search for: 0_stable

This has been mentioned before on this list, but in order for md5.c to compile successfully (OpenBSD 3.3), the following must change in md5.c: #if defined( __FreeBSD__ ) || defined( __OpenBSD__ ) # include <sys/endian.h> Change this to be: #if defined( __FreeBSD__ ) || defined( __OpenBSD__ ) # include <machine/types.h> # include <machine/endian.h> And -E is an invalid

...version tags > > > > Can someone please tell me which version tags I should be > using when checking out with cvs. Right now I have two > directory trees: > > 1) /u1/src/asterisk/asterisk-stable > > 2) /u1/src/asterisk/asterisk-dev > > #1 has "Tv1-0_stable" in the CVS/Tags file. I haven't used > tags a whole lot, but I'm guessing that I checked this tree > out with "cvs -z9 co -r v1-0_stable asterisk". I just tried > to do "cvs -z9 update -d" and it wouldn't complete. Am I > doing something wron...

...rand new machine with Redhat 9.0. I made sure that I had: bison cvs gcc kernel-source libtermcap-devel ncurses-devel newt-devel openssl1096b openssl-devel readline41 readline-devel zlib zlib-devel When I went to get Asterisk I did the following: cvs checkout zaptel libpri and cvs checkout -r v1-0_stable asterisk I built zaptel then libpri and had no problem, but asterisk complains with the following error: chan_zap.c:2772: `ZT_EVENT_DTMFDIGIT' undeclared (first use in this function) How do I fix this? Thanks, Hugh

Greetings, This is probably some configuration issue, but for some reason my system has stopped playing a ringing sound when an extension is dialed. The phone rings but there is no ring sound in the ear piece. Gene Kochanowsky

What is the no authority found problem? And how can I register with * on IAX. It keeps rejecting the request telling that XXX not dynamic host. rejected any idea THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030520/a8a5907d/attachment.htm

Hello, The v1-0_stable cvs release doesn't include the recent change ('poll' instead of 'select') in channel.c. Will it end up there any time soon, or we need to use cvs head to pick up this change? Thank you. Alex Zarubin Webley Systems -------------- next part -------------- An HTML attachment wa...

on the asterisk site, it was stated while ago, how to download stable version. like cvs checkout -r v1-0_stable asterisk-addons zaptel libpri but now it's not their. is stable-version removed from the CVS ? or is their some different procedure ? thank you -- Atif

Dear List, I'm running an Asterisk 1.0 server with 4 HFC cards and bri-stuff behind an "Anlagenanschlu?" with 8 B-channels in Germany. It worked fine with Deutsche Telekom, but since we switched to Arcor nothing works at all. After some debugging, I called Arcor helpdesk who told me that they do not have a D-Channel and cannot signal one at my request either. Is it possible to use

Hello everyone, As far as I can tell, if we try to forward a voicemail (by going into voicemail and saying that we want to forward it to another extension) it crashes asterisk. voicemail.conf does not seem to be where I should be looking. Any ideas? I did a 'cvs checkout -r v1-0_stable asterisk' when checking out from CVS. Should I be on a newer version? TIA, -Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050105/3b587724/attachment.htm

Hi All, I am a newbie and having some trouble getting Asterisk to work. I have checked out zapata zaptel libpri and asterisk (both v1-0_stable and regular--in separate directories). All built according to the documentation I have found and installed correctly. I have modified/created zaptel.conf zapata.conf as attached. I get the following messages/error during asterisk startup. == Manager registered action StopMonitor == Mana...

...nd a last one on thursday (24 june) Since this one, my H323 calls loose they audio, both sides. Calling directly from Gatekeeper is ok, so problem comes from h323 asterisk channel. I saw few people telling about similar problem begining of month, does they solve their problem? I also grab an 1-0_stable version to test if this problem is also there but I get an "undefined symbol: ast_channel_walk_locked" in app_groupcount.so module, so asterisk failed. What does it mean? Thanks for any hint. -- Dash

...sistently different about the calls with silent start compared to those without the problem. The SIP INVITE is sent to the phone, the phone rings, and OK's the call. The RTP stream begins straight away. I'm using cvs-head as of 2004-06-10, CVS-D2004.06.09.14.00.00, recently updated from v1-0_stable dated a few weeks before. This problem existed with both versions. The phones are Cisco 7960's with SIP image 6.3, the calls are incoming on an E1 connected to a TE410P. Calls are queued and the phones statically defined as agents in the queue which has a ringall strategy. There is no queue an...

...ent: Sunday, November 21, 2004 11:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Queue Patch - estimated hold time announcements On a pre 1.0 version I was running I patched queue.c to add estimated hold time announcements. Stable 1.0 (cvs checkout -rv1-0_stable asterisk) does not appear to of included this patch and of course patching the current queue.c with the patch I have fails. I looked at the Matis to see if an updated queue.c file is available or a patch for the current but I did not see anything - not too sure how these updates are included. Tri...

Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context

I have recently noticed that the "Action: Originate" options in asterisk 1.0 CVS has changed sometime between 2/23 and 3/18. I have a 2/23/04 CVS installation (cvs checkout -r v1-0_stable asterisk ) that allows me to make calls like this using the Manager Interface on port 5038. action: login login: admin secret: mypass action: originate exten: 200 context: stations channel: SIP/agent007 I have a 3/18/04 CVS installation that does NOT work the same way. En...

...lists.digium.com Subject: [Asterisk-Users] Manager Interface "Action: Originate" changed I have recently noticed that the "Action: Originate" options in asterisk 1.0 CVS has changed sometime between 2/23 and 3/18. I have a 2/23/04 CVS installation (cvs checkout -r v1-0_stable asterisk ) that allows me to make calls like this using the Manager Interface on port 5038. action: login login: admin secret: mypass action: originate exten: 200 context: stations channel: SIP/agent007 I have a 3/18/04 CVS installation that does NOT work the same way. En...

Hi, I downloaded Asterisk using this command a couple of weeks ago... # cd /usr/src # export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot # cvs login # cvs checkout zaptel libpri asterisk Can someone tell me what I need to type in order to get the latest stabe version when I rebuild my server ? Thanks, Paul. -------------- next part -------------- An HTML attachment was scrubbed...

...path from SLINR to UNKN Apr 23 11:22:23 WARNING[1097748]: Unable to restore read format on 'Modem[i4l]/ttyI5' Apr 23 11:22:23 NOTICE[1097748]: Unable to find a path from UNKN to SLINR Apr 23 11:22:23 WARNING[1097748]: Unable to restore write format on 'Modem[i4l]/ttyI5' Running v1-0_stable * from CVS. Regards, Tobias

Hi, about select vs poll differences, like in cvs message below <snip> > > Modified Files: > Tag: v1-0_stable > BUGS > Log Message: > Update 1.0 BUGS file to be aware of select vs. poll stuff <snip> > +* The number of channels that can be simultaneously run through Asterisk > + may be more limited than in development releases due to the use of > + select vs. poll. High density...

...;") in new stack -- Playing 'vm-login' (language 'en') **Here I push a button** May 9 18:26:18 WARNING[98311]: chan_sip.c:5027 receive_info: Unable to retrieve DTMF signal from INFO message from 5e9f165cd1aea27c@159.144.58.141 By re-installing an older (cvs checkout -r v1-0_stable asterisk) version - everything works fine again... thats with NO config changes at all.. Has someone removed some support for the transporting of DTMF (eg, info?) - I am using... dtmfmode=info in sip.conf with BudgeTone-100's (sent with absolutely no signatures or attachments) -------------- n...