Re: [asterisk-users] if statement recording - after hours
There are multiple ways to do time-of-day routing.
ExecIf w/ IFTIME, GotoIfTime, and ExecIfTime.
I put some examples below.
Sincerely,
Brian LaVallee
On 9/12/14, 10:05, Eric Wieling wrote:
See ExecIf in the output of core show applications. The IF function might be useful,
see core show functions. I assume the Asterisk Book also covers this.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Thursday, September 11, 2014 5:43 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] if statement recording - after hours
In my dial plan I have these two lines:
exten =
_NXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten = _NXX,n,MixMonitor(${recordfilename},b)
[main]
exten = _NXX,1,NoOP(Check Time)
same = GotoIfTime(9:00-17:00,mon-fri,*,*?open:closed)
[open]
exten = _NXX,1,NoOP(Normal Call)
[closed]
exten = _NXX,1,NoOP(Take a Message)
; end
[main]
exten = _NXX,1,NoOP(Check Time)
same = n,ExecIf(${IFTIME(9:00-17:00,mon-fri,*,*?1:0)}?Goto(closed))
same = n,NoOp(Process Normal Call)
[closed]
exten = _NXX,1,NoOP(Take a Message)
; end
[main]
exten = _NXX,1,NoOP(Check Time)
same = n,ExecIfTime(9:00-17:00,mon-fri,*,*?open)
same = n,NoOp(Take a Message)
[open]
exten = _NXX,1,NoOP(Normal Call)
; end
How to add if statement to execute these line only after let say 5pm. To
record conversation only after 5pm.
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[asterisk-users] FYI: Block Comments
Hello, Here's a fun issue that recently caused me some serious heartache. Hope this helps others from making the same mistake. Did you know that the configuration parser supports block-comments. Like an idiot, I've been highlighting text between dashes. [example] ; -- Most of my comments were single-line like this -- exten = _1X.,1,NoOp(This would load fine) ; ;-- I did not even use block-comments like this one in my config file --; ; exten = _2X.,1,NoOp(Loads OK) ; ;-- But look! -- I missed a space between the semi-colon and two dashes -- exten = _3X.,1,NoOp(This would NOT load) ; -- The parser stopped loading anything past the above mistake -- ; -- Missing that space started a block-comment - Arghhh! -- exten = _4X.,1,NoOp(This would NOT load either) ; -end Guess I have to change my highlight syntax, avoiding dashes in the future. Sincerely, Brian LaVallee -- _ -- 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] Error: 'LENGTH' is not a recognized built-in function name
On 8/11/14, 11:31, Matthew Jordan wrote: On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat deepaksingh.ra...@gmail.com wrote: On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat deepaksingh.ra...@gmail.com wrote: Hi, I modified the query in res/res_config_odbc.c. Original: SELECT MAX(LENGTH(var_val)) FROM %s WHERE filename='%s' Modified: SELECT MAX(LEN(var_val)) FROM %s WHERE filename='%s' I rebuilt the code and installed Asterisk again. Now static realtime is working. Should I file a bug report? You're more than welcome to; at the same time, the number of people using MS SQL Server with Asterisk is not tremendously high - at least when compared with the alternatives. Unfortunately, this is one place where making things compatible is problematic: MySQL and PostgreSQL (which are far more likely to be used with Asterisk) along with Oracle use LENGTH, not LEN. Another option, just add LENGTH as a user-defined function to MS SQL. ||-- Something like this...| CREATE FUNCTION LENGTH (@input) ||BEGIN RETURN ( LEN(@input) ) END| Your solution, as it is currently, wouldn't be acceptable, as it would cause far more problems than it would solve. About the only way I could see solving this would be to make it configurable some place. Given the relatively few number of people who use MS SQL Server, I wouldn't expect this issue to receive a lot of attention without a patch. -- _ -- 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] The plain old PBX functionality
