provisioning of a Linksys SPA942 with client side ssl-authentication? Where do i get the CA from? I'm aware of the Sipura mass deployment howto on voip-info.org, but it doesn't cover the authentification part. Thanks Christian

Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with "no response to our critical packet". Calls to voicemail and internal extensions work fine. I understand that everything points to a NAT problem, but I don't understand how it could be because: 1) It does not affect

Hi, I am running a Asterisk 1.6 box in our Data Centre, and have a number of users connecting to that box, as their PBX. Calls in and out work fine, as does voicemail. The PBX at the Data Centre has an External IP, Nat?d to it by the firewall, and the relevant ports are open. The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this

I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN number has

I'm getting "Got SIP response 406 "Not Acceptable" back from 10.0.1.2" occasionally when try to dial to SPA942 , anyone has any idea on this before i consider Firmware upgrade? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080326/2fd2c557/attachment.htm

Hi All: I have experienced some big problems on an asterisk production server under 1.2.18: First of all, a very rare message like this... No application Macro ??? -- Saved useragent "Linksys/SPA922-5.1.7" for peer 1363 Jun 6 15:08:24 WARNING[406]: pbx.c:1720 pbx_extension_helper: No application 'Macro' for extension (pbx-incoming, 1133, 1) == Spawn extension

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a

I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from "astlinux" to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming