Re: [asterisk-users] tls on asterisk 13
2015-07-08 13:11 GMT-06:00 Joshua Colp jc...@digium.com: You probably want to add rewrite_contact=yes to your endpoint. This will cause it to reuse the existing connection established from the phone. Generally the port provided by the phone is not reachable. Hi Joshua , I add the option you recommended but still can not connect, the strange thing is that I get another message always using TLS transport [Jul 8 14:28:45] NOTICE[2498]: res_pjsip/pjsip_distributor.c:256 log_unidentified_request: Request from 'X00X sip:X00X@172.16.8.55' failed for '172.16.8.179:5065' (callid: 5ece51c0-9ed5173a@172.16.8.179) - No matching endpoint found --- Transmitting SIP response (479 bytes) to TLS:172.16.8.179:5065 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 172.16.8.179:5065;rport=5065;received=172.16.8.179;branch=z9hG4bK-27b9198a Call-ID: 5ece51c0-9ed5173a@172.16.8.179 From: X00X sip:X00X@172.16.8.55;tag=ff2e31b0cc3d380ao3 To: sip:172.16.8.55;tag=z9hG4bK-27b9198a CSeq: 54 NOTIFY WWW-Authenticate: Digest realm=asterisk,nonce=1436387325/20cc7b903ffd92277b22c633e27854de,opaque=5b36911758ac6b0e,algorithm=md5,qop=auth Server: Asterisk PBX 13.4.0 Content-Length: 0 regardss -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tls on asterisk 13
2015-07-08 13:09 GMT-06:00 Ryan, Travis ry...@oscarwinski.com: Asterisk13 can do native tls with each phone? Nice. any example? 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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)! err=120111 (Connection refused) [Jul 8 11:09:46] ERROR[14733]: pjsip:0 ?:tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:46] WARNING[14733]: pjsip:0 ?: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=31917 (tdta0x7f53c000dcb0)! err=120111 (Connection refused) someone has had good results with tls my config [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cert_file=/etc/asterisk/keys/asterisk.crt priv_key_file=/etc/asterisk/keys/asterisk.key method=tlsv1 [] type=endpoint context=XX-Xip disallow=all allow=ulaw allow=alaw transport=transport-tls direct_media=no force_rport=yes rtp_symmetric=yes mailboxes=@default auth= aors= media_encryption=sdes dtmfmode=rfc4733 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP-TLS
2015-06-05 14:29 GMT-06:00 Luca Bertoncello lucab...@lucabert.de: I think it is a problem on Asterisk for OpenWRT... :( Regards Luca Bertoncello (lucab...@lucabert.de) compilation problems with the module srtp , check the module module show like srtp -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP-TLS
2015-06-05 12:21 GMT-06:00 Luca Bertoncello lucab...@lucabert.de: 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 handle_tcptls_connection: FILE * open failed! And of course it does NOT connect... Any idea? Thanks Luca Bertoncello (lucab...@lucabert.de) -- Hi lucas , dou you try this: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway Eurotech
2015-03-27 10:52 GMT-06:00 Carlos Rojas crt.ro...@gmail.com: I Ricky I have worked with this gateway few years ago, it's good product, they have gateways with PRI connectors and SIP. The quality is good, and it woks good with asterisk or regular PBXs. Hi carlos , thank for your advice, I could ask a favor?, this is the trunk that I have in my asterisk and the gw tells me Unregistered [testsip] context=boss type=friend host=1.1.1.1 # ip gateway port=5060 canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw dtmfmode=rfc2833 in gateway - General - SIP client Name ip port usersecret testsip 1.1.1.1 5060 myboy my123 -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Answer
2015-03-23 11:08 GMT-06:00 ricky gutierrez xserverli...@gmail.com: 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/140SIP/110,d) exten = 501,n,Hangup() not work for me, it ring but does the function of auto answer Any idea? I found the problem, my mistake, annex the solution for someone else to help exten = 501,1,SIPAddHeader(Call-Info: answer-after=0) exten = 501,n,Dial(SIP/140SIP/137SIP/112SIP/113SIP/122SIP/120SIP/131SIP/132SIP/116SIP/136SIP/111SIP/125SIP /124) -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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/140SIP/110,d) exten = 501,n,Hangup() not work for me, it ring but does the function of auto answer Any idea? -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:
2015-03-18 12:54 GMT-06:00 ricky gutierrez xserverli...@gmail.com: 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 ' RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client ' RROR[2545]: res_xmpp.c:3468 xmpp_client_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 -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:
2015-03-18 10:52 GMT-06:00 ricky gutierrez xserverli...@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 -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:
