similar to: Crontab script to check health of Asterisk server?

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

Would anybody mind if the the following command options where added to MeetMeAdmin? 0 - 9, * and # I'm considering hacking the code to add these commands to play the DTMFs to the specified user as tones and hope that the SIP or IAX channels then work with these correctly. -HJC

I've just moved into 3.3v PCI servers and found that my clone X100P cards were lying about the 3.3v supported notch. Can I use a Wildcard TDM400P without any modules as a timer for MeetMe in a 64 bit 3.3v server? Will I still need to plug the hard disk power cable into it? Is there a better cheaper 3.3v MeetMe timer? (Boss doesn't trust the kernel timer.) -HJC

Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___________________________________________________________ Yahoo! Answers - Got

We are using rsync to mirror around 10 of our websites from dedicated server A to dedicated server B runing RHEL 3.0 and rysnc v2.6.3pre1 in a daemon mode. I am using the following crontab script command on dedicated server A: /usr/local/bin/rsync -avz -e "ssh -i /home/users/admin/publicKey-for-server-B" /home/serverA/site3 serverB::home/mirror Dedicated server B (where the

I'm having trouble with Polycom 501 phones that asterisk forgets how to reach them. /etc/asterisk/sip.conf: [general] context=default MusicOnHold=default port=5060 bindaddr=0.0.0.0 srvlookup=no;yes language=en dtmfmode=rfc2833 maxexpiry=600 defaultexpiry=120 [502] type=friend username=502 secret=pass host=dynamic mailbox=502@rm callerid= "Operator" <502> context=rm

Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that?

I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not been able to send useful callerid info between them (callerid becomes "serverB"). serverA register statement: (serverB has the exact opposite statement) register => serverA:serverApassword at IP_of_serverB_nic/serverB users.conf of serverA: users.conf of serverB: [serverB] [serverA] type=friend

Hello List, i would like to copy/mirror/rsync the backup files from ServerA to ServerB. Since i must assume that ServerA is hacked (public server with a few services), i would like to "pull" the files to ServerB (save server with no services). That means i do something like: rsync -avz ServerA:/backup ServerB:/backup IF ServerA gets hacked and the files get zeroed out (every file

Hi, I am trying to connect two Asterisk servers using IAX2. Everything works fine when I couple them within a LAN segment, but not when I connect them using WAN connections. I made sure that the routers' ports are mapped properly and checked this with additional ssh rules. ServerA is a Raspberry box with the vendor's Asterisk version 1.8.13.1 and ServerB is normal CentOS 7 box with

Hi, I'm using the Read command the read a DTMF tone. In this read command I play a voice-file. But now when I press one off they keys of my telephone the voice-file will stop playing a the program go the next priority. Is it possible to play the voice-file until the right DTMF tone is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V.

hi I just joined this list, as I am having really strange issue and so far been unable to find the root cause.I will give you scenario to see if anyone knows what could be the cause: 1: 2 windows 64bit enterprise server serverA and Server B 2: both on same physical network no firwalls not routers 3: both on same domain controller ( windows NT4) I can connect to serverA as under from a linux

I have a problem with Samba: I want to make trust between two Samba domains. I have setup the trust on the DOMAINB server then on the Samba DOMAINA server "net rpc trustdom establish DOMAINB" I then get the following: Password: [entered password] Could not connect to server SERVERB [this is the PDC for the DOMAINB domain] Trust to domain DOMAINB established When I then try

this is the case: serverA [ CentOs 5.6 kernel 2.6.18-238.12.1.el5.centos.plus, Samba ver. 3.5.21 ] serverB [ CentOS 5.6 kernel 2.6.18-348.1.1.el5.centos.plus, Samba ver. 3.6.6-0.129.el5 ] clientA [ WindowsXP ] clientB [ WindowsXP ] The serverA shares via Samba the resource [test] [global] workgroup = DMIT netbios name = SAMBA server string = DMIT domain server interfaces = eth0

hi all, I am trunking via iax2 2 asterisk serverses if both of them have static ip addresses, I can connect them using no password, password or auth rsa with a pair of keys. If one of them has dynamic ip address and need to register on the other server, I can connect them with no password, but I am not able to do that using keys. The question is: which is the right register syntax to use when

Hi list, I set up two asterisk servers , 1001 is the first asterisk server's sip user, and 2001 is the second asterisk server's sip user. Each of them work well, but I don't konw how to connect them. I want to let the user in 1th Asterisk can call the user in 2nd Asterisk. First asterisk server ip : 192.168.3.101 Second asterisk server ip : 192.168.3.102 can someone

All, I have 3 servers. All 3 are CentOS 5.5. All 3 have identical /etc/ssh/sshd_config files. I used ssh-keygen (with no arguments) to generate keys with no password. I then added all 3 id_rsa.pub keys to the authorized_keys file. With this set up, I should be able to ssh between all 3 boxes without needing a password. The problem is that one of the servers keeps asking for a password even with

Hi! I need some help Well i have this cenario: 1 ip04 running asterisk [A] 1 pc running asterisk [B] I nedd to make calls from A to B, and B to A. Via sip The A-B calls are working. Now I need to configure the dial plan to call B-A either to sip numbers and Fxs. Anyone can help me? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I''ve tried to implement puppetmaster High Availability (mon+heartbeat). Herefore, the puppet client and puppet master are running on both servers. When the puppet client starts up, it generates a certificate, public and private key for the machine it runs on. When the puppet master starts up, it changes something so that the puppet client have no valid certificate anymore (the

I have a small call center running with Asterisk 1.4.4 and Unicall. Everything seems to be working but twice now we had to reset the server because all lines stopped working. You can see users dialing in and reaching the queue but the agents never get the call and the lines are not released. I saw that there is a new Zaptel driver which fixes a racing condition with a TE110P card which is