similar to: SIP REFER method.

Hi. It is propbably a really naive problem, but I really couldn't find answer how to connect two Astrisks via SIP. I managed to do it via IAX without any problem. But this is a test installation and I would like to connect them via SIP. So I have two computers: pbx1 - 10.1.3.207 pbx2 - 10.1.3.204 pbx1 handles extensions 1xx and pbx2 for extensions 2xx. I would like to call user from pbx2 to

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

Hi ML! Please look at this: - I've got no lucene-index: > # ls /dane/domeny/mejor.pl/mail/marcin/.maildir/lucene-indexes > ls: cannot access /dane/domeny/mejor.pl/mail/marcin/.maildir/lucene-indexes: No such file or directory - run search in body: > echo -e "a select bugzilla\na search body flto\na logout\n" | /usr/libexec/dovecot/imap -u marcin at mejor.pl > * PREAUTH

Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user

Hello, Let SIP extensions 1001 and 1002 belong to an Asterisk calling group : whenever an coming call reaches this calling group, both extensions 1001 and 1002 receive a SIP INVITE message which makes these 2 phones starting to ring. When a callee picks up his phone, the other extension receives a CANCEL or BYE message which stops ringing. Is there any option you can include in CANCEL or BYE

Hi, I am clustering objects using the agnes() function and the UPGMA clustering method (function = "average"). Everything works well, but apparently something is wrong with the dendrogram. For example: x<-c(102,102.1,112.5,113,100.3,108.2,101.1,104,105.5,106.3) y<-c(110,111,110.2,112.1,119.5,122.1,102,112,112.5,115) xy<-cbind(x,y) library(cluster) UPGMA.orig<-agnes(x)

Hi people, I have a few SPA-942 around, all of them work fine except one. The one behind NAT.. In every phone you can: * Pickup a Call on one of the line buttons, * Create a new call on another button * Press "xferLx" to join those to calls. This works everywhere except on the one behind NAT. After a lot of messing around with all the options possible I gave up and subscribed

Hi to all excuse me but i don't understand what is the correct configuration needed to setup a transfer with REFER in SIP. I've tried the transfer() application, but i've experienced some problem, i can't reproduce the error in a clear debug environment but randomly the call crash before to be transferred to the final peer. on the wiki

On Wed, Feb 17, 2010 at 10:42:43AM +0100, Daniel Mack wrote: > Fix the following compile time error: > > drivers/built-in.o: In function `nouveau_connector_detect': > /home/daniel/src/linux/jup/linux-2.6/drivers/gpu/drm/nouveau/nouveau_connector.c:243: undefined reference to `acpi_lid_open' > > Signed-off-by: Daniel Mack <daniel at caiaq.de> > Cc: David Airlie

Signed-off-by: Marcin Ko?cielnicki <koriakin at 0x04.net> --- drivers/gpu/drm/nouveau/nv04_graph.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/gpu/drm/nouveau/nv04_graph.c b/drivers/gpu/drm/nouveau/nv04_graph.c index 396ee92..d561d77 100644 --- a/drivers/gpu/drm/nouveau/nv04_graph.c +++ b/drivers/gpu/drm/nouveau/nv04_graph.c @@ -543,7 +543,7 @@

Hi, I have been using dovecot 1.0 beta 8 on Debian Sarge for a couple of days and I experienced some problems when removing folders.. As an IMAP client I use Thunderbird 1.5 and IMAP folders are on NFS. Here are the steps to re-produce the problem: 1. Run Thunderbird as usual 2. Create folder 3. Remove folder - In my case folder is not removed but moved to Trash (Thunderbird setting) 4. I go to

I am having issues with transfers (SIP/REFER) using Asterisk 1.6. You will find the SIP debug below. There are three phones in this setup. 5253 and 5258 are Aastra 53i telephones, 101 is a standard phone connected through an Audiocodes gateway. All phones are registered in context "phones" and are set to not allow reinvites. All phones can dial each other directly. The dialplan

I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Here is sip debug from pressing "transfer->blind->dial dest->Dial Key" (note both

Hi, I thought it would be fun to try to write a render method that uses erb for rending. I''m not much of a programmer, but I still like to try: module Test require ''erb'' def render(m) ERB.new(IO.read("templates/layout.html")).result(binding) do ERB.new(IO.read("templates/#{m}.html")).result(binding) end end end This

Hi I need to know if Asterisk supports the full features of the SIP REFER message (i.e blind and supervised transfers). I'm trying to do a supervised transfer through Asterisk from a VoiceXML application using the <transfer> tag and setting bridge="true" (i.e <transfer name="transfer1" bridge="true" connecttimeout="10s"> ) but as soon as

Hello! I'm trying to setup additional namespace. I'd like to have this setting per user and keep it in postgresql. Global namespace is defined in this way: namespace { inbox = yes location = prefix = separator = . } (mail_location is getting from db) I don't know how should looks like database field:( Now i have such configuration: user_query = select .... dovecot_location AS

Hi! I'm using dovecot 2.1.12-r1. When remote imap server rejects connections then I can't login into account and MTA can't deliver emails to maildir also. I'm attaching snippets of logs: 2012-12-15T18:05:43.072594+01:00 meteor dovecot: imap-login: Login: user=<marcin at mejor.pl>, method=PLAIN, rip=2001:470:1f0b:1ab3:50af:bef4:6918:78ad, lip=2001:470:1f15:1b61::2,

Hi to all, I am sending a SIP REFER message to Asterisk from a VoiceXML application using the <Transfer> element to do a Transfer through Asterisk. I need to know if Asterisk supports the full features of the SIP REFER message because if i set 'bridge=true' in the <transfer> element of the VoiceXML application to supervise the call, Asterisk sends a NOTIFY message with

Hi! I'm not sure is it the same as described by Andrew Lewis, this is whay i'm posting. Log is: 2011-03-31T17:40:08.369598+02:00 hermes dovecot: imap-login: Login: user=<marcin at mejor.pl>, method=PLAIN, rip=62.121.127.119, lip=193.238.12.139, mpid=19221, TLS 2011-03-31T17:40:08.684562+02:00 hermes dovecot: imap(marcin at mejor.pl): Error: Corrupted dbox file

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Date: Wed, 19 Aug 1998 23:30:58 +0200 (EEST) From: Marcin Bohosiewicz <marcus@venus.wis.pk.edu.pl> To: redhat-announce-list@redhat.com Subject: screen-3.7.4 (security update) Message-ID: <Pine.LNX.3.96.980819232924.15924A-100000@venus.wis.pk.edu.pl> I updated my package after BUGTRAQ fix for tmp-races in screen package. I built