similar to: SIP Channel hangup problem with re-INVITE enabled - ugrent

Hi, I still have the same problem trying to configure ITSP failover in extensions.conf for a connected PRI. Any comments thoughts or direction would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with multiple DIDs we will spread them across two or three ITSPs so that all inbound connectivity will not be lost if one of them has an issue. I

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

I am looking at using R to analyze time series data containing a tidal component. I need to remove the tidal signal to extract the time series of the phenomena I seek to study. A browse of R-project search engines has not been too fruitful? I've found 'hoa' and 'Rwave', but need further help getting started. THANKS. -wa

I wanted to enable SSL on some alternate ports so that a limited number of people could try SSL access. But doing so enabled STARTTLS in IMAP, so that all IMAP users got surprised (at least those whose clients attempted to use it automatically). e.g.: # IP or host address where to listen in for SSL connections. Defaults # to above non-SSL equilevants if not specified. imaps_listen =

If SIP channel driver needs to connect to a remote GW over a dedicated SIP trunk BUT the remote GW has a 'standby' in case of failure, how can the sip configuration file be configured for the remote GW when there are actually two IP addresses. If the main remote GW fails control automatically switches to the standby GW, so how could the SIP configuration file hande this switch and support

Hi all, I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable fixed phone which connects to my Asterisk through PSTN. Currently, I can use my fixed phone to edit and send messages to the Asterisk. However, I cannot make my Asterisk to send messages to the fixed phone. The SMS command displays TX and RX records, hang for a while and then stops with non-zero exits. I read

On Sat, 28 May 2016 16:21:33 -0700, Michael Graczyk <mgraczyk at google.com> wrote : > Hi Marc, Hi Micheal. > On Sat, May 28, 2016 at 10:44 AM, Marc Lavallée <marc at hacklava.net> > wrote: > > I subscribed because your discussion on the IETF draft ("Ambisonics > > in an Ogg Opus Container") was mentioned on the sursound list. > > Thanks for

Hi everybody, I try this : [incoming_from_fxo_card] exten => s,1,Answer() exten => s,2,Background(filename) exten => s,3,Dial($(INTERNAL_SIP_TEL)) But * wait the file is finish before make Dial to SIP channel. Background(filename) (from voip-info.org) => Starts playing a given sound file, but immediately returns, permitting the sound file to play in the background while the next

I wish to know some implementation details about mbox. When you you mbox format in IMAP, are you doing to keep the same message UID across IMAP session if you know that any other program can modify the mailbox as it wants, for example, two messages can be exchanged. In this case, will the UID kept the same ? If they keep their UID, how do you identify the messages ? Thanks, -- DINH V.

hello all of you ! I write you to make you know some bugs. Could you write me back to tell me if it will be corrected ? VFAT ----- when you copy a file that contain accents such as : te?st, with a " on the letter u, the characters that follows the "?" miss all. SMBFS ------ When you list files that contains accents such as : te?st, (as above), there is no more accents, a

I am using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18 Spectralink wireless IP phones. Most of the Spectralink phones have entries in 'sip show channels' that do not go away. None of the other phones do this. Is there anyway to remove these entries without restarting Asterisk? Any ideas on what could be done to prevent this? Example output: xxx.xxx.xxx.xxx 541

When messages marked \Recent are deleted and expunged, the patch now sends the new message count and the new recent messages count (although the message count is not really necessary). >From RFC 3501 errata (ftp://ftp.cac.washington.edu/imap/rfc3501-errata) << Section 6.4.3, page 49: old: The EXPUNGE command permanently removes all messages that have the \Deleted flag set

On Mon, Apr 25, 2016 at 9:32 AM, Timothy B. Terriberry <tterribe at xiph.org> wrote: > Jean-Marc Valin wrote: >> >> Would it make sense to allow an arbitrary number of channels and just >> "truncate" the list of channels. For example, two-channel ambisonics >> would be W plus X and three-channel would be W, X and Y. The idea is >> that you would get

? Is there a way to send an event notification from one Channel Driver to another Channel Driver??If I wanted to have one channel driver do something custom depending on the call state of the other channel driver?involved in the call, how could you send some kind of event or state notification directly to this other channel driver? ? Thanks -------------- next part -------------- An HTML

We have Asterisk1.8.11 and can not move to a newer version right now. But when we run Asterisk as a service, the -r option does not result in giving the CLI prompt? Did the option to get the CLI change? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130722/903d1851/attachment.htm>

Just found out about Dovecot and it seems to work great. My problem is that it compiles and runs on Redhat just fine. On the other hand it compiles on suse and solaris but will not work. When I start dovecot it silently stops running. No entries in the log or anything. Are there any specific instructions for compiling it on solaris x86 using gcc. Thanks

Hello, According to our experiences, most recent dovecot incompatible with eudora pop3 client - eudora (5.1, 6), pop3, 'leave on server' enabled: the clients receive the full inbox *every time* they check the inbox on the server. [Users using other MUAs experienced the same once, at switching from uw-pop3d/imapd to dovecot.] That is absolutely devastating, especially for home users

Dear Dovecot developers, I have several question about imap client: - Is there any difference if we want to create imap client that connect to Imap server with user mail format Mailbox or Maildir? Dovecot support both of them, are we need to create different imap client to support both of them? - Does anyone have imap client library other than UW's c-client? thanks for your

When trying to update (various) packages using update.packages() I get the following error message for various packages package 'fBasics' successfully unpacked and MD5 sums checked Error in unpackPkg(foundpkgs[okp, 2L], foundpkgs[okp, 1L], lib) : malformed bundle DESCRIPTION file, no Contains field > This happens with other packages besides fBasics (Matrix, as well as others) and I

If I have Asterisk setup with local SIP phones configured but need to call a SIP phone which is not local but actually on another VLAN, what would the dialplan need to look like? ? Could the Asterisk dialplan directly call a SIP phone which is not a local phone within its sip.conf and dialplan, if the Directory Number and IP is known (or host name)? ? Didn't plan on needing a SIP trunk so