similar to: Re: Asterisk-Users Digest, Vol 5, Issue 221

I don't think it's the snom, (the break key is set to "off") the "#" key is not being interpereted by the PBX as an attempt to initiate a transfer. Is this an error in my extensions.conf? Brian > >Message: 4 >Date: Wed, 15 Dec 2004 19:39:39 -0500 >From: Info <info@idatasys.com> >Subject: Re: [Asterisk-Users] Help with transferring a second call

Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mitul at enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] No sound with internal

Hi! I want to tell asterisk to simply pass-through any codecs that my phones support. I have to use codecs that are not popular and implemented by a third-party, asterisk has nothing to do with them. I've made a test with g722 (that asterisk doesn't support), i've set all my two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it

Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module

This was reported to me by a colleague in China, so I may not be reproducing exactly what they are seeing (which I suspect is rw2011), but this is what I see: > version _ platform i386--netbsdelf arch i386 os netbsdelf system i386, netbsdelf status major 2 minor 1.1 year 2005 month 06 day 20 language R > help(Hershey) : : If the 'vfont' argument

Hi! Maybe you can help me. I want to tell asterisk to simply pass-through any codecs that my phones support. I have to use codecs that are not popular and implemented by a third-party, asterisk has nothing to do with them. I've made a test with g722, i've set all my two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it possible that

I am unable to get my 9133i to register with my asterisk server. I am including config files below, this a simple test network so there's nothing secret in the config files. I have upgraded the phone to the latest software version (1.4.3) I'm not sure what the problem is. I can call the phone from a softphone, but the 9133i says "no service" on the screen and I can't dial

Dear R users, I would like to use sub- and super-script in axis labels. I assume this is best done using Hershey symbols. When trying to find information on using Hershey font symbols in axis labels, I came across the following discussion thread: http://maths.newcastle.edu.au/~rking/R/help/02a/1857.html Have Hershey font implementations moved on since then? Thanks, Sander. --

Hello! The question is simple: What is the escape sequence for a new line when using Hershey fonts? I obviously tried '\n' but it didn't work (see the sample below). I looked at 'demo(Hershey)' but all it only shows escape sequences for printable characters. The sample I've been using to try to find the escape sequence is below. You can comment or un-comment the

Full_Name: Albrecht Gebhardt Version: 0.99.0 OS: alpha, osf4.0 Submission from: (NULL) (143.205.180.40) DEC cc doesn't like c++ style comments // like this one Please use standard cc comments or #if 0 ... #endif A quite large patch follows (I hope it will pass mailing with not too much wrapped lines): --- ./src/main/g_her_glyph.c.dec-cc.patch Mon Feb 7 14:48:10 2000 +++

Ray Brownrigg wrote: > I have a question from a Chinese colleague in Beijing, and wondered if > you were able to help answer it. Just out of curiosity, how do you know my contact e-mail? > I know about the Hershey fonts, and > that Kanji will give a few hundred hanzi in R, but is there anything > else that we can do to get a larger number of characters? [SSLib is > some

I wonder if we couldn't just use the PNG , SVG, support from the GNU plotutils from which we already have the Hershey scalable vector fonts [-> help(Hershey)] The new plotutils (July 2000) do support these (and will probably even better in the future, i.e., we can build on others people free software. Paul [who built "Hershey" into R], do you think this path is worth pursuing?

I am unable to get a my Draytek working with our Asterisk server. I can make/recieve calls but get no audio. I have tried the various codecs at the Vigor end but still getting nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE

Hi, I have been trying to get my asterisk box to send SMS's to my Panasonic dect phone via a Linksys pap2. I believe I have the message centers setup correctly between * and the phone. The pap2 is configured to only use G711a. The Asterisk version is 1.0.7. In my /etc/asterisk/extensions.conf I have [smsphone] exten = 199,1,Goto(smsmorx,${CALLERIDNUM},1) [smsmorx] exten =

What options do the others offer over one another? Is there a difference, or is it simply a case of one superseding the other? Would be good to put something together that details the different methods, and which situations maybe they better suit, if any. Ben -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

Yes that is the plan. I'll have to find some time to graft the patch onto 2.1 mainline and post it here. KJ -----Oorspronkelijk bericht----- Van: Myke Place [mailto:mp@trans.xmission.com]Namens Myke Place Verzonden: donderdag 18 november 2004 22:14 Aan: Klaas Jan Wierenga Onderwerp: Re: [Icecast] Dumping streams to a file? Is the plan to eventually move this from -trunk to the mainline

This is a call coming in through the ISDN to 7940's. Answering with non-codec capability 1 - Is that the problem? SIP Debugging Enabled We're at 10.1.0.11 port 18406 Answering/Requesting with root capability 8 Answering/Requesting with preferred capability 4 Answering/Requesting with preferred capability 8 Answering with non-codec capability 1 <<<<<<------------- 12

Hi, I'd like to know if it is possible to use Hershey vector fonts to create very primitive musical notation. If I can hang some whole notes on these lines X11() plot(0,0, xlim=c(0,10), ylim=c(0,10)) # Staves: for (i in c(seq(from=2,to=2.8,by=0.2),seq(from=4,to=4.8,by=0.2))) { abline(h=i) } it is enough. Best wishes, Atte Tenkanen University of Turku, Finland

I've just installed asterisk on a Debian Linux (apt-get it) And i have placed two sip phones in sip.conf and i'm testing parking with them I have phone1-SIP/1000 and phone2-SIP/1007 The following happens if i park from calling party and everything is OK 1. Pickup Phone2 and call to Phone1 2. Talk 3. Phone2 dials #700 and parks the call (it is placed in 701) 4. Phone2 is hangup 5. Pickup

Dear All, I installed an Asterisk on a linux PC, and X-Lite on two Windows PCs, all in a LAN. But, when I make phone call from one X-Lite to another, I always get Call Failed: 404 not found. Here is my sip.conf: [Phone1] type=friend host=dynamic ;defaultip=192.168.1.103