On 8/8/14, 14:05, Gergo Csibra wrote: Hi, back in the old analog telephony days there was digital PBX-es and digital system phonesets. This phonesets have had many individual illuminatable buttons connected with extensions. The PBX can show on the buttons if some extension is ringing (blinks) or busy (constant light), and the user can transfer the call with one touch (pressing one of this button). Because of the peer-to-peer nature of SIP, many of the digital PBX features can be difficult to reproduce. If you consider where the 'brains' of the system reside, you can see the reason. In the traditional digital PBX, all functionality was controlled by the PBX itself. It was a Master/Slave communication model. Phones were basically dumb terminals, how a button functioned was determined by the digital PBX. With SIP, phones and servers are peers. Master/Slave roles don't exist with SIP. Control is determined by the device that initiated the session. I will not go into the pro's and con's. But by dialing a URL, it's possible to entirely exclude 'the server' from a call. I search this functionality in Asterisk. What versions, and what extension functions (or other settings), and what VoIP phones can do this? The key thing in the SIP architecture to understand, the server DOES NOT control the phone. How a button functions depends on how each individual phone is configured. How a phone reacts to an instruction, depends on how the phone is configured. While it's possible to host a phone configuration template on the Asterisk server for all phones to use, it's actually independent from the Asterisk software. Depending on the make/model of the phone, most of the basic features (hold, transfer, redial) are available by default. To duplicate the digital PBX features you're looking for, will involve two groups of settings. Configuration on the server -and- configuration on the phone. SIP phones are NOT dumb terminals, you have to configure them to operate how you want. Sincerely, Brian LaVallee -- _ -- 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] OPTIONS Request without username - Forbidden
Hi Rafael, It's nothing to worry about -and- you might not be able to fix it. But it's nothing to worry about. -- Asterisk is using OPTIONS like a ping, qualify=yes. Since 403 is a *valid* SIP reply, the remote SIP service is considered reachable. My carrier replies with 405 Method Not Allowed, but it still indicates the SIP connection is up and working. -- Some carriers do not support OPTIONS. This is normally due to a proxy or other security mechanisms. Remember, OPTIONS is a request for what commands will be accepted. Sometime, you just don't want to advertise that kind of information. -- Check an INBOUND call (INVITE) and it will typically show what the carrier allows. If OPTIONS is not listed, there's nothing you can do. IP CARRIER_IP.sip LOCAL_IP.sip: UDP, length 870 E.@.9.9:=...j.p.n$BINVITE sip:212555@LOCAL_IP:5060 SIP/2.0 Via: SIP/2.0/UDP CARRIER_IP:5060;branch=z9hG4bKdac2492a2a1a086867cfb73fb2b5c8ac Via: SIP/2.0/UDP PROXY_IP:5060;branch=z9hG4bK09B55db052ffec696bd From: sip:212555@PROXY_IP:5060;tag=gK094dc1e4 To: sip:212555@CARRIER_IP:5060;tag=as2953dd14 Call-ID: 1980326667_35899190@PROXY_IP CSeq: 7852 INVITE Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,PRACK,UPDATE snip Accept: application/sdp Sincerely, Brian LaVallee On 6/25/14, 11:30 PM, Rafael Visser wrote: Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is 403 Forbidden. Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt (pg 67), we found carol, so it makingme see that i am missing some config. OPTIONS sip:ca...@chicago.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877 Max-Forwards: 70 To: sip:ca...@chicago.com Is it wright? How can i instruct FREEPBX to send the username in the option request? Sorry for this silly question but a found no answer googling. Thans in advance. rv This is the debug of the case Reliably Transmitting (NAT) to 201.217.31.XX:5060: OPTIONS sip:201.217.31.10 SIP/2.0 Via: SIP/2.0/UDP 18x.16.204.XXX:6060;branch=z9hG4bK1d8715df;rport Max-Forwards: 70 From: Unknown sip:59x212376...@186.16.204.xxx:6060;tag=as4491c6af To: sip:201.217.31.10 Contact: sip:59x212376...