search for: 9876

Good day all I've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,MeetMe,9876 When I go onto x-lite and type 9876 it gives me this error in asteris *CLI> Apr 29 12:50:12 WARNING[-1221493840]: pbx.c:1198 pbx_extension_helper: No application 'MeetMe' for extension (ext_test_1, 9876, 1) == Sp...

Hi all, I have started the rsync in deamon mode. When I use the following command: rsync --port 9876 u018::treeA it seems to work fine but no files are copied from the server. Using rsync without the deamon works fine. I tried to debug it a little bit, but I am unable to figure out what's wrong. 337:david@u050 /export/treeA> rsync -r -vvv --port 9876 u018::treeA opening tcp connection t...

One problem when migrating from an NT-PDC to a 3.0.4SAMBA-PDC, the user ABC has different SID's. So I want to change the SID of user ABC on the SAMBA-PDC with the profiles-command. Suggest: User ABC in Domain NT has SID S-1-5-21-1234 User ABC in Domain LINUX has SID S-1-5-21-9876 So, on the Samba-Server I did: Profiles -c S-1-5-21-9876 -n S-1-5-21-1234 I get the error message: Could not open (null): Bad address Can somebody tell me how to use the profiles-command ???? Thanks a lot Dirk

...Range Compression (RX/TX): 0.00/0.00 DND: no Echo Cancellation: 128 taps currently OFF Wait for dialtone: 0ms Actual Confinfo: Num/0, Mode/0x0000 Actual Confmute: No Hookstate (FXS only): Offhook Here's a debug from a 911 call. [Jan 31 11:29:53] DEBUG[9876][C-00000005]: pbx.c:4890 pbx_extension_helper: Launching 'Dial' -- Executing [s at InFromSIP:1] Dial("SIP/SIP-00000005", "DAHDI/1/2177772001") in new stack [Jan 31 11:29:53] DEBUG[9876][C-00000005]: sig_analog.c:820 analog_available: analog_available 1 [Jan 31 11:...

I would just like to confirm that this is suppose to work with Ogg Vorbis streams. I'm using 2.2-kh5 at a hosting company. http://servername.com:9876/admin.cgi?pass=mypassword&mode=updinfo&song=a_song_title I'm sure I have the correct servername, port and password, but it's not working. I can do other admin tasks. When I brought this up a few months ago I'm sure someone said it was implemented in v2.2. Thanks, Ross Levi...

Thanks for your reply Karl > It may be that you are just using the wrong url for updating metadata, > http://servername.com:9876/admin/metadata?mount=/stream.ogg&mode=updinfo&song=a_song_title That will be the problem. But how do I specify the userid and password in the URL? Regards, Ross.

Hi. I have the following line in the default context of all my internal extensions: exten => 9876,1,Transfer(125) When I dial extension 9876 from any sip phone, * dutifully transferrs it to extension 125, which is just what I want. Unfortunately when I dial 9786 from my Zap connected analogue phone, the transfer doesn't go through and the dialplan drops through to a hangup. debug outpu...

I'm not going to put this upstream because there's no benefit. However it is useful to record the patch on the mailing list. Rich.

...umber." On the console I get: "unable to open pseudo channel". As indicated in previous posts I do not have any Digium cards in the system. When making the zaptel part of the system I did uncomment ztdummy.o in the MODULES= line. Extensions.conf contains: exten =>2663,1,Meetme,9876 meetme.conf is: [rooms] conf => 9876 If meetme doesn't work at all without a real card that is ok. It can wait. If it'll work at least somewhat with ztdummy then obviously I've missed something. Any ideas? Robert

...with an Eicon Diva Server 2M ISDN card (connected to S0 bus of another PBX). This * box is on a 'live', non-nat IP address. I also have a couple of budgetone phones, one behind NAT and one not. When I place an outgoing call, I get the following messages: -- Executing Dial("SIP/filbert-9876", "CAPI/288:333") in new stack -- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0x5 -- Called 288:333 -- Setting up echo canceller (PLCI=0x201, function=1, options=2, tail=64) > sent FACILITY_REQ (PLCI=0x201) -- CAPI[contr1/288]/0 answered SIP/f...

