search for: spa922

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 ca...

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 s...

...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: "SPA922" <sip:5403@mysipserver.com> Call-ID: 9b51726c-3e768481@192.168.0.110 CSeq: 5504 REGISTER Max-Forwards: 70 Contact: "SPA922" <sip:5403@x.x.x.x:1025>;expires=3600 User-Agent: Linksys/SPA942...

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 manu...

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 you suggest a...

Hy all of you, I've successfully installed a freepbx solution with 10 extensions : - 5 on Linksys SPA922 - 1 on Linksys SPA942 - 1 on Thomson ST022 Everything seems to work fine with all the hardphones excepts last week. The thomson has a strange behaviour. It can reach french mobile cell phones but when it reaches "fix" phones, the correspondant can't hear the caller. What is very st...

...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 correctly. The handsets are Linksys SPA922 The issue we are getting is in transferring calls, which happens like this :- 1. Incoming call from pstn/viop provider 2. Call is answered by a user 3. Call needs to be transferred 4. Xfer button is pushed, other user is called, answered, and they speak about the call 4b. The incoming call is hel...

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 (pbx-incoming, 1133, 1) exited non-zero on 'SIP/1210-081aa708' Jun 6 15:08:24 WARNING[406]: pbx.c:1720 pbx_exte...

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 there, I've asked this question to the BSD group too, but I'd like to know whether anybody else had similar experiences on Linux 2.6.20 etc.?? FreeBSD 6.2 Asterisk 1.4.5 (and 1.4.3 from ports) Sip phone - PBX(*) -IAX2-VROUTER(*)- SIP-Voip provider (SPA901 & SPA922 phones) We've see a situation where the IAX2 appears to "loose"/drop the voice data to be sent to the SIP side of things. This happens "semi" intermittently, but we can reliably regenerate it at >40 alaw calls (even on a dedicated 1G network) and also with G729 (but a ta...

Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears MoH. 3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595 hears no ringing When xfer is pressed and the extension is dialled: U 203.89.001.001:5060 -> 121.98.001.001:1034...

Hi all, I’m trying to rewrite Diversion header when call forwarding is done on the phone. The phone sends "302 Moved Temporarily" response and sets Diversion header to a local number, but before Asterisk sends this call towards TSP provider I need to change Diversion header to a full PSTN number. I am using PJSIP_HEADER in a pre-dial handler (configuration is below). On the same