2015-03-18 11:13 GMT-06:00 ricky gutierrez xserverli...@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? 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 ' RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client ' RROR[2545]: res_xmpp.c:3468 xmpp_client_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 -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GXP 1405 and asterisk
2015-03-12 13:07 GMT-06:00 Bryant Zimmerman brya...@zktech.com: SIPAddHeader(Alert-Info:\;info=ring3) In the phone config add the value ring3 and select Account # / Call Settings / Match Incoming Caller ID (Matching Rule) In the first rule place the word ring3 and select your ring tone. This will cause the selected ringtone to be used when calls with the info value of ring3 is matched can not get it to work any idea o tips? regardss work gr8 , thnk thnk ..Bryant -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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/310SIP/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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup call gw FXO
2015-03-05 6:11 GMT-06:00 Steve Davies davies...@gmail.com: 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 I tried to contact support sangoma, I send a log to them and they have not contacted me! ,a disappointment asterisk shows active channels, zombie type ;) , for example the extension 160 call the 122, 122 is not connected and tells me this on the phone , I have the impression that rtptimeout not working as it should http://pastebin.com/vTZ0WGqq look cli asterisk: 200.62.89.140(None) koV6foZnHTr3gEf (nothing) No Rx: REGISTER guest 200.62.89.140(None) 690e01185aa2f36 (nothing)No Rx: REGISTER guest 200.62.89.140gatewayVEGA0010-0C09-6C8EF (ulaw) No Rx: ACKgatewayVEGA 200.62.89.140(None) 5db8c434570dfb9 (nothing)No Rx: REGISTER guest 190.184.84.10(None) 3654c4f8-1fd27d (nothing)No Rx: NOTIFY guest 5 active SIP dialogs regardss -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup call gw FXO
On Wednesday, March 4, 2015, ricky gutierrez xserverli...@gmail.com wrote: 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 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 -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 that might help rpt timeout , I have put in 30s, but does not work any advice? 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] account code
Hi list , I have a question with account codes, all my outgoing calls
are authenticated, but now the boss wants to monitor these calls with
the codes.
example: maria has an extension 110, but peter was in place and use
the phone maria , maria then says that she did not make that call to
that number of cell.
like to know who made it?, I think the pin code is my friendo , my
users have a four-digit pin to authenticate, I'm thinking of using the
cdr field userfield
as I can do to read the pin code and write 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 = _92XXX,n,Dial(SIP/${EXTEN:1}@gw,40,rRT)
same = n,Busy(3)
same = n,Hangup
any idea?
regardss
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Re: [asterisk-users] situation with ivr and four-channel gateway
2015-03-02 3:44 GMT-06:00 A J Stiles asterisk_l...@earthshod.co.uk:
Ah. *Incoming* calls are not something that is within your control; they have
already been routed onto a line by your telco. So you will need to speak to
someone at your telco about doing this.
Hi Aj, I call to telco and say they can not in GSM, only on lines are analogous
As a temporary measure, you could try setting up divert-on-busy so SIM1
diverts to SIM2, SIM2 diverts to SIM3, SIM3 diverts to SIM4 and SIM4 diverts
to SIM1. You can do this with specially-crafted Dial() statements,
With asterisk or the openvox gw?
or by
temporarily inserting the SIMs in an old mobile phone. See your telco's
website for details of setting up call diversion.
these guys do not help much! .
the ivr worked perfect with DEVICE_STATE , thank john!
exten = t,1,ExecIf($[ ${DEVICE_STATE(SIP/${EXTEN})} = INUSE ]?Busy)
exten = t,n,Dial(SIP/110,38,t)
same= n,Dial(SIP/162,40,t)
same= n,Hangup()
thnk for all help.
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Re: [asterisk-users] situation with ivr and four-channel gateway
2015-02-27 10:25 GMT-06:00 A J Stiles asterisk_l...@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 = _NXXX,n,Dial(SIP/1003/${EXTEN},55,rT)
exten = _NXXX,n,Dial(SIP/1004/${EXTEN},55,rT)
exten = _NXXX,n,Dial(SIP/1001/${EXTEN},55,rT)
exten = _NXXX,n,Dial(SIP/1002/${EXTEN},55,rT)
exten = _NXXX,n,Playback(all-circuits-busy-now)
exten = _NXXX,n,Hangup()
my main number is registered on 1002 channel gsm 1
the problem is that my pbx all incoming calls using only the channel
gsm 1 , the idea is that an incoming call to channel 1 is passed to
channel 2
regardss.
