similar to: s4 built in sip client and 481 call/transation does not exist error

Hi i have recently moved a mailbox to dovecot and intend to use it via imap and i have run into a problem. when i imported the mails apparently the UID order got reversed compared to the date order, result is that some clients like on android devices download the oldest mails instead of newest. it is not so interesting to have several years old mail show up at the top of the mailbox. is

I am sending this problem for 2nd time. Please help. Thanks _____ From: Omar McKenzie [mailto:omckenzie@trenetinc.com] Sent: Thursday, September 08, 2005 9:57 AM To: 'asterisk-users@lists.digium.com' Subject: Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist" I am not able to get softphone registered (active) with * . new installation , new user

I am not able to get softphone registered (active) with * . new installation , new user Able to get server started , and phone appears to register . gets the SIP reponse 481 message Register SIP '4009' at 192.168.200.10 port 2199 expires 120 Unregistered SIP '4009' Register SIP '4009' at 192.168.200.10 port 9428 expires 120 Saved useragent

One of our remote user's phone reports frequently: Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from <IP> What can I do ??? bye Ronald

I have a PCPHONELINE SIS phone set it up to asterisk Registered SIP '205' at 24.172.221.22 port 2770 expires 120 (Port changes every time) Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 24.172.221.22(24.172.221.22 is my phones IP) Anyone have an idea what the problem is? Jeff

Asterisk is sending a 481 in response to an INVITE for reasons I do not understand. Here is the INVITE: INVITE sip:8009499014 at X.YYY.32.3:5060;transport=udp SIP/2.0 Record-Route: <sip:X.YYY.32.10;lr=on;ftag=247898> To: <sip:8009499014 at X.YYY.32.10 :5060>;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 From: "Scott Thompson" <sip:7166359474 at X.YYY.32.10>;tag=247898

Hello all! I can see a strange problem during invite in dialog in the context of timer handling. Given is the following incoming call from provider at 8.195.88.234 (2 at 2) to my asterisk at 28.19.57.152 (1 at 1): After 900s suddenly *asterisk* starts the timer reinvite - I would have expected the reinvite started by the provider as usual. The expected reinvite by the provider is started

Hi All, I am trying to use iconnecthere to make outbound calls. I am behind a linksys router. I keep getting this error 481 "Call Leg/Transaction Does Not Exist". Does anyone have any prior experience with this problem. Any leads will be much appreciated. Attached are the conf files and logs #SIP.CONF ; SIP Configuration for Asterisk [general] port = 5060 ; Port

Hello, 'factor' returns <NA> for my S4 object when the class is given an "as.character" method. Here is a minimal example: > setClass("foo", contains="numeric") > bar <- new("foo", 12) > factor(bar) [1] 12 Levels: 12 > setMethod("as.character", "foo", function(x) paste("x=", x@.Data)) [1]

Dear all, Since my package is based on S4 classes, I would like to know how to add a slot to an existing S4 class without breaking the code of users of my package. Assume the following S4 class: setClass("MyClass", representation(name = "character", type = "character", data = "data.frame" ), prototype(name =

R-devel, I'm interested in looking at some examples of existing R packages that rely heavily on S4 classes to get a feel for varying styles and package organization techniques. Could you recommend any packages that might serve as a good starting point? Thanks in advance, Jeff

R-devel, I'm interested in looking at some examples of existing R packages that rely heavily on S4 classes to get a feel for varying styles and package organization techniques. Could you recommend any packages that might serve as a good starting point? Thanks in advance, Jeff

pkgA's NAMESPACE has importFrom(graphics, plot) exportClasses("A") exportMethods("plot") R/foo.R has setClass("A") setMethod("plot", "A", function(x, y, ...) {}) During R CMD INSTALL pkgA_1.0.tar.gz we are told ** preparing package for lazy loading Creating a generic function for 'plot' from package

> Another five lines that have no place in an AD DC smb.conf, the 'vfs > object' line especially. thanks for the suggestions, but even with the basic smb.conf file compiled by the classicupgrade command we have the same issue on client machines, so it appears those parameters aren't creating the problem -- Lorenzo Milesi - lorenzo.milesi at yetopen.it YetOpen S.r.l. -

Alex pointed out the consequences after a transaction is not committed when a subvolume is deleted, so in case of a crash before an actual commit happens will let the subvolume reappear. Original post: http://www.spinics.net/lists/linux-btrfs/msg22088.html Josef''s objections: http://www.spinics.net/lists/linux-btrfs/msg22256.html While there''s no need to do a full commit for

Hi, The title tells it all. How do I remove a transation when using the Globalize plugin. Thanks. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFFw@public.gmane.org To unsubscribe from this group, send

Hi, Since CTDB tree has been merged in Samba tree, any new CTDB development would be done in Samba tree. Till combined Samba+CTDB is released, CTDB fixes would be released as minor releases starting with 2.5.1. Amitay. Changes in CTDB 2.5.1 ===================== Important bug fixes ------------------- * The locking code now correctly implements a per-database active locks limit. Whole

Hi, I am using SIP extensions connected to the PSTN with the CAPI Channel driver. All works fine except that one of the sip phones keeps ringing when the caller hangs up before extension is answered. The phones are grandstream 100, though we get the same behaviour using other phones (X-lite, Kphone). It behaves the same regardless of whether the incoming call is from a SIP extension or an

I have an Avaya 4602 IP phone that was previously working with Asterisk. It was being used elsewhere for several months, and I recently set it up again to work with Asterisk. Everything works fine for several minutes -- I am able to receive and make calls as expected. However, after a few minutes, and every few minutes thereafter, I get the following message on the console: -- Got SIP

Hi. We're trying to upgrade an old NT domain to AD. It's our second upgrade, and while the first was successfull this one has raised some issues for existing Windows 7 clients. If we disconnect the computer from the domain and join it back to the new S4 AD it works. Existing clients throws this error in Samba: Kerberos: AS-REQ b1rd42nbtmp648$@NT4DOMAIN from ipv4:10.0.0.42:49472 for krbt