similar to: Problem with SIP-Phones and * audio-files

Hello list, i've got some problems getting a E1 line with a E100P up and running (germany). # cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard E100P E1/PRA Card 0" HDB3/CCS/CRC4 YELLOW RED "YELLOW RED" sounds not so good. When launching asterisk, enabling pri debug on that span, i see outgoing attempts: ;---------- snip -------- > [00 01 7f ] > Unnumbered

Hello List, is someone operating a DID /P2P / Anlagenanschluss with more than one HFC-Based ISDN-Card ??? I have now 12 hours of setup-troubles behind me with Colt-Telekom, where we did not get it working with two HFC-based cards. Here the setup: - 2 HFC-ISDN-Cards (the one from Conrad-Electronic) - bri-stuff.0.0.2 (with the asterisk-sources from the download.sh-skript) - two NTBAs from

Hi all, I searched through the archives, but found nothing... Is there a possibilty, to take over a call ?? I have for example two extensions.. 102 and 103.... if 102 is ringing, but noone one the desk, I want, that 103 can answer this call on his phone, by just typing some digits... has someone such a setup running, and perhaps some hints for me ??? Thanks in advance.... -- Bye Ernst

strace on file access in asterisk shows that * is not even attempting to access the voice files. If I *manually* load app_playback.so, app_macro.so, and then pbx_config.so, I they will load and I get a dialplan. Ok, that's a problem -- autoconf is clearly not working, or there's some other related issue. So I try to use the demo and do "dial 500". This should connect and

Hi, I understand from earlier discussions that the reason dovecot doesn't support proxying of other SASL mechanisms than those which supply the plaintext password is that in general it would be possible to proxy any SASL mechanism since it might protect against man-in-the-middle attacks (which would prevent proxying). However, that has led to choice between letting users use PLAIN (or

Hi, I am fairly new at working with Asterisk. I am having a call quality issue that I really need to get ironed out before we go to rollout the system in a week. Any help would be greatly appreciated!!! Even if it is just pointing me in the right direction. My current setup: I have Asterisk Setup on a Dell Server. It has 2 T100P cards. One will be for out T1 PRI from the Phone Company (We

For whom asked me support for capi devices, that's here: http://www.junghanns.net/asterisk/ I'm using a AVM B1 card. also AVM passive card (FRITZ!PCI) works.... Then is you use SuSe all is configured by yast... Hello, probably is a feature what I'm asking for but because of my inexperience to asterisk this is my question: I've configured CAPI ISDN to receive calls. When I

Hello, I've found GlusterFS to be an interesting project. Not so much experience of it (although from similar usecases with DRBD+NFS setups) so I setup some testcase to try out failover and recovery. For this I have a setup with two glusterfs servers (each is a VM) and one client (also a VM). I'm using GlusterFS 3.4 btw. The servers manages a gluster volume created as: gluster volume

CentOS Errata and Bugfix Advisory 2016:0178 Upstream details at : https://rhn.redhat.com/errata/RHBA-2016-0178.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 41bcfe83e915dfe6408766d8c5d7d172fffab42e55c39f44ee7ded90ef9bbdfd cyrus-sasl-2.1.26-20.el7_2.i686.rpm

http://bugzilla.mindrot.org/show_bug.cgi?id=1041 Summary: Allow the admin to specify PAM service name Product: Portable OpenSSH Version: -current Platform: All OS/Version: Linux Status: NEW Severity: enhancement Priority: P2 Component: PAM support AssignedTo: bitbucket at mindrot.org

I'm having some difficulty with chan_local dial requests. It seems that when a chan_local call is picked up, that the native bridge "pops" the environment back to the settings of the original call. This is unexpected and leads to very frustrating results. My example below is a very distilled sample of a much more complex dialplan problem I'm having with chan_local, but it

I am messing around with gluster management and I've added a couple bricks and did a rebalance, first fix-layout and then migrate data. When I do this I seem to get a lot of failures: gluster> volume rebalance MAIL status Node Rebalanced-files size scanned failures status --------- -----------

Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. Trace messages here : -------------------- == New H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party

Ok so I've just built and installed a CVS (HEAD) version of asterisk on RHFC2 running a 2.6.13.3 kernel.org kernel. I installed the samples via "make samples". Everything seems to work except one thing. I'm trying to do the connect to the Digium IAX demo server portion of the demo (dial 500) and I just get the following messages. I am behind a NAT server and did NOT change

I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight. I took the additional step of nuking my modules directory first... When I used the command asterisk -v to start asterisk, it seemed to go along and get to the point where asterisk is running(ie Asterisk Ready). At that point it was eating all available CPU. I went ahead and tried to register a softphone to it via IAX2, which

Hi all I am using Asterisk 1.2.2 on frdora core 4. i have two sip UA. if i put canreinvite=yes voice Ok on both sides. and if i change canreinvite=no there is no voice (media through asterisk) one thing more if i try to use playback application for playing some sound file it is also working (like exten => 500,1,Playback(demo-abouttotry) this is working). here is sip.conf

Hi all, New user to asterisk having just got it compiled and installed. Running with no digium hardware (yet) and no soundcard in asterisk box. Problem is using the sample configs with a sip phone added as follows [2203] type=friend username=2203 secret=2203 host=dynamic defaultip=192.168.0.2 dtmfmode=inband canreinvite=yes the console on * when running with -vvvvc says :- (whenb trying to

On 25 Feb 2015, at 20:59, Peter Mogensen <apm at one.com> wrote: > So, why not just extend the support for proxy authentication forwarding > to any single-handskake SASL-IR mechanism, which doesn't use > channel-binding? (which includes PLAIN, but also GS2-KRB5, and possibly > others). Yeah, I guess it would work for several of the auth mechanisms. It's a lot of work

Hi, i've just got 2 grandstream phones and when I try to connect them with * I get the following: -- Playing 'demo-abouttotry' (language 'en') WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 02da8d3d-a8de-c7e1-5f07-c032ab23d909@10.1.1.179 for seqno 59134 (Response) I've seen there was some discussion on this already but i

Hi, I've successfully installed Asterisk 1.0 on Fedora Core 2 with the 2.6.8 kernel. I have two other computers running X-lite connecting to it. I've been able to set them up so I can dial extensions "123" and "124" to talk between them. I'm able to access the default "1000", "500", and "600" extensions and they all seem to work.