similar to: asterisk 1.2.7.1 crashing my newly built system

Hello all, I'm looking to troubleshoot some echo issues and possibly "studdering" while using the speakerphone with the Intellitouch ITC3002. On the hardware end, i have a 2621 setup as the router with some policy maps for qos, and a 2900XL with cos priority set to 5. I have setup created 2 vlans currently (really small office), 1 for corp/data, and 2 voice vlan. Asterisk is on

Hey guys, I'm actively trying to get the "big" picture on how all this works and relates to each other. I've gone through some basic examples from the book and from the sample files just fine. Now, I've setup an account with a VOIP provider which does IAX termination (exgn.net) After getting an account and following their steps, I can make calls out using my IAX (cubix) and

I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to

Yes, I have the exact same problem. :( -----Original Message----- From: Tomislav Vojvodic [mailto:tomislav@vox-mundi.net] Sent: Thursday, May 11, 2006 5:48 AM To: xytek@hotmail.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

Hello all, I have two test beds running the exact same version of asterisk 1.2.7.1, latest of zaptel, libpri, etc.. Test bed #1 (Solaris 9,sparc ultra 5): This one is closer to a "production" machine, in that it is connected to a sip provider thru an iax2 connection and have an incoming DID configured. I can send and receive calls. Test bed #2 (Slackware Linux 10.2, AMD XP chip):

Does anyone know where to find an example or able to provide an example of how to do the following: When asterisk answers a call... Ask for number to dial...then dial that number? I am basically dialing into the asterisk box and then wanting it to take the digits I enter and dial them on an outbound zap trunk... I basically am just not sure how to have asterisk accept the digits and then use

Hi, I'm running * 1.2.7.1 on Red Hay 9.0 w/ a TDM 400 2 x FXS, 1 x FXO. I'm using a VTech cordless that makes three short beeps when someone another extension is picked up, presumably this lets you know if someone is trying to listen in.. Everything works, except the VTech now makes the three beeps everytime you try to use it, even if another extension is in use. It seems as though the

Hello smb.conf # Global parameters [global] workgroup = MYSERVER realm = interno.mydomain.com.br netbios name = DC-LINUX server role = active directory domain controller server services = s3fs, rpc, nbt, wrepl, ldap, cldap, kdc, drepl, winbindd, ntp_signd, kcc, dnsupdate idmap_ldb:use rfc2307 = yes ldap server require strong auth =

Sorry if this hit the list twice.. but I didn't see it come through: Hey guys, Just for archival purposes, I have setup the Intellitouch ITC-3002 (2006) SIP phones for use with asterisk (1.2.7.1). After a few "gotcha's", I was able to do transfer's, moh's, push a button to check voicemail, callerID, etc.. One big gotacha was the dial timeout (by default!) is set to

Hi, I was using Asterisk with Hylafax via IAX Modem. It works fine until I upgraded to Asterisk 1.2.7.1 I didn't change any configuration but it seems that Asterisk does not get the call from IAXModem anymore. I'm doing something like this Asterisk <--> IAXModem <--> Hylafax Usually when I use sendfax -n -d 260XXX somefile I'll see Asterisk receiving the call in

Has anyone managed to add the bristuff patch to 1.2.7.1 successfully? My attempts has ended up bad, so if anyone has a working patchfile for 1.2.7.1 I would be grateful to receive it. Thanks, Vidar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060430/1c56d94e/attachment.htm

Hi, I'm trying to find how to configure Asterisk 1.2.7.1 to allow two EyeBeam (3015c) to send Instant Messages between them... But I cannot find anything that explains how to do it! Anybody as a clue? is it possible? Now, when we try to send an Instant Message in the eyeBeam it says: "User not available". In asterisk console appears a message saying: ------ Apr 18 17:13:22

Hi! After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax doesnt work any more. I've installed spandsp-0.0.2pre25 (the same problem with spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile patch from the same directory. When starting asterisk I always get the follwing error message: [app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325

(This is a shameless copy-paste from the note I posted on http://bugs.digium.com/view.php?id=5090) I have again backported the whole T.38 shebang to the stable branch. The port was based on two versions of the t38passthrough branch: r19125, the latest unconflicted automerge, and r13623, the latest version without the new chan_sip flag structure. Basically, the port contains everything that

The make seems to go okay. [root@somebox asterisk-1.2.7.1]# uname -a Linux somebox.org 2.6.16-1.2080_FC5smp #1 SMP i686 i686 i386 GNU/Linux mkdir -p /var/lib/asterisk/sounds/digits mkdir -p /var/lib/asterisk/sounds/priv-callerintros for x in sounds/digits/*.gsm; do \ if grep -q "^%`basename $x`%" sounds.txt; then \ install -m 644 $x

Hi Last friday I have upgraded to Asterisk 1.2.7.1 (bristuff-0.3.0-PRE-1p.tar.gz). Since that I have a problem with my Asterisk box. I am receiving these messages: May 24 09:30:11 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2). May 24 09:30:12 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2).

Hi list! Are there any changes in the behaviour of the Dial command between 1.2.7.1 and 1.2.8.? I am forwarding calls to my legacy PBX using : exten => s,1,Dial(Zap/g1/8210,90,r) Ever since I upgraded to 1.2.9 it seems as if the Legacy PBX is no longer receiving the extension I am calling on the PBX and the call gets dropped to the switchboard extension on the legacy PBX. Did I goof up

Any plans to get the actual diffs of file changes into puppetdb? Right now you get the hashes of the buckets on the host, but if diffs could get into puppetdb, it could be extremely useful. -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To unsubscribe from this group and stop receiving emails from it, send an email to

I was using Asterisk 1.2.4 since it came out then, due to a disk crash, I had to install everything from scratch. Thus I started with a fresh Fedora Core 3 (basic, only what needed for Asterisk) and Asterisk 1.2.7.1 driving my Digium TDM400 card. With 1.2.4 and earlier versions, I used to be able to register with up to six SIP providers without a hitch. Now I am not able to register with more