similar to: sip authentication

Hi, Is there a module within the Asterisk standard distribution that provides SNMP features? Is there any third party software for that purpose? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/918b5ebf/attachment.htm

Hi, I use Asterisk 1.0.5 and TDM04B. When an incoming call over ZAP channel hangs-up, it takes 10 seconds until Asterisk realize that. How can I shorten the time of hang-up detection? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050614/de3eec0f/attachment.htm

Hi all, I have a problem starting the ztdummy. Here is what happens: [root@asterisk /]# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for ztdummy After this, ztdummy is visible with lsmod, but when I try MeetMe, I get following: == Parsing

Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Jan 28

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have

Hi, I have stress tested the Asterisk Voicemail. We have encountered problem with simultaneous calls that are sent to the same mailbox. It occurred that several calls were writing to the same file. It seems that there is a synchronization issue in the Voicemail application. Did someone else find this issue? What would be the solution/workaround for it? Regards, Stojan Sljivic

Hi, We have phones registered at another soft switch, and would like to use Asterisk as a Voicemail system. Is it possible and how to configure Asterisk to send NOTIFY messages (for MWI) to the endpoints that are not registered to the Asterisk? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL:

I just discovered that the SIP channel has undergone some major improvements. I'm now able to dial any SIP URL with dial, couldn't get it to work earlier, all domains had to be defined in SIP.conf. This, in addition to the SIPDOMAIN variable, makes the SIP channel even more useful. Thank you, Mark, for your additions! Now, ENUM/E.164 will propably work even better. I'll give it a

This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider

Hi all! I have setup a replicated/distributed gluster cluster 2 x (2 + 1). Centos 7 and gluster version 3.12.6 on server. All machines have two network interfaces and connected to two different networks, 10.10.0.0/16 (with hostnames in /etc/hosts, gluster version 3.12.6) 192.168.67.0/24 (with ldap, gluster version 3.13.1) Gluster cluster was created on the 10.10.0.0/16 net, gluster peer

Marcus, Can you share server-side gluster peer probe and client-side mount command-lines. On Tue, Apr 10, 2018 at 12:36 AM, Marcus Peders?n <marcus.pedersen at slu.se> wrote: > Hi all! > > I have setup a replicated/distributed gluster cluster 2 x (2 + 1). > > Centos 7 and gluster version 3.12.6 on server. > > All machines have two network interfaces and connected to

Hi again Xavi, I did some more testing on my virt machines with same setup: Number of Bricks: 1 x (2 + 1) = 3 If I do it the same way, I upgrade the arbiter first, I get the same behavior that the bricks do not start and the other nodes does not "see" the upgraded node. If I upgrade one of the other nodes (non arbiter) and restart glusterd on both the arbiter and the other the arbiter

Yes, In first server (urd-gds-001): gluster peer probe urd-gds-000 gluster peer probe urd-gds-002 gluster peer probe urd-gds-003 gluster peer probe urd-gds-004 gluster pool list (from urd-gds-001): UUID Hostname State bdbe4622-25f9-4ef1-aad1-639ca52fc7e0 urd-gds-002 Connected 2a48a3b9-efa0-4fb7-837f-c800f04bf99f urd-gds-003 Connected ad893466-ad09-47f4-8bb4-4cea84085e5b urd-gds-004

I made a recusive diff on the upgraded arbiter. /var/lib/glusterd/vols/gds-common is the upgraded aribiter /home/marcus/gds-common is one of the other nodes still on gluster 10 diff -r /var/lib/glusterd/vols/gds-common/bricks/urd-gds-030:-urd-gds-gds-common /home/marcus/gds-common/bricks/urd-gds-030:-urd-gds-gds-common 5c5 < listen-port=60419 --- > listen-port=0 11c11 <

Hi Xavi, Copy the same info file worked well and the gluster 11 arbiter is now up and running and all the nodes are communication the way they should. Just another note on something I discovered on my virt machines. All the three nodes has been upgarded to 11.0 and are working. If I run: gluster volume get all cluster.op-version I get: Option Value ------

Hi Kotresh, Yes, all nodes have the same version 4.1.1 both master and slave. All glusterd are crashing on the master side. Will send logs tonight. Thanks, Marcus ################ Marcus Peders?n Systemadministrator Interbull Centre ################ Sent from my phone ################ Den 13 juli 2018 11:28 skrev Kotresh Hiremath Ravishankar <khiremat at redhat.com>: Hi Marcus, Is the

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is

What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. Yuan Liu

During the recent week, we've worked hard to add more of the contributed international support to Asterisk. A big step was taken yesterday when Mark added international support for saynumber() to CVS. We now have a first version of support for * Danish * German * English * Swedish * Norwegian * Portuguese * Italian * French All of these require that you add your own sound files. There are