search for: 723

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2003 Jul 08
0
SIP Problem (previous post) .. information might be relevant
...the other end (remote) and not in my control. help :) please!! Dave Signal=0 Duration=250 (no NAT) to 216.52.153.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: "21382890" <sip:21382890@217.168.168.5>;tag=as6556b0d9 To: <sip:723@216.52.153.207>;tag=26845C24-FDA Date: Tue, 08 Jul 2003 22:22:57 GMT Call-ID: 14bce0f47fb42b734f7904ca351a4220@217.168.168.5 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 117 INFO Contact: <sip:723@216.52.153.207:5060> 10 headers, 0 lines set_destination: Parsing <sip:723@...
2004 Jan 16
4
G.723.1 codec
Hi, Want to do some experiments with the G.723 codecs - where can I download the 723 source code for Asterisk? I know there are some ongoing discussion regarding patents and license fees for the g.723 but I have some hardware on which I only have the 723 and need to test it privately. Thanks! Dan __________________________________________...
2003 Jul 08
2
oh323 problem (small one)
...23 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial("Zap/1-1", "OH323/723@216.52.153.206") in new stack 1:04.782 ThreadID=0x4958a540 H323 Attempt to use invalid URL "723@216.52.153.206:1720" -- Couldn't call 723@216.52.153.206 -- Hungup 'H323:0' == Everyone is busy at this time -- Executing Hangup("Zap/1-1"...
2003 Aug 08
2
Re2: Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO]
...2 in start_thread () from /lib/tls/libpthread.so.0 Rgds Sip Rtp ----- Original Message ----- From: "Michael Manousos" <manousos@inaccessnetworks.com> To: <asterisk-users@lists.digium.com> Sent: Friday, August 08, 2003 3:56 PM Subject: Re: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk > > Sip Rtp wrote: > > Hi List, > > > > I am facing the reverse problem as stated here.I am > > using ATA 186 to make > > and recieve call to * through OH323 driver. > > When I use G711 codec in the ATA to make call then > > then as s...
2008 Jan 17
3
vector generation
Dear Contributors: I have the next vector: "Z" 526 723 110 1110 34 778 614 249 14 I want to generate a vector containing the ratios of all the values versus all the values of the z vector. I mean a vector containing the values of 526/723, 526/110, and so on, 723/723, 723/110, and so on, and so on. Is this doable in a simple way?? Thanks in advance aga...
2005 Jan 15
0
ppp connection only every second time
...n 15 14:00:49 waltz pppd[752]: LCP terminated by peer Jan 15 14:00:52 waltz pppd[752]: Using interface ppp0 Jan 15 14:00:53 waltz pppd[752]: Exit. /var/log/messages: Jan 15 13:59:11 waltz pptp[720]: anon log[main:pptp.c:243]: The synchronous pptp option is NOT activated Jan 15 13:59:11 waltz pptp[723]: anon log[ctrlp_rep:pptp_ctrl.c:243]: Sent control packet type is 1 'Start-Control-Connection-Request' Jan 15 13:59:11 waltz pptp[723]: anon log[ctrlp_disp:pptp_ctrl.c:721]: Received Start Control Connection Reply Jan 15 13:59:11 waltz pptp[723]: anon log[ctrlp_disp:pptp_ctrl.c:755]: Clie...
2004 Sep 21
1
IP phones AT-723 or AT-323
Is anybody familiar with these IP phones AT-723 or AT-323 I think it is made by this company: http://www.atcom.com.cn/at723E.html -- #Joseph
2006 Nov 07
0
[723] trunk/wxruby2: Added WindowCreateEvent and WindowDestroyEvent + event handlers (AF)
...patch .copfile {border:1px solid #ccc;margin:10px 0;} #patch ins {background:#dfd;text-decoration:none;display:block;padding:0 10px;} #patch del {background:#fdd;text-decoration:none;display:block;padding:0 10px;} #patch .lines, .info {color:#888;background:#fff;} --></style> <title>[723] trunk/wxruby2: Added WindowCreateEvent and WindowDestroyEvent + event handlers (AF)</title> </head> <body> <div id="msg"> <dl> <dt>Revision</dt> <dd>723</dd> <dt>Author</dt> <dd>brokentoy</dd> <dt>Date&l...