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Re: [asterisk-users] situation with ivr and four-channel gateway
2015-02-25 18:23 GMT-06:00 John Kiniston johnkinis...@gmail.com:
I'd recommend using DEVICE_STATE
On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's not
'NOT_INUSE' then dial it, Otherwise dial SIP/102
exten =
101,1,ExecIf($[${DEVICE_STATE(SIP/101)}=NOT_INUSE]?Dial(SIP/101,40))
same = n,Dial(SIP/102,40,t)
same = n,Hangup()
Hi john and Steve , I do tests with advice
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Re: [asterisk-users] situation with ivr and four-channel gateway
2015-02-26 10:45 GMT-06:00 A J Stiles asterisk_l...@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 OpenVox card, or
MOBILE=DAHDI/r1
for other makes. Now you need your Dial() statements to be something like
Dial(${MOBILE}/${EXTEN},180
Calls will then be made by trying each span in turn until an available one is
found. So if you have an incoming call on span 1, Asterisk will try spans 2,
3 and 4 in turn before giving up. It also will remember which span it used
last, and start with the next one next time; so the calls should be
distributed roughly evenly across your SIMs.
For more information about this (and some other modes you can use which do
slightly different things than r), see
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
(yes, it refers to Zaptel; but the syntax is the same for DAHDI and EXTRA
channels).
Hi A J , I have a sangoma gsm gateway 4channels , not use chan dahdi
http://gnuforever.homelinux.com
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[asterisk-users] 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= n,Set(TIMEOUT(response)=5)
same= n,Wait(1)
same= n,Background(/tmp/ivr/menu)
same= n,WaitExten(5)
exten = 0,1,Playback(pls-wait-connect-call)
exten = 0,n,Goto(operadora,101,1)
exten = _10[1-3],1,Dial(SIP/${EXTEN},40,t)
same= n,Hangup
exten = i,1,Playback(invalid)
same= n,Goto(IVRINMA,s,2)
exten= t,1,Dial(SIP/101,38,t)
exten= t,n,GotoIf($[${DIALSTATUS} = BUSY]?2,1:)
exten = 2,1,Dial(SIP/102,38,t)
same= n,Hangup()
## the second option, if possible ###
I have a gw wiht 4 port gsm , my provider gives me 4 lines and one of
them is the main , the problem is that all my incoming calls using
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 : XXX1 Main Number
Channel 2 : XXX2 other
Channel 3 : XXX3 other
Channel 4 : XXX4 other
regardss
--
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http://gnuforever.homelinux.com
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[asterisk-users] 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 support openvox is Chinese and the difference in time zone is high. my trunk is connected 5001/5001X.X.X.X D Yes Yes5060 Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline] I follow this guide , but not work http://www.lojamundi.com.br/download/gateways-gsm/openvox/Quickstart_Guide_of_OpenVox_GSM_Gateway_VS-GW2120_Series_Connect_with_Asterisk_Server.pdf -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SEMI-OFFTOPIC openvox
:984783842@50.X.X.X;tag=as77fb37e2 Call-ID: 7f55e32e-e4c6e11a@172.16.8.179 CSeq: 102 INVITE Server: inmaconsa-Voice-Sip-ipbx Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-Asterisk-HangupCause: Call Rejected X-Asterisk-HangupCauseCode: 21 Content-Length: 0 2015-01-19 10:24 GMT-06:00 ricky gutierrez xserverli...@gmail.com: 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 support openvox is Chinese and the difference in time zone is high. my trunk is connected 5001/5001X.X.X.X D Yes Yes5060 Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline] I follow this guide , but not work http://www.lojamundi.com.br/download/gateways-gsm/openvox/Quickstart_Guide_of_OpenVox_GSM_Gateway_VS-GW2120_Series_Connect_with_Asterisk_Server.pdf -- rickygm http://gnuforever.homelinux.com -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SEMI OFF-TOPIC - Fail2ban
2015-01-09 9:05 GMT-06:00 Tech Support aster...@voipbusiness.us: Hello; Did you remember to uncomment the dateformat in /etc/asterisk/logger.conf? That's necessary for fail2ban to work. Logger.conf [general] dateformat=%F %T Hi , I'll show my logger dateformat=%F %T ; ISO 8601 date format use_callids= yes appendhostname= no security= security,notice 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SEMI OFF-TOPIC - Fail2ban