@18x.16.204.xxx:6060 Call-ID: 4f02699e2632410c359e1ee43a021...@186.16.204.xxx:6060 CSeq: 102 OPTIONS User-Agent: FPBX-2.11.0(1.8.25.0) Date: Wed, 25 Jun 2014 13:47:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- SIP read from UDP:201.217.31.XX:5060 --- SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 18x.16.204.XXX:6060;received=18x.16.204.XXX;branch=z9hG4bK1d8715df;rport=5060 From: Unknown sip:59x212376...@18x.16.204.xxx:6060;tag=as4491c6af To: sip:201.217.31.XX;tag=aprqngfrt-nm50ea1c6 Call-ID: 4f02699e2632410c359e1ee43a021...@18x.16.204.xxx:6060 CSeq: 102 OPTIONS This is the peer. * Name : desde-XopaXo-2376XXX Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-trunk Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 201.217.31.10 Addr-IP : 201.217.31.10:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 595212376458 SIP Options : timer Codecs : 0xe (gsm|ulaw|alaw) Codec Order : (ulaw:20,alaw:20,gsm:20) Auto-Framing : No Status : OK (36 ms) Useragent: Reg. Contact : Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No * Name : desde-XopaXo-2376XXX Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-trunk Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres
Re: [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info
There seem to be a number of places number presentation could go wrong. Since the PRI 'used to' show toll-free numbers correctly, you need to look at the gateway. Can you debug the ISDN message on the gateway? See how the toll-free is being sent to the carrier. Since you are looking to manipulate the ISDN message via SIP, it all comes down to how the gateway handles the desired functions. Sincerely, Brian LaVallee On 6/26/14, 11:24 PM, Positively Optimistic wrote: We're using a Earthlink PRI converted to SIP via a MediaGateway. I assume the mediagateway will convert the headers to something that PRI can understand. On Thu, Jun 26, 2014 at 9:22 AM, Eric Wieling ewiel...@nyigc.com wrote: It depends on your carrier.With some carriers, such as Verizon SIP, you do this using P-Asserted-Identity. With Verizon SIP, if they can’t figure out how to bill the call, it will be rejected. With Level 3 SIP, you can use From: or PAID but if the number you present to them is not on your account, they will bill the call against the “failsafe” number on your account. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Positively Optimistic *Sent:* Thursday, June 26, 2014 10:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header for presentation? We can't possibly be the only people in the world that has faced this challenge. Searching the internet has provided limited results. -- _ -- 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 -- _ -- 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] Multiple Servers: Multiple Peers: call-limit
I would like to enforce call-limit across multiple servers. Is there
any way to pass a call-limit variable between servers 01 02, as shown
below? Use a global call-limit between multiple servers and peer
connections.
A -- 01 -- Z
A -- 02 -- Z
A is using round-robin to reach Z, but in the event that 01 or 02 fail,
I want the full call-limit available to A. The call-limit is only
applied between A and the middle servers. For the sake of discussion,
let's say call-limit=10 for both, and the total limit should also be 10.
Since my round-robin configuration will fall-back to the other server,
calls can reach a maximum of 20. Not a state I want to allow.
#server_a_extensions.conf
[SERVER01]
exten = _X.,1,NoOp(Use: First Server)
same = n,Dial(SIP/A-to-01-to-Z/${EXTEN})
same = n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?SERVER02,${EXTEN},1)
same = n,NoOp(yes, it's incomplete)
[SERVER02]
exten = _X.,1,NoOp(Use: Second Server)
same = n,Dial(SIP/A-to-02-to-Z/${EXTEN})
same = n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?SERVER01,${EXTEN},1)
same = n,NoOp(yes, it's incomplete)
I've though about passing the variable between the middle servers in a
SIP message, side communication channel. But, hoping there might be a
simpler solution.
