similar to: Registration problem

Hi, I just purchased a 9331i phone. It's great! However, having a slight issue with line 2 and 3. line 1 works fine. But when I try to make a call with line 2 or 3 I get a fast busy from the phone. Asterisk shows this: Mar 23 15:19:12 NOTICE[18185] chan_sip.c: Failed to authenticate user No User <sip:No%20User@mysipserver.myhost.com:5060>;tag=34c1cd2bf55cb8d Can anyone with a

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'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

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

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

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 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

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

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

Thank you to Rowland, L.P.H. van Belle, the entire samba team, and others that help. I've been running samba in our fire department for over a year. I'm running a self compiled samba version 4.11.9, bind version 9.11.3, BIND9_DLZ, isc-dhcp-server version 4.3.5 with fail over (between the 2 DC's) on Ubuntu 18.04 on Odroid N2 aarch64. I have 2 identical DC's on the network FDDC10

Does anyone knows why in some OpenSSH patches for FIPS we have something like: SSLeay_add_all_algorithms(); if (FIPS_mode() && !FIPSCHECK_verify(NULL, NULL)) { fprintf(stderr, "FIPS integrity verification test failed.\n"); exit(3); } This block of code is always in main() soon after starting service/client. Why are they

David Mosberger <david_mosberger@hp.com> writes: > Hi Peter, > > Hope you don't mind my asking you this directly (I wasn't sure if it > would be appropriate to post this question to R-devel; It would. > if you think it > is, please feel free to forward it). Done. > I'm wondering whether there are any plans to extend R so it could > handle

Hi list, using dovecot in a more complex environment nearly always requires use of ACLs and shared mailboxes/folders. Now, to make life more easy for the users, we can use 'dovecot-shared' files to eable per-user \seen-flagging. Is it possible to make this issue configurable in the dovecot config, something like "always keep a 'dovecot-shared' file in users maildir, even if

hi everybody, got a strange issue with 1.2.4 on SLES11. When listing the user's available subscriptions with any client, every shared mailbox that can be subscribed is listed twice, once with it's correct name, 'folder' for example. the second appereance is without the last letter of the name, 'folde' for example. strange behaviour, since this happens with any client. i

Hi folks, i'm using 3.0.30 on Debian etch, one big share (~250 G) with a separate recycler, built with the vfs object. Files which are deleted are moved to that recycler, now my question: Are files that are copied/moved to another share also moved to the recycler? Thanks in advance, i.A. Hendrik Pahl Systemingenieur -- LWsystems GmbH & Co. KG ++ http://www.lw-systems.de/impressum

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>

I have a CentOS 4.3 box that is giving me some serious problem. During the "yum update" from 4.2 -> 4.3 I lost my connection (okay, I accidentally closed the window). I logged back on and did another yum update, and it appeared as though things completed properly. Everything is working fine, up until one of my hosting clients noted that he is now unable to use ftp to update his web

Hi Al, Here's a set of patches that converts a bunch (but not yet all!) to the new mount API. To this end, it makes the following changes: (1) Provides a convenience member in struct fs_context that is OR'd into sb->s_iflags by sget_fc(). (2) Provides a convenience helper function, vfs_init_pseudo_fs_context(), for doing most of the work in mounting a pseudo filesystem.