2015-01-09 3:53 GMT-06:00 Stefan Gofferje li...@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 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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@173.230.133.20,SessionID=0x169f528,LocalAddress=IPV4/UDP/173.230.133.20/5060,RemoteAddress=IPV4/UDP/63.141.229.58/5078,Challenge=770e84a3 [2015-01-08 15:20:20] SECURITY[21515] res_security_log.c: SecurityEvent=ChallengeSent,EventTV=1420752020-854997,Severity=Informational,Service=SIP,EventVersion=1,AccountID=sip:102@173.230.133.20,SessionID=0x169f528,LocalAddress=IPV4/UDP/173.230.133.20/5060,RemoteAddress=IPV4/UDP/198.204.241.58/5074,Challenge=23965594 I modified the fail2ban with the filter, but still not detected asterisk.conf log_prefix= \[\]\s*(?:NOTICE|SECURITY)%(__pid_re)s:?(?:\[\S+\d*\])? \S+:\d* failregex = ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - Wrong password$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - No matching peer found$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - Username/auth name mismatch$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - Device does not match ACL$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - Peer is not supposed to register$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - ACL error \(permit/deny\)$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - Not a local domain$ ^%(log_prefix)s Call from '[^']*' \(HOST:\d+\) to extension '\d+' rejected because extension not found in context 'default' \.$ ^%(log_prefix)s Host HOST failed to authenticate as '[^']*'$ ^%(log_prefix)s No registration for peer '[^']*' \(from HOST\)$ ^%(log_prefix)s Host HOST failed MD5 authentication for '[^']*' \([^)]+\)$ ^%(log_prefix)s Failed to authenticate (user|device) [^@]+@HOST\S*$ ^%(log_prefix)s (?:handle_request_subscribe: )?Sending fake auth rejection for (device|user) \d*sip:[^@]+@HOST;tag=\w+\S* $ ^%(log_prefix)s SecurityEvent=(FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPassword),EventTV=[\d-]+,Severit y=[\w]+,Service=[\w]+,EventVersion=\d+,AccountID=\d+,SessionID=0x[\da-f]+,LocalAddress=IPV[46]/(UD|TC)P/[\da-fA-F:.]+/\d+,Rem oteAddress=IPV[46]/(UD|TC)P/HOST/\d+(,Challenge=\w+,ReceivedChallenge=\w+)?(,ReceivedHash=[\da-f]+)?$ ignoreregex = -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OFF TOPIC] monit
2014-12-29 4:51 GMT-06:00 Doug Lytle supp...@drdos.info: I use monit, but I only watch the pid check process asterisk with pidfile /var/run/asterisk/asterisk.pid start program = /usr/sbin/service asterisk start stop program = /usr/sbin/service asterisk stop Doug work fine my friend , thnk -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [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 timeout when I log in (monit interface) I see the status of asterisk is NOT MONITORED port 5038 is ready netstat -an | grep 5038 tcp0 0 127.0.0.1:5038 0.0.0.0:* LISTEN someone on the list who is running successfully?, I am using asterisk 11.15 With CentOS 6.5 x64 regards list. -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] motif and other xmpp
Hi, here again, I'm going around with this problem and can not find a solution, but I put different context within xmpp.conf, asterisk believes xmpp messages between users are SIP message. any idea? 2014-11-17 16:56 GMT-06:00 ricky gutierrez xserverli...@gmail.com: 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 asterisk WARNING[20242][C-002e]: pbx.c:6646 __ast_pbx_run: Channel 'Message/ast_msg_queue' sent to invalid extension but no invalid handler: context,exten,priority=nica,s,1 any idea? -- rickygm http://gnuforever.homelinux.com -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 asterisk WARNING[20242][C-002e]: pbx.c:6646 __ast_pbx_run: Channel 'Message/ast_msg_queue' sent to invalid extension but no invalid handler: context,exten,priority=nica,s,1 any idea? -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] JABBER_STATUS CODE 7
I think asterisk does not respect this, I have added several within xmpp.conf buddy Client: ejabberd Buddy: alci...@xmpp.domain.com Resource: asterisk-xmpp node: http://www.asterisk.org/xmpp/client/caps version: asterisk-xmpp Google Talk capable: no Jingle capable: yes Client: operadora Buddy: ce...@xmpp.domain.com Resource: asterisk-xmpp node: http://www.asterisk.org/xmpp/client/caps version: asterisk-xmpp Google Talk capable: no Jingle capable: yes Buddy: ejabb...@xmpp.domain.com Buddy: alci...@xmpp.domain.com Resource: asterisk-xmpp node: http://www.asterisk.org/xmpp/client/caps version: asterisk-xmpp Google Talk capable: no Jingle capable: yes 2014-10-09 17:10 GMT-06:00 Marcelo Terres mhter...@gmail.com: Retrieves the numeric status associated with the buddy identified by jid. If the buddy does not exist in the buddylist, returns 7. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_JABBER_STATUS_res_xmpp -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] JABBER_STATUS CODE 7
2014-10-13 14:44 GMT-06:00 Matthew Jordan mjor...@digium.com:
The error message is pretty explicit about what you asked it to look for:
{quote}
acf_jabberstatus_read: Resource alcides of buddy operadora@172.16.8.59
was not found.