...t=0 0 m=audio 14384 RTP/AVP 4 0 8 3 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 a=ice-ufrag:6eea7b135aa990d714a67931034ac864 a=ice-pwd:0494fccd28b93f8a2b544b4741203e4f a=candidate:H67c355d6 1 UDP 2130706431 fd31:aeb1:48df::2 14384 typ host a=candidate:Hc1e19c54 1 UDP 2130706431 2607:abcd:9876:5432::247 14384 typ host a=candidate:Hf5a91683 1 UDP 2130706431 2001:1234:5678:abcd::2 14384 typ host a=candidate:Hdccea0f2 1 UDP 2130706431 2001:123:ab:123::2 14384 typ host a=candidate:Hcbb5ed22 1 UDP 2130706431 fe80::21f:c6ff:fec4:926a 14384 typ host a=candidate:H67c355d6 2 UDP 2130706430 fd31:a...

...l imageio server for testing. 1. install ovirt-imageio-daemon 2. configure it for testing # /etc/ovirt-imageio-daemon/daemon.conf [daemon] # directory with this structure # certs/ # vdsmcert.pem # keys # vdsmkey.pem pki_dir = test/pki poll_interval = 0.1 [images] port = 9876 [tickets] socket = /tmp/ovirt-imageio-daemon.sock 3. Now you can run it /usr/bin/ovirt-imageio-daemon To upload, without real engine, you can install your own ticket like this: 1. Create a ticket json: $ cat ticket.json { "uuid": "test", "size&...

On Thu, Jun 14, 2018 at 09:24:48PM +0300, Nir Soffer wrote: > On Thu, Jun 14, 2018 at 9:16 PM Nir Soffer <nirsof@gmail.com> wrote: > > + headers = {"Content-Type": "application/json", > > + "Content-Length", str(len(buf))} There were a few Python syntax errors such as this one. They can be found by running: make -C v2v check

Karl Heyes wrote: > It uses http auth, so you can use > http://user:pass@servername.com:9876/....... Ah, well that's no good then. Windows doesn't support http auth in a URL anymore, and I'm wanting send this from a Windows application. Is there any chance of adding support for something like this... mount=/stream.ogg&user=username&pass=password&mode=... Regard...

...= '61364' BufferSize = '61364' dwBufferLength = '61364' BufferSize = '61364' dwBufferLength - BufferSize = '0' ------------ < 10kb: Opening Audio File 'Steel guitar_4.ogg' vi->channels = '2' vi->rate = '44100' pcmTotal = '9876' pcmTotal * vi->channels * 2 = '39504' BufferSize = '39504' dwBufferLength = '43264' BufferSize = '39504' dwBufferLength - BufferSize = '3760' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/piperma...

...t require a valid remote certificate. * starttls: Use STARTTLS command instead of doing SSL handshake immediately after connected. Can anyone point me in the right direction? Thanks so much! :) -- Monika Janek Systems Administrator, Side Effects Software Toronto, Ontario Canada 416-504-9876 x207 www.sidefx.com

...er option is to start the server from your tests like this. This > is how we run our tests. > > from ovirt_imageio_daemon import server > from ovirt_imageio_daemon import config > > config.daemon.pki_dir = test/pki > config.daemon.poll_interval = 0.1 > config.images.port = 9876 > config.tickets.socket = "/tmp/ovirt-imageio-daemon.sock" > > server.start(config) > > # run your test here... > > server.stop() [...] > However you can run virt-v2v locally against an oVirt instance without > > needing VMware. The command is rather len...

...ing from Cyrus to Dovecot. What would be the most stable version to install on a production system. Would 2.x be alright? I'm feeling lucky. System is Debian Lenny. Thanks for any information! Monika -- Monika Janek Systems Administrator, Side Effects Software Toronto, Ontario Canada 416-504-9876 x207 www.sidefx.com

...il spool? I'm debating between Debian with xfs or FreeBSD with zfs. I'm not sure which way to go. I'm migrating from cyrus. I have about 50 users so it's not a large setup. Cheers! Monika -- Monika Janek Systems Administrator, Side Effects Software Toronto, Ontario Canada 416-504-9876 x207 www.sidefx.com

I have an office up and running with 40 SIP handsets. Currently when an incoming call is parked, they then dial ext 9876 which I have setup to do a page (it does this through an agi script that uses the Polycom autoanswer to page via the phones). What I would like to be able to do is transfer the call to a new extension #699 for example, and have that park the call and then connect the parker to the paging exten...