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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

...g 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? > > regardss > I'm confused this is not a patch, it's just garbage ;), I'm making a connection xmpp with asterisk and not connected, at the cli shows me the message every second: RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client '...

...nd 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 that might help rpt timeout , I have put in 30s, but does not work any advice? regardss -- rickygm http://gnuforever.homelinux.com

...one 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

...isk.crt priv_key_file=/etc/asterisk/keys/asterisk.key method=tlsv1 [XXXX] type=endpoint context=XX-Xip disallow=all allow=ulaw allow=alaw transport=transport-tls direct_media=no force_rport=yes rtp_symmetric=yes mailboxes=XXXX at default auth=XXXX aors=XXXX media_encryption=sdes dtmfmode=rfc4733 regardss -- rickygm http://gnuforever.homelinux.com

...te it there? [call-out-analog] exten => _9.,1,Authenticate(/var/lib/asterisk/key.txt,am,4) exten => 9.,n,Set(CDR(userfield)=pin-users)}) exten => _9.,n,Set(__SIP_CODEC=alaw) exten => _92XXXXXXX,n,Dial(SIP/${EXTEN:1}@gw,40,rRT) same => n,Busy(3) same => n,Hangup any idea? regardss -- rickygm http://gnuforever.homelinux.com

...and does not work, >> >> this is the trace of which is for hanging up the channel: >> >> http://pastebin.com/y410Rhzt >> >> I was thinking that might help rpt timeout , I have put in 30s, but >> does not work >> >> any advice? >> >> regardss >> >> >> >> something strange, I have some extensions not connected to Asterisk and > if I call, I get the message busy, the version I'm using is asterisk 11.15 > > > -- > rickygm > > http://gnuforever.homelinux.com > -- > ____________________...

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 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? regardss --...

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

...is successful the message changes and is not exactly what you mention: ## SecurityEvent="SuccessfulAuth",EventTV="1420832883-140932",#### I think this type of connection attempts messages with my asterisk that fail2ban not detected. I'm no expert, but the log not lie ;) regardss -- rickygm http://gnuforever.homelinux.com

...ty>1</priority><c xmlns='http://jabber.org/protocol/caps' node='http://pidgin.im/' hash='sha-1' ver='I22W7CegORwdbnu0ZiQwGpxr0Go='/><x xmlns='vcard-temp:x:update'><photo/></x></presence> <-------------> any idea? regardss -- rickygm http://gnuforever.homelinux.com

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.

...t_reconnect: No XMPP connection >> available when trying to connect client ' >> [2015-03-18 12:53:49] ERROR[2545]: res_xmpp.c:3468 >> xmpp_client_reconnect: No XMPP connection available when trying to >> >> I hope not bother to write directly matt >> >> regardss > Hi , any help , any info? > > regardss > > >

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

...g this number and is always busy , and the other three are always free, it is possible that the call is transferred to another channel? Channel 1 : XXXXXXX1 "Main Number" Channel 2 : XXXXXXX2 "other" Channel 3 : XXXXXXX3 "other" Channel 4 : XXXXXXX4 "other" regardss -- rickygm http://gnuforever.homelinux.com

...is successful the message changes and is not exactly what you mention: ## SecurityEvent="SuccessfulAuth",EventTV="1420832883-140932",#### I think this type of connection attempts messages with my asterisk that fail2ban not detected. I'm no expert, but the log not lie ;) regardss -- rickygm http://gnuforever.homelinux.com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/he...

...gt; > 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 that might help rpt timeout , I have put in 30s, but > does not work > > any advice? > > regardss > > > > something strange, I have some extensions not connected to Asterisk and if I call, I get the message busy, the version I'm using is asterisk 11.15 -- rickygm http://gnuforever.homelinux.com -------------- next part -------------- An HTML attachment was scrubbed... URL:...

...one 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-...

...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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every...

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