Sincerely,
Brian LaVallee
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Re: [asterisk-users] T1 Card RED ALARM
Since there are a number of setting that could be causing the alarm, AMI/B8ZS, SF/ESF, etc... Start with a loop-test, make sure the card can communicate with itself (using the current settings). Connect the following pins: 01 (RX-) -- 04 (TX+) 02 (RX+) -- 05 (TX-) Sincerely, Brian LaVallee On 6/25/14, 1:16 PM, arun kumar wrote: Cables are workig fine in my other box. On 25 Jun 2014 00:46, Steve Totaro stot...@totarotechnologies.com wrote: Remember to always check your cables first. Thanks, Steve T On Tue, Jun 24, 2014 at 1:47 PM, arun kumar arunvsadni...@gmail.com wrote: Thank you Josh for your valuable reply. I will do try changing the server and let you know what happening. ~Arun On Tue, Jun 24, 2014 at 8:39 PM, Josh Metzger joshdmetz...@gmail.com wrote: On Tue, Jun 24, 2014 at 5:25 AM, arun kumar arunvsadni...@gmail.com wrote: Hello All, I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do connect T1 lines it goes in RED. When I do connect the same line on a different Server (Same Model T1 Card) it works fine. How do I examine/diagnose my T1 Card for any hardware failures. I heard about loopback test , how helpful it is? Here are my configuration /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured Thanks ~Arun It could still be some sort of system config issue, even if you think everything is configured the same. Have you tried moving the T1 card from the Bad system to the good system? That will at least help narrow down if it's a bad card / port, or a config issue. -Josh -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] Play announcement only once in a Call Queue after 10 seconds
Hi Jonas, While I don't work with queues, but you could playback announce-holdtime before putting the caller into the queue. exten = _X.,1,NoOp(Post Queue Announcement) same = n,Answer() same = n,Wait(10) same = n,Playback(announce-holdtime) same = n,Queue(real_queue) Brian On 6/25/14, 10:11 PM, Jonas Kellens wrote: Hello, how can I create the following scenario : I have a Call Queue and I want to play an announcement, but only once after about 10 seconds. The current option |periodic| |-| |announce| |-| |frequency| keeps on playing the announcement indefinitely. (it should have an option 'once' like the option |announce-holdtime|) Kind regards, Jonas. -- _ -- 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] Maximum number of users
Hi Bilal,
Assuming you have the latest hardware, sufficient memory, cpu, etc...
The key to determine the maximum number of users comes down to the
office type, RTP path, network interface, and primary codec used.
First we need to determine the over-subscription rate, how many people
will be using the phones at any given time.
For a call center, the ratio is 1:1.
For a normal office, the industry standard is 4:1.
{This ratio is also used to determine the number of PSTN channels you
will need too}
Will the PSTN connections be Digium card(s) in your server or external
gateway(s)?
Assuming Diguim card(s), the RTP will be going through your server.
Determine the network interface. 10/100/1000baseT
Then we need to consider the largest codec used, and divide the
available bandwidth by the typical packet size.
µ-law/A-law is roughly 80 kbps, so we can support 128/1280/13107 audio
streams.
Divide that by 2 (just to be safe) and allow RTP in both directions.
64/640/6553
Now multiple the result by the over-subscription ratio. 4:1 =
256/2560/26212
So we see that the maximum number of users is 2560 for a normal office
when there is a 100baseT NIC in your Asterisk server.
You would also need to have 640 channels (28 T1 PRI's) connecting to the
PSTN.
/Using SIP trunks to connect to the PSTN through the same 100baseT NIC
will reduce the maximum number of users you can support./
The real challenge is not supporting thousands of users (IP Phones),
it's connecting a sufficient number of PSTN connections to support those
users.
Sincerely,
Brian LaVallee
On 12/18/13, 11:45 PM, bilal ghayyad wrote:
Hello;
Can someone advise me what is the maximum number of users (IP Phones)
that can be supported by asterisk 1.8 or later?
Regards
Bilal
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Re: [asterisk-users] Multiple IAX2 Trunks Load balancing
Are you looking for something like this?
Note: This will continuously go between the two trunks until the caller
hangs up, can be fixed by adding loop counter.
