similar to: 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? >

2015-03-18 10:52 GMT-06:00 ricky gutierrez <xserverlinux at gmail.com>: > 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 > 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

Hi everybody, I've a matter with the queue annoucement with the "thereare", because if I put just one member in my configuration (member => SIP/2098), the ivr gave me that I was the firt or second in the next at the queue. But the problem is, if I add one member (eg: member => SIP/2098 and member => SIP/2099), the ivr don't gave me the range but It play the

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 =>

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

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

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

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)!

2015-01-09 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

Hi list! I'm trying to configure my Asterisk to accept SIP-TLS connections, too. I followed this HowTo: http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/ But as soon I try to connect to my Asterisk using SIP-TLS I get on Asterisk-CLI: == Problem setting up ssl connection: error:140760FC:lib(20):func(118):reason(252) [Jun 5 20:16:25] WARNING[20826]: tcptls.c:669

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=>

Looking at the pastebin, the Vega device sends a CANCEL with reason: Reason: Q.850 ;cause=16. Cause 16 is normal clearing and suggests that the original caller has disconnected. I would take a look at the Vega's logs Regards, Steve On Thu, 5 Mar 2015 at 11:41 ricky gutierrez <xserverlinux at gmail.com> wrote: > > > On Wednesday, March 4, 2015, ricky gutierrez

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

I have a multi-homed machine (quite a few IP addresses on one of the interfaces) For SIP I found that using externaddr in sip.conf would make it much more reliable with ICE and RTP using the correct IP Is there an equivalent setting for XMPP / motif.conf? I saw bindaddr in gtalk.conf but it doesn't appear to be mentioned in the source code for chan_motif

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

Hi list, I try to install asterisk on vps server , but fails when I want to install dahdi [root at shark dahdi-linux-2.6.3-rc1]# make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' You do not appear to have the sources for

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

Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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

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