strange, I put the fqdn to ejabberd, and now , not shows the code 7
[Oct 13 14:53:08] WARNING[4609][C-000f]: res_xmpp.c:1617
acf_jabberstatus_read: Could not find buddy in list:
'operad...@xmpp.domain.com'
-- Executing [0@locales:1] Set(SIP/5002-0010, STATUS=) in new stack
-- Executing [0@locales:2] GotoIf(SIP/5002-0010,
0?disponible:nodisponible) in new stack
-- Goto (locales,0,6)
-- Executing [0@locales:6] JabberSend(SIP/5002-0010,
ejabberd,operad...@xmpp.domain.com,Llamada perdida de5002) in new
stack
--- XMPP sent to 'ejabberd' ---
message type='chat' to='operad...@xmpp.domain.com'
from='aster...@xmpp.domain.com/asterisk-xmpp'bodyquot;Llamada
perdida de5002quot;/body/message
-
-- Executing [0@locales:7] Hangup(SIP/5002-0010, ) in new stack
== Spawn extension (locales, 0, 7) exited non-zero on 'SIP/5002-0010'
--- XMPP received from 'operadora' ---
message from='aster...@xmpp.domain.com/asterisk-xmpp'
to='operad...@xmpp.domain.com' type='chat'bodyquot;Llamada perdida
de5002quot;/body/message
-
-- Executing [s@messages1:1] NoOp(Message/ast_msg_queue,
Mensaje hacia usuarios XMPP) in new stack
-- Executing [s@messages1:2] JabberSend(Message/ast_msg_queue,
ejabberd,allan@172.16.8.59,Llamada perdida de5002) in new stack
--- XMPP sent to 'ejabberd' ---
message type='chat' to='allan@172.16.8.59'
from='aster...@xmpp.domain.com/asterisk-xmpp'bodyquot;Llamada
perdida de5002quot;/body/message
-
-- Executing [s@messages1:3] NoOp(Message/ast_msg_queue, Estado
del mensaje ) in new stack
-- Executing [s@messages1:4] Hangup(Message/ast_msg_queue, )
in new stack
== Spawn extension (messages1, s, 4) exited non-zero on
'Message/ast_msg_queue'
Do you have a buddy operadora@172.16.8.59 with a resource of alcides?
Based on the provided output, it does not appear as if you have that
buddy/resource combination, in which case the result of 7 is what I
would expect.
I have put it in both
Client: alcides
Buddy: ce...@xmpp.domain.com
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Buddy: aster...@xmpp.domain.com
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Buddy: operad...@xmpp.domain.com
Resource: 36500272461413222444766262
node: http://pidgin.im/
version: I22W7CegORwdbnu0ZiQwGpxr0Go=
Google Talk capable: no
Jingle capable: yes
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Client: operadora
Buddy: ce...@xmpp.domain.com
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Buddy: operad...@xmpp.domain.com
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Resource: 36500272461413222444766262
node: http://pidgin.im/
version: I22W7CegORwdbnu0ZiQwGpxr0Go=
Google Talk capable: no
Jingle capable: yes
Buddy: ejabb...@xmpp.domain.com
Buddy: alci...@xmpp.domain.com
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Buddy: rica...@xmpp.domain.com
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Buddy: ad...@xmpp.domain.com
Resource: asterisk-xmpp
node:
Re: [asterisk-users] JABBER_STATUS CODE 7
Marcelo but now the code does not show it, is empty
xmpp show connections
Jabber Users and their status:
[admin] ad...@xmpp.domain.com - Connected
[ricardo] rica...@xmpp.domain.com - Connected
[alcides] alci...@xmpp.domain.com - Connected
[allan] al...@xmpp.domain.com - Connected
[cesar] ce...@xmpp.domain.com - Connected
[operadora] operad...@xmpp.domain.com - Connected
[ejabberd] aster...@xmpp.domain.com - Connected
2014-10-13 15:27 GMT-06:00 Marcelo Terres mhter...@gmail.com:
You always need to use your jabber domain in jabberid.
