similar to: disk on vm with kvm

Displaying 20 results from an estimated 4000 matches similar to: "disk on vm with kvm"

2015 Mar 04
2
hangup call gw FXO
I'm having some problems with a vega sangoma, if a call comes into my ivr and hangs up, the call continues to ring and leaves hanging the channel, I have to restart Asterisk and everything works Ok my sangoma is a vega 50 , 4 FXO . I tried different tone of countries and does not work, this is the trace of which is for hanging up the channel: http://pastebin.com/y410Rhzt I was thinking
2018 Apr 26
2
cluster of 3 nodes and san
Hi list, I need a little help, I currently have a cluster with vmware and 3 nodes, I have a storage (Dell powervault) connected by FC in redundancy, and I'm thinking of migrating it to proxmox since the maintenance costs are very expensive, but the Doubt is if I can use glusterfs with a san connected by FC? , It is advisable? , I add another data, that in another site I have another cluster
2014 Nov 17
1
motif and other xmpp
Hi list, I have a big doubt!, I have some users with ejabberd and am using motif to make some calls to extensions, here works fine, the problem is when I want to send a message to another user on ejabberd and asterisk take this message as part him, like a sip message , the other user does not receive this message xmpp User A xmpp == Chat to == User B xmpp (not receive the message) look cli
2015 Mar 27
2
Gateway Eurotech
Hi, I know there are people with much experience in asterisk, and I want to ask if anyone had experiance with this gw http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/ I'm having trouble getting connect with asterisk anyone has any production? regardss -- rickygm http://gnuforever.homelinux.com
2015 Mar 12
2
GXP 1405 and asterisk
Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com
2014 Oct 01
1
JABBER_STATUS CODE 7
Hi all,I hope to find a solution with the help of the list, I'm trying to get the status of my extensions with ejabberd , the idea is to visualize my users ejabberd incoming calls or missed. I'm testing with my operator extension with this code but only get the missed call notification does not show me where the call is coming. my piece of code [operadora] exten =>
2018 Apr 27
0
cluster of 3 nodes and san
Hi, any advice? El mi?., 25 abr. 2018 19:56, Ricky Gutierrez <xserverlinux at gmail.com> escribi?: > Hi list, I need a little help, I currently have a cluster with vmware > and 3 nodes, I have a storage (Dell powervault) connected by FC in > redundancy, and I'm thinking of migrating it to proxmox since the > maintenance costs are very expensive, but the Doubt is if I can
2018 Apr 27
1
cluster of 3 nodes and san
>but the Doubt is if I can use glusterfs with a san connected by FC? Yes, just format the volumes with xfs and ready to go For a replica in different DC, be careful about latency. What is the connection between DCs? It can be doable if latency is low. On Fri, Apr 27, 2018 at 4:02 PM, Ricky Gutierrez <xserverlinux at gmail.com> wrote: > Hi, any advice? > > El mi?., 25 abr. 2018
2015 Mar 18
2
res_xmpp.c:3468 xmpp_client_reconnect:
Hi list , this is a bug? ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client regardss -- rickygm http://gnuforever.homelinux.com
2015 Jul 08
6
tls on asterisk 13
Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed to make it work, all my terminals spa Cisco 5XX look my cli [Jul 8 11:09:16] ERROR[14733]: pjsip:0 <?>: tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:16] WARNING[14733]: pjsip:0 <?>: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)!
2015 Mar 23
1
Auto Answer
Hi , I'm having some problems with functions enable auto answer in some Grandstream GXP 1405 , I have enabled this feature in the snom 821 phone and work gr8 , in the gandstream not work, I enable the function on the phone "Allow Auto Answer by Call-Info: yes Dialplan: exten => 501,1,SIPAddHeader(Call-Info: answer-after=2) exten => 501,n,Page(SIP/140&SIP/110,d) exten
2015 Feb 25
5
situation with ivr and four-channel gateway
Hi list, I need your help ,I have an incoming call x the ivr and the operator takes the call. ext "101" , If a second call reenters and the operator is talking, I want to send to the extension 102 I use the Variable DIALSTATUS , but not working check IVR [IVRINMA] exten => s,1,Wait(1) exten => s,n,Set(CHANNEL(language)=es) same=> n,Set(TIMEOUT(digit)=4) same=>
2015 Jan 09
2
SEMI OFF-TOPIC - Fail2ban
2015-01-09 3:53 GMT-06:00 Stefan Gofferje <lists at home.gofferje.net>: > > Do you really want to detect "ChallengeSent"? That should occur also on > legitimate login processes... > Hi , strange thing is that I still have not this asterisk in production and I see many attempts Connection. Now keep in mind that when a connection of authentication is successful the
2013 Nov 14
1
DAHDI with (CDR(userfield)
Hi list, I need some help to improve my cdr, now in my company are asking me how to know which of my phone numbers are most used when receiving calls from the PSTN and incoming the IVR was thinking about using userfield field, and I'm trying to do, I have at the moment 4 channel DAHDI ; DAHDI CHANNEL 3=23XXXXX6 context=in callerid=asreceived group=1 signalling=fxs_ks channel => 3
2015 Jan 08
4
SEMI OFF-TOPIC - Fail2ban
Hi list , someone on the list has seen this type of connection attempts in asterisk, fail2ban does not stop 2015-01-08 14:59:47] SECURITY[21515] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="1420750787-386840",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:100 at
2015 Jan 19
2
SEMI-OFFTOPIC openvox
Hi list, I write on the list looking for help, buy a openvox gw gsm for four channels and I'm a little disappointed with the support openvox, for some reason , The call doesn?t get trough support tells me it was my asterisk server, but does not really work me and my internal calls are working perfectly, I tested with another sangoma FXO gateway and works perfectly. the problem is that
2014 Dec 29
2
[OFF TOPIC] monit
Hi list , I'm trying to run monit with asterisk, starting as simple # My PBX Asterisk check process asterisk with pidfile /var/run/asterisk/asterisk.pid start program = "/etc/init.d/asterisk start" with timeout 60 seconds stop program = "/etc/init.d/asterisk stop" with timeout 60 seconds if failed host 127.0.0.1 port 5038 then restart if 5 restarts within 5 cycles then
2015 Feb 27
2
situation with ivr and four-channel gateway
2015-02-27 10:25 GMT-06:00 A J Stiles <asterisk_list at earthshod.co.uk>: > O.K. So what does your existing Dial() statement in extensions.conf look > like? > apology, put the gateway was sangoma but is a openvox , all my outgoing calls out for this context: [my-mobile-out] exten => _NXXXXXXX,n,Dial(SIP/1003/${EXTEN},55,rT) exten =>
2015 Feb 26
2
situation with ivr and four-channel gateway
2015-02-26 10:45 GMT-06:00 A J Stiles <asterisk_list at earthshod.co.uk>: > > You just need to use call groups. > > In your chan_extra.conf (if it's an OpenVox) or chan_dahdi.conf, add > something like > group=1 > to the definition for each span. > > Now in the [globals] section of your dialplah, have something like > MOBILE=EXTRA/r1 > for an
2015 Mar 18
1
res_xmpp.c:3468 xmpp_client_reconnect:
2015-03-18 11:13 GMT-06:00 ricky gutierrez <xserverlinux at gmail.com>: > Hi , I'm trying to apply this patch from the source asterisk > asterisk-11.16.0 and when I apply it shows me this message > > asterisk-11.16.0]#patch -p0 < refs > patch: **** Only garbage was found in the patch input. > > is the correct way to apply the patch or am I doing wrong? >