similar to: Re: Asterisk-Users digest, Vol 1 #520 - 9 msgs

Hi, I recall seeing a sample extensions.conf file that allowed tollfree calls to be routed via iaxtel to the US and the NL, but I must be going blind, because I've scoured the list but can't find it. Can someone send it to me if they have it? Much appreciated. Thanks! --- Paul Cheng M?ty?s kir?ly ut 10 H-1121 Budapest HUNGARY paul.cheng@alum.mit.edu mobile: +36 30 381-9311

Hi, I searched the archives about this, but didn't find any references. When I make an outbound SIP call, the call completes and everything is fine, but in the Asterisk console, I keep getting a huge stream of warning messages: "WARNING[1200876848]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames" I thought I saw this in a post earlier, but I

We recently took a few Polycom Soundpoint IP 500 to test out in Asterisk environment. So far it has been good. Call Hold, Transfer, DMTF etc. However, I do notice every now and then the Polycom fails to register with Asterisk. Asterisk console outputs the following: Feb 3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response: Unable to determine sequence number from '' Feb 3

[Resent due to non-descriptive subject line.] Hi folks When connecting two SIP users, is there any way to stop Asterisk from sending SIP 183 Session Progress messages, either globally or per-peer? Scenario as follows: Call from UA1 to Asterisk (UA2) to UA3. UA3 sends RTP before SIP OK to Asterisk (UA2). Asterisk (UA2) detects early audio from UA3 and sends 183 Session Progress with SDP to

For someone that's network-aware, but hasn't sat down and plowed through umpteen SIP-related RFC's and memorized the standards, is there a good primer on troubleshooting SIP issues? I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk and my Sipura 942's, for instance... Not sure what these are... perhaps the qualify keepalives? In which case, I guess

I asked on the IRC channel last night and was told the IAXTEL had been down for a few months now. It had a very poor uptime.. Maybe someone can tell us why the uptime was so poor. Alex Message: 9 Date: Wed, 9 Jul 2003 01:05:00 +0200 From: Paul Cheng <asterisk@klarium.com> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IAXTEL toll-free Reply-To:

hi there tried to use subset with drop=TRUE, but all the 'old' levels are preserved, i.e. when calling e.g. ftable a lot of zeros are displayed >x<-subset(LREG, (kir=='AA' | kir=='BB') & (type=='t1' | otype=='t2'), drop=TRUE, select=c(event, kir, type)) > ftable(x) i explicit have to call factor like

Hi all, In the INVITE from my SIP provider to Asterisk i can see that the Allow Header includes an INFO Method, but Asterisk replies a 200 OK with an Allow Header without INFO Method. But in the RFC3261 (20.5) you can read: "All methods, including ACK and CANCEL, understood by the UA MUST be included in the list of methods in the Allow header field, when present. " My SIP provider

Dear Asterisk community! I am having trouble with a project concerning the 183 Session Progress SIP messages. Asterisk seems to only accept these when there is also a Session Description (SDP) included in the message. I also verified this by looking at the code. However for a project we are working with a trunk to a third party system (Alcatel) and they are insisting that this behavior is

Hi, I have an Alteon in test (a sip/rtp load balancer). This Alteon sends to the asterisk box a "SIP OPTIONS" to know if asterisk is alive. However, asterisk sends me a 404 message and not a response like, for example, a Thomson (200 + SDP) I wrote a very little script (you can find it at the end of the email) to send an Options message to asterisk/phones to try. It works

Happy New Year to everyone! I'm trying to track whether this bug: http://llvm.org/viewvc/llvm-project?revision=163923&view=revision has been fixed in the Apple LLVM version we use: Apple LLVM version 4.2 (clang-425.0.28) (based on LLVM 3.2svn) Any ideas how I can check that? Thanks, Ákos Somorjai Developer Support Manager GRAPHISOFT | Graphisoft Park 1. Budapest 1031 Hungary | +36 1

W/o downtime: see the wiki page below, the "Converting" section. Briefly for my case something like this should work and generally seems simple (no syntax checking yet, pseudo-code like): * Configuration uses mail_location = mbox:~/mails * setup per-user mail location and do for each user individually in a serial manner: -- doveadm sync maildir:~/Maildir; mbox is synced to Maildir,

hi, I have an 20G archive of pic files which would have to be mirrored onto another server. It contains large JPEGs, around 10k files, one of the servers is located in Boston, the another is in Budapest, Hungary. I use rsync rsync://remote_box/remotedir localdir/ -zcvr --progress --size-only The issue is that the remote server sends the filelist very slow. Looking into the rsyncd process with

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

2018-06-05 15:27 GMT+02:00 George Joseph <gjoseph at digium.com>: Thank you very much, George for replying. > > > On Tue, Jun 5, 2018 at 3:35 AM Olivier <oza.4h07 at gmail.com> wrote: > >> Hi, >> >> After a long discussion with a friend, I would like to ask here: >> >> 1.According SIP RFCs, is possible/recommended to have different values in

Just in case other people on the list have this problem... Begin forwarded message: > From: "George Lin" <glin@cosini.com> > Date: Thu Aug 14, 2003 6:54:46 AM Europe/Budapest > To: "Paul Cheng" <asterisk@klarium.com> > Subject: RE: FW: [Asterisk-Users] SIP NAT question > > Dear Paul, > > Thanks for the suggestion. It works now. > >

Hi, Please correct me if I''m wrong but it seems to me that in the case of a remote file source the path must contain the plain filename after the module and paths with directory elements are not supported. What I wanted is the following: +++++ fileserver.conf +++++ [system] path /services/puppet/common/ allow * +++++ site.pp +++++++++++++ file {

Hi Everyone, I perform a snapshot and a zfs send on a filesystem with a recordsize of 16k, and redirect the output to a plain file. Later, I use cat sentfs | zfs receive otherpool/filesystem. In this case the new filesystem''s recordsize will be the default 128k again. The other filesystem attributes (for example atime) are reverted to defaults too. Okay, I can set these later,

Hi! I was really stupid! I have defragmented my ext3 partition with e2defrag, altought i have done that many times in debian without problems on my new gentoo installation it had bad results. When i wanted to boot this partition i got serious e2fsck errors. It has reported that (only) inode 8 has illegal blocks, so i have run e2fsck -fy /dev/hda2, which has cleared the illegal blocks in inode 8.

Is there any solution for exporting R databases into SAS directly? Thanks Istv?n J?nosi Planim?ter Kft. Budapest, Hungary 36-1-320-49-55; 36-1-452-0545 janosi at planimeter.hu www.planimeter.hu -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or