Regards,
Marcelo H. Terres
mhter...@gmail.com
IM: marc...@jabber.mundoopensource.com.br
http://www.mundoopensource.com.br
http://offtopicsandfun.blogspot.com
http://biertasters.blogspot.com
http://twitter.com/mhterres
On Mon, Oct 13, 2014 at 6:06 PM, ricky gutierrez xserverli...@gmail.com
wrote:
2014-10-13 14:44 GMT-06:00 Matthew Jordan mjor...@digium.com:
The error message is pretty explicit about what you asked it to look for:
{quote}
acf_jabberstatus_read: Resource alcides of buddy operadora@172.16.8.59
was not found.
strange, I put the fqdn to ejabberd, and now , not shows the code 7
[Oct 13 14:53:08] WARNING[4609][C-000f]: res_xmpp.c:1617
acf_jabberstatus_read: Could not find buddy in list:
'operad...@xmpp.domain.com'
-- Executing [0@locales:1] Set(SIP/5002-0010, STATUS=) in new
stack
-- Executing [0@locales:2] GotoIf(SIP/5002-0010,
0?disponible:nodisponible) in new stack
-- Goto (locales,0,6)
-- Executing [0@locales:6] JabberSend(SIP/5002-0010,
ejabberd,operad...@xmpp.domain.com,Llamada perdida de5002) in new
stack
--- XMPP sent to 'ejabberd' ---
message type='chat' to='operad...@xmpp.domain.com'
from='aster...@xmpp.domain.com/asterisk-xmpp'bodyquot;Llamada
perdida de5002quot;/body/message
-
-- Executing [0@locales:7] Hangup(SIP/5002-0010, ) in new stack
== Spawn extension (locales, 0, 7) exited non-zero on 'SIP/5002-0010'
--- XMPP received from 'operadora' ---
message from='aster...@xmpp.domain.com/asterisk-xmpp'
to='operad...@xmpp.domain.com' type='chat'bodyquot;Llamada perdida
de5002quot;/body/message
-
-- Executing [s@messages1:1] NoOp(Message/ast_msg_queue,
Mensaje hacia usuarios XMPP) in new stack
-- Executing [s@messages1:2] JabberSend(Message/ast_msg_queue,
ejabberd,allan@172.16.8.59,Llamada perdida de5002) in new stack
--- XMPP sent to 'ejabberd' ---
message type='chat' to='allan@172.16.8.59'
from='aster...@xmpp.domain.com/asterisk-xmpp'bodyquot;Llamada
perdida de5002quot;/body/message
-
-- Executing [s@messages1:3] NoOp(Message/ast_msg_queue, Estado
del mensaje ) in new stack
-- Executing [s@messages1:4] Hangup(Message/ast_msg_queue, )
in new stack
== Spawn extension (messages1, s, 4) exited non-zero on
'Message/ast_msg_queue'
Do you have a buddy operadora@172.16.8.59 with a resource of alcides?
Based on the provided output, it does not appear as if you have that
buddy/resource combination, in which case the result of 7 is what I
would expect.
I have put it in both
Client: alcides
Buddy: ce...@xmpp.domain.com
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Buddy: aster...@xmpp.domain.com
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Buddy: operad...@xmpp.domain.com
Resource: 36500272461413222444766262
node: http://pidgin.im/
version: I22W7CegORwdbnu0ZiQwGpxr0Go=
Google Talk capable: no
Jingle capable: yes
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Client: operadora
Buddy: ce...@xmpp.domain.com
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Buddy: operad...@xmpp.domain.com
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version: asterisk-xmpp
Google Talk capable: no
Jingle capable: yes
Re: [asterisk-users] JABBER_STATUS CODE 7
anyone here?