2008 Jan 17
1
Res: vector generation
hi Juan, It is not so elegant, but work fine. I know that our colleagues can do it on a simple line. z<-c(526,723,110,1110,34,778,614,249,14) v1<-NULL v2<-NULL for (i in 1:(length(z)-1)) { for (j in i:length(z)) { v1<-rbind(v1,z[i]) v2<-rbind(v2,z[j]) } } df<-data.frame(cbind(v1=v1,v2=v2)) names(df)<-c("v1","v2") df$ratio<-df$v1/df$v2 Kind regards, Miltinho Brazil...
2008 Jan 08
28
1.9.3 release, rakefile
Hi I''d like to put out a 1.9.3 release perhaps later this week/weekend. If you have a chance to test the build and samples esp with latest rubygems, please do. There are still some bugs on the list, and samples to do, but this should address all the build/install probs that have come up. And it would be good to get some testing and feedback on some of the new classes. A note on the
2003 May 29
1
a beginner's SIP question ..
I am trying to get asterisk to dial this address : sip:723@216.52.153.207 Using a softphone on my PC (217.168.168.49) it dials immediately and I get a voice prompt .. I have configured an extension, 1303 on asterisk, modifying the demo configuration : exten => 1303,1,Dial(SIP/723@216.52.153.207) When from my softphone I dial sip:1303@217.168.168.51...
2011 Jun 13
2
[Bug 723] New: extensions/libxt_NFLOG.man definines invalid range for --nflog-group
http://bugzilla.netfilter.org/show_bug.cgi?id=723 Summary: extensions/libxt_NFLOG.man definines invalid range for - -nflog-group Product: iptables Version: CVS (please indicate timestamp) Platform: All OS/Version: All Status: NEW Severity: trivial...
2003 Jun 03
0
Asterisk terminates unexpectedly with SIP call and G.723 codec
Hi, I'm using a Cisco ATA186 and iConnect to complete PSTN calls to the US. I've noticed that when I set the Cisco ATA to use LBRCodec to 0 (g.723 instead of g.729), AudioMode 0x00150015 and RxCodec, LxCodec to 0, (use g.723) Asterisk will connect to iConnect, successfully natively bridge the call and then about two seconds later not just drop the call, but terminate unexpectedly. The asterisk daemon will stop uncleanly. If I set the Rx...
2003 Jun 06
1
more about SIP ...
I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Cal...
2007 Oct 15
11
What web GUI are people happy with?
...asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy at manistee.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071015/e5447466/attachment.htm
2003 Sep 30
1
[Bug 723] Password expire not working properly
http://bugzilla.mindrot.org/show_bug.cgi?id=723 Summary: Password expire not working properly Product: Portable OpenSSH Version: -current Platform: Sparc OS/Version: Solaris Status: NEW Severity: major Priority: P2 Component: Build system Assigne...
2003 May 31
1
Passing audio stream through Asterisk or not?
Hi all, One short question. When one extension (let's say ATA-186, SIP image, G.723 codec selected) try to call an external SIP address like: SIP/user@domain.com, where another identical ATA-186 is available with G.723 codec selectrd, after the signaling phase, the call is established through Asterisk or directly between the two ATAs? There is no G.723 codec available on Asterisk...
2003 Jul 08
1
oh323 prob :)
...Asterisk to dial an h323 call termination service .. right now getting this message: -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial("Zap/1-1", "OH323/h323:723@216.52.153.206") in new stack 5:59.330 H323 Cleaner H323 Connection ip$localhost/18729 terminated. ERROR[1230546240]: File chan_oh323.c, Line 704 (oh323_call): H323:0: Could not call h323:723@216.52.153.206. -- Couldn't call h323:723@216.52.153.206 -- Hungup ...
2004 Apr 13
0
AW: IP Phones that support G.723 on H.323
> > Does anyone know of Phone that supports G.723 on H.323. > Innovaphone tiptel 200 for example. http://www.innovaphone.com/webneu2/products/en_IP200.asp One of the nicest phones I've seen so far, h.323 only though. Bye, Martin
2004 Sep 27
1
Cisco IP phone G.723
Hi, I have 1 phone it is a 7910 that when I try to make calls it seems to be trying to use g.723. The error I get is cannot find a path from GSM to G723. Am I missing something? Is there a way to set the codec in the phone or in the skinny.conf? I'm stumped -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-u...