similar to: Caller-ID name delivered in double-quotes

Hi, I noticed that my 4.8R box seems to have a NFS problem. I use a FreeBSD/i386 4.8R box as a NFS server, and a FreeBSD/i386 5-CURRENT (as of today) as the NFS client. On the client, /a (on the server) can be mounted successfully with -o rw,bg,intr,mntudp options. However, when a large amount of write operations such as "cp -r /home/ncvs /a" are performed on the client, the

What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from 1.4.0.) Yuan Liu

Hi folk, I'm trying to register an extension through softphone and got stuck.I got below error:- [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: missing sent-by in Via header [Nov 22 07:31:28] ERROR[3522]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("", "(null)", ...): Name or service not known [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:16056

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

I know I'm getting tired of looking at code, but why is the function register_verify defined in 2 different files? chan_iax2.c line 3860 static int register_verify(int callno, struct sockaddr_in *sin, struct iax_ies *ies) chan_sip.c line 4869 /*--- register_verify: Verify registration of user */ static int register_verify(struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req,

Hi Mark, Current CVS, * isn't correctly remembering the tag added to the To header by a server. For instance: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 81.96.69.210:5062;branch=7b858c09 From: steve-ata186 <sip:asterisk@81.96.69.210:5062>;tag=14925711 To: <sip:18478974611@4.42.235.170>;tag=t2907cab0911c8g Call-ID: 232752ec518d398f25ce03f45a8940f3@81.96.69.210 CSeq: 102 INVITE

My asterisk process produced the following errors this morning: Dec 8 10:44:07 WARNING[50315282]: rtp.c:829 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:2352 sip_alloc: Unable to create RTP session: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:8024 sip_request: Unable to build sip pvt data for

Is anyone out there using app_dbodbc (http://www.bkw.org/~brian/app_dbodbc.c)? Any problems with it? I was able to get it all working, but it causes * to segfault every now and then. It does not appear to be related to any specific function (ODBCget,ODBCput,ODBCdel,ODBCdelltree). It is 100% repeatable. If I noload the module, everything works fine, but when its running, after calls to any of the

Hi, We are preparing to make the release of OpenSSH 4.8 soon, so we would greatly appreciate testing of snapshot releases in as many environments and on as many operating systems as possible. The highlights of this release are: * Added chroot(2) support for sshd(8), controlled by a new option "ChrootDirectory". Please refer to sshd_config(5) for details, and please use this

Hello everybody! [Introduction] This is a quite long message, but I think the problem is interesting. [The problem] Does anyone know how can I tell Asterisk that a certain user has a certain telephone number (or address)? For example, I have some registered users, but nor the client (X-lite) nor the server (Asterisk) specifies what telephone number has the user. I don't want to

My keys are secured with a passphrase. That's good for security, but having to type the passphrase either at every login or at every invocation of ssh(1) is annoying. I know I could invoke ssh-add(1) just before invoking ssh(1), if I keep track of whether I invoked it already, or write some hacky scripts; but the rest of OpenSSH is wonderfully usable without any hacks. Hence, this patch.

Hi... Here''s a working Perl script that I want to be able to do in Ruby: use WWW::Mechanize; my $url = "http://host/tt?name=chris"; my $mech = WWW::Mechanize->new(); $mech->add_header( Referer => "http://chrismcmahonsblog.blogspot.com" ); $mech->add_header( Cookie => "messageid=170118; memberid=1007"); $mech->get($url) so the header

hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

Am I missing something here? I see if (needvideo) { /* only if video response is appropriate */ add_line(resp, m_video->str); add_line(resp, a_video->str); add_line(resp, hold); /* Repeat hold for the video stream */ } else if (p->offered_media[SDP_VIDEO].offered) { snprintf(dummy_answer,

Hi list, I'm attempting to re-create a Repeated Measures Compositional Analysis as described in the work by Aebischer et. al. (Ecology. 1993. 74(5): 1313-1325). In this paper they describe transitions of data into a log ratio difference matrix, from which they obtain two matrices using a monova routine. I am able to produce the second of the two matrices, but I'm having trouble

hi, i was searching for an efect in wxruby like fxshutter from fxruby. http://www.fxruby.org/doc/examples.html is it possible to build an skeleton like this in wxruby? i am ruby beginner -- Posted via http://www.ruby-forum.com/.

I'm handling mail for several domains, let's call them a.com, b.com, and c.com. I have certificates for each of these domains individually via certbot (letsencrypt) and nginx is happy with all of that. Since I initially configured the site to handle mail only for a.com, my /etc/postfix/main.cf file currently has these two lines: smtpd_tls_cert_file =

Hi, The Polycom 600 phones do not natively bridge with Asterisk. I've solved the problem, but I'm not sure how general it is, so I thought I'd ask this list for advice. It's necessary to use a recent Asterisk CVS for this, since there was a problem with session versions in earlier CVS builds. The problem now is the Via field. When the reinvite goes out, the branch number

I know asterisk isn't a real SIP proxy and is more of a multi-protocol pbx with limited SIP support, but... ... is it possible if you have a central registration server that handles all of your dialplan routing and several asterisk PSTN gateways that it routes calls to for an outbound SIP conversation using reinvites and NOT have the registrar box try and send ANY RTP traffic back to the

Wow, thanks Brian! Everything I saw said the patch was only needed on older releases. I've updated several times over the last week. I patched two systems today, one 3/11/05 and one 3/17/05 and now they both work. Should have posted here sooner! Brian G. On Thu, 2005-03-17 at 13:28, Brian Buhrow wrote: > Hello. I'm writing in response to your message to the ASterisk-users >