2014-10-01 8:09 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
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 =
0,1,Set(STATUS=${JABBER_STATUS(ejabberd,operadora@172.16.8.59/alcides)})
same= n, GotoIf($[0${STATUS} = 1]?disponible:nodisponible)
same= n(disponible),
JabberSend(ejabberd,operadora@172.16.8.59,Llamada Entrante
${CALLERID(num)})
same= n,Dial(SIP/5001)
same= n,Hangup()
same= n(nodisponible),
JabberSend(ejabberd,operadora@172.16.8.59,Llamada perdida de
${CALLERID(num)}
)
same= n,Hangup()
look the log
Oct 1 08:04:10] NOTICE[4789][C-0028]: res_xmpp.c:1631
acf_jabberstatus_read: Resource alcides of buddy operadora@172.16.8.59
was not found.
-- Executing [0@locales:1] Set(SIP/5002-0029, STATUS=7) in new
stack
-- Executing [0@locales:2] GotoIf(SIP/5002-0029,
0?disponible:nodisponible) in new stack
-- Goto (locales,0,6)
-- Executing [0@locales:6] JabberSend(SIP/5002-0029,
ejabberd,operadora@172.16.8.59,Llamada perdida de 5002) in new
stack
[Oct 1 08:04:34] WARNING[13482][C-0005]: pbx.c:6646
__ast_pbx_run: Channel 'Message/ast_msg_queue' sent to invalid
extension but no invalid handler: context,exten,priority=default,s,1
not work for me, and I think this should work asterisk receiving presence
status
--- XMPP received from 'operadora' ---
presence from='operadora@172.16.8.59/12233853371412171752845116'
to='operadora@172.16.8.59/asterisk-xmpp'showchat/showpriority1/priorityc
xmlns='http://jabber.org/protocol/caps' node='http://pidgin.im/'
hash='sha-1' ver='I22W7CegORwdbnu0ZiQwGpxr0Go='/x
xmlns='vcard-temp:x:update'photo//x/presence
-
any idea?
regardss
--
rickygm
http://gnuforever.homelinux.com
--
rickygm
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[asterisk-users] JABBER_STATUS CODE 7
Hi all,I hope to find a solution with the help of the list, I'm trying
to get the status of my extensions with ejabberd , the idea is to
visualize my users ejabberd incoming calls or missed.
I'm testing with my operator extension with this code but only get the
missed call notification does not show me where the call is coming.
my piece of code
[operadora]
exten =
0,1,Set(STATUS=${JABBER_STATUS(ejabberd,operadora@172.16.8.59/alcides)})
same= n, GotoIf($[0${STATUS} = 1]?disponible:nodisponible)
same= n(disponible),
JabberSend(ejabberd,operadora@172.16.8.59,Llamada Entrante
${CALLERID(num)})
same= n,Dial(SIP/5001)
same= n,Hangup()
same= n(nodisponible),
JabberSend(ejabberd,operadora@172.16.8.59,Llamada perdida de
${CALLERID(num)}
)
same= n,Hangup()
look the log
Oct 1 08:04:10] NOTICE[4789][C-0028]: res_xmpp.c:1631
acf_jabberstatus_read: Resource alcides of buddy operadora@172.16.8.59
was not found.
-- Executing [0@locales:1] Set(SIP/5002-0029, STATUS=7) in new stack
-- Executing [0@locales:2] GotoIf(SIP/5002-0029,
0?disponible:nodisponible) in new stack
-- Goto (locales,0,6)
-- Executing [0@locales:6] JabberSend(SIP/5002-0029,
ejabberd,operadora@172.16.8.59,Llamada perdida de 5002) in new
stack
[Oct 1 08:04:34] WARNING[13482][C-0005]: pbx.c:6646
__ast_pbx_run: Channel 'Message/ast_msg_queue' sent to invalid
extension but no invalid handler: context,exten,priority=default,s,1
not work for me, and I think this should work asterisk receiving presence status
--- XMPP received from 'operadora' ---
presence from='operadora@172.16.8.59/12233853371412171752845116'
to='operadora@172.16.8.59/asterisk-xmpp'showchat/showpriority1/priorityc
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
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Re: [asterisk-users] chan_motify / res_xmpp bind address?