;
; extensions.conf
;
[LOADBALANCE]
exten = _X.,1,NoOp(Connect to least used trunk)
; - show active count
exten = _X.,n,NoOp(Calls:
${GROUP_COUNT(TRUNK01CNT)}/${GROUP_COUNT(TRUNK02CNT)} of
${MATH(${GROUP_COUNT(TRUNK01CNT)}+${GROUP_COUNT(TRUNK02CNT)},int)})
; - goto least used trunk
exten = _X.,n,GotoIf($[${GROUP_COUNT(TRUNK01CNT)}
${GROUP_COUNT(TRUNK02CNT)}]?TRUNK02,${EXTEN},1:TRUNK01,${EXTEN},1)
;
;
[TRUNK01]
exten = _X.,1,NoOp(Using Trunk 01)
; - set trunk used counter
exten = _X.,n,Set(GROUP()=TRUNK01CNT)
; - dial trunk
exten = _X.,n,Dial(IAX/T01/${EXTEN})
; - add loop counter to stop infinite loop
exten = _X.,n,NoOp(Use next TRUNK02is congestion or chanunavial)
; - next trunk if CONGESTION
exten = _X.,n,GotoIf($[${DIALSTATUS}=CONGESTION]?TRUNK02,${EXTEN},1)
; - next trunk if CHANUNAVAIL
exten = _X.,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?TRUNK02,${EXTEN},1)
exten = _X.,n,Hangup()
;
;
[TRUNK02]
; - same as above
exten = _X.,1,NoOp(Using Trunk 02)
exten = _X.,n,Set(GROUP()=TRUNK02CNT)
exten = _X.,n,Dial(IAX/T02/${EXTEN})
exten = _X.,n,NoOp(Use next TRUNK01is congestion or chanunavial)
exten = _X.,n,GotoIf($[${DIALSTATUS}=CONGESTION]?TRUNK01,${EXTEN},1)
exten = _X.,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?TRUNK01,${EXTEN},1)
exten = _X.,n,Hangup()
;
; - end
On 12/14/13, 4:41 PM, Muhammad Usman wrote:
Friends let me define the scenario please;
Scenario:
2 asterisk servers (A B) are connected using 05 IAX2 trunks between
them. The machine A is running asterisk Openvpn server in TUN mode
(5 instances with difference IP addresses for clients). The machine B
is running asterisk with 05 OpenVPN clients using 05 bandwidths. The
IAX trunks are established between each pair of P-2-P ip address of
machine A (The OPENVPN Server) machine B (The Openvpn client).
Requirement:
Required dial plan configuration at machine A for incoming calls from
VoIP Switch/VOS which can forward the calls to IAX2 trunks in round
robin fashion like Load Balancing. If any trunk goes down it starts
forwarding the traffic to other available trunks when it gets UP the
dialplan should perform as desired. Like L.B Fail-over scenarios.
On Fri, Dec 13, 2013 at 8:52 PM, Hans Witvliet aster...@a-domani.nl
mailto:aster...@a-domani.nl wrote:
On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote:
On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks.
I want
to load balance incoming calls over IAX2 trunks. If any trunk goes
down the calls traffic will be shared with other available trunks.
When it gets Up the script is supposed to perform as desired
i.e in
load balance mode.
Thanks in advance.
Hans said:
Perhaps it is possible to do the L.B. at the O.S. or network
level, and let
all trunks appear to asterisk to one single trunk.
Don asks:
What's the value of load balancing multiple IAX trunks between
the same
system pair? What resources are being balanced?
++
Perhaps the O.P. can explain about his intentions...