Hi, I've been trying to talk xmpp with asterisk with ICE-UDP, but still does not work 2014-07-18 7:26 GMT-06:00 Daniel Pocock dan...@pocock.pro: 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 -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] try to work asterisk 11.11 with ice-upd
I have configured support for ice in sip.conf, and made a connection with motif to jingle, but does not work for me [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955 jingle_interpret_ice_udp_transport: Received ICE-UDP transport information on session '8b4hdffbt37vg' but ICE support not available -- Executing [s@xmpp-in:1] NoOp(Motif/allan-ce76, llamada de usuario XMPP ) in new stack motif.conf [jingle] context=xmpp-in transport=ice-udp allow=ulaw allow=alaw allow=h263 allow=h264 connection=admin any idea? -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] try to work asterisk 11.11 with ice-upd
#rpm -qa | grep uuid uuid-1.6.1-10.el6.x86_64 libuuid-2.17.2-12.14.el6_5.x86_64 uuid-devel-1.6.1-10.el6.x86_64 and res_rtp_asterisk was added in the compilation rtp.conf rtpstart=1 rtpend=2 icesupport=yes 2014-07-15 12:19 GMT-06:00 Joshua Colp jc...@digium.com: ricky gutierrez wrote: I have configured support for ice in sip.conf, and made a connection with motif to jingle, but does not work for me [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955 jingle_interpret_ice_udp_transport: Received ICE-UDP transport information on session '8b4hdffbt37vg' but ICE support not available -- Executing [s@xmpp-in:1] NoOp(Motif/allan-ce76, llamada de usuario XMPP ) in new stack Do you have the uuid development library installed? It is an optional dependency and without it res_rtp_asterisk will not be built with ICE support. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] try to work asterisk 11.11 with ice-upd
I'm reading the wiki and says that by default is active, I have it set in sip.conf and rtp.conf icesupport=yes Usage By default ICE support is enabled in res_rtp_asterisk. It can be explicitly disabled by setting icesupport to no in the rtp.conf configuration file. Icon ICE support is only used for communication between a remote endpoint and Asterisk. It is not used when directmedia is enabled and active for a session. The rtp.conf configuration file also now contains settings for a STUN server and TURN server. If these settings are not set support for the respective item is disable. https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support 2014-07-15 12:41 GMT-06:00 ricky gutierrez xserverli...@gmail.com: #rpm -qa | grep uuid uuid-1.6.1-10.el6.x86_64 libuuid-2.17.2-12.14.el6_5.x86_64 uuid-devel-1.6.1-10.el6.x86_64 and res_rtp_asterisk was added in the compilation rtp.conf rtpstart=1 rtpend=2 icesupport=yes 2014-07-15 12:19 GMT-06:00 Joshua Colp jc...@digium.com: ricky gutierrez wrote: I have configured support for ice in sip.conf, and made a connection with motif to jingle, but does not work for me [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955 jingle_interpret_ice_udp_transport: Received ICE-UDP transport information on session '8b4hdffbt37vg' but ICE support not available -- Executing [s@xmpp-in:1] NoOp(Motif/allan-ce76, llamada de usuario XMPP ) in new stack Do you have the uuid development library installed? It is an optional dependency and without it res_rtp_asterisk will not be built with ICE support. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- rickygm http://gnuforever.homelinux.com -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] try to work asterisk 11.11 with ice-upd
chan_jingle2 is supported in Asterisk 11? 2014-07-15 13:28 GMT-06:00 ricky gutierrez xserverli...@gmail.com: I'm reading the wiki and says that by default is active, I have it set in sip.conf and rtp.conf icesupport=yes Usage By default ICE support is enabled in res_rtp_asterisk. It can be explicitly disabled by setting icesupport to no in the rtp.conf configuration file. Icon ICE support is only used for communication between a remote endpoint and Asterisk. It is not used when directmedia is enabled and active for a session. The rtp.conf configuration file also now contains settings for a STUN server and TURN server. If these settings are not set support for the respective item is disable. https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support 2014-07-15 12:41 GMT-06:00 ricky gutierrez xserverli...@gmail.com: #rpm -qa | grep uuid uuid-1.6.1-10.el6.x86_64 libuuid-2.17.2-12.14.el6_5.x86_64 uuid-devel-1.6.1-10.el6.x86_64 and res_rtp_asterisk was added in the compilation rtp.conf rtpstart=1 rtpend=2 icesupport=yes 2014-07-15 12:19 GMT-06:00 Joshua Colp jc...@digium.com: ricky gutierrez wrote: I have configured support for ice in sip.conf, and made a connection with motif to jingle, but does not work for me [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955 jingle_interpret_ice_udp_transport: Received ICE-UDP transport information on session '8b4hdffbt37vg' but ICE support not available -- Executing [s@xmpp-in:1] NoOp(Motif/allan-ce76, llamada de usuario XMPP ) in new stack Do you have the uuid development library installed? It is an optional dependency and without it res_rtp_asterisk will not be built with ICE support. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- rickygm http://gnuforever.homelinux.com -- rickygm http://gnuforever.homelinux.com -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