In some situations it makes sense though:
If you have to connect two servers, and use different kind of
infrastructure / multiple providers...
hw
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Regards:
(Muhammad ? )
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[asterisk-users] ARA: realtime: sip.conf: context
The answer is probably no, but I would like someone else to confirm it. When using The Asterisk RealTime Architecture, can context work in sip.conf like it does for extensions.conf ? switch = Realtime/[context]@[family][/options] More specific, is there a way to use a single table and limit the query results to a specific subset for sip.conf ? I know this can be solved using VIEW in MySQL, but I want to avoid that. WORKS: SELECT * FROM extensions.conf WHERE context = [context]; DOES NOT WORK: SELECT * from sip.conf WHERE context = [context]; Below, you can see how it works for extensions and does NOT work on the context field of the sippeers table, is there any field that can be used? Sincerely, Brian LaVallee ---=== ;# extconfig.conf ; [settings] ; sippeers = mysql,database,sippeers moresippeers = mysql,database,moresippeers extensions = mysql,database,extensions other = mysql,database,other ; ; end===--- ---=== ;# extensions.conf ; [unique85] ; switch = Realtime ; Equivalent to: SELECT * FROM database.extensions WHERE context = unique85; ; - uses current [this] context. ; switch = Realtime/limited@extensions ; Equivalent to: SELECT * FROM database.extensions WHERE context = limited; ; switch = Realtime/specific@other ; Equivalent to: SELECT * FROM database.other WHERE context = specific; ; ; end ===--- ---=== ;# sip.conf ; [general] ; switch = Realtime ; Equivalent to: SELECT * FROM database.sippeers; ; switch = Realtime/anything@moresippeers ; Equivalent to: SELECT * FROM database.moresippeers; ; NOT EQUIVALENT TO: SELECT * FROM database.moresippeers WHERE context = anything; ; ; end ===--- -- _ -- 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] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
Hi Nick, Going from DS1 to OC-n is a multi-step process. Typically requiring a hardware device to handle each MUX step. But you can find hardware that handles multiple MUX steps together. VT1.5 is just a raw OC-n channel containing a single DS1. An M13 device converts between DS3 and DS1. A DACS (DCS or DXC) provides M13 conversion, sometimes even capable of extracting the raw VT1.5 signal directly to DS1. The ILEC transport option you choose really depends on the terminating interface. Do you want to connect with a DS3 or OC-n? No matter what hardware you choose, you will need to convert to single copper pairs (DS1/T1) to connect to your Asterisk boxes. So an M13 or DCS will be necessary to reach the DS1 level. The device you choose depends on budget and growth expectations. Typically a DCS is an expensive investment, handling hundreds of DS3's. An M13 device is typically a small unit that handles one or two DS3's. The advantage comes when you add the 29th DS1. With VT1.5 it's just adding a single channel, DS3 will require another whole DS3 to get an additional DS1. Sincerely, Brian LaVallee From: Nick Khamis sym...@gmail.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 12 Jun 2013 16:19:06 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] ILEC Interconnect Hello Everyone, We are looking to interconnect with a local ILEC over an OC-n transport layer. They basically gave us two options in terms of mapping the SONET to the DS3: * VT1.5s mapping * DS1s mapping The second option is quite clear. We would MUX the connection, and plug the lines into qaud t1 cads etc... The tech mentioned that with the second option we would also need a DACS to convert back to M13 mapping. I was scared to tell him that I could not follow can someone explain that to me kindly :). I don't know much about VT1.5 mapping. Can someone kindly explain what the benefits or lack of are in choosing that option. Also what type of additional equipment we would need? In case I have overlooked something, can you gents please tell me what I will need in terms of hardware in both cases (minus routers and switches). What we are looking at is: CO | | | OC-n | v DS3 MUX | | v 21 Asterisk boxes with quad T1s Kind Regards, Nick. -- _ -- 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 -- _ -- 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] Initial REGISTER Request: Contains Credentials before 401
My SIP provider is not happy that credentials (in the Authorization header field) are provided in the initial REGISTER request. The SIP provider ONLY wants the credentials AFTER rejecting the message with a 401. I know it's dumb, because the RFC says that the the initial REGISTER message MAY include credentials. If it fails, the proper authentication method is included in the 401. I know there is nothing wrong, it is how SIP is supposed to work. However I would like to keep my SIP provider from complaining. Asterisk is NOT SUPPORTED by the SIP provider. Does anyone in the Asterisk community know how to avoid sending the credentials until AFTER receiving a 401? Any suggestions would be appreciated! Sincerely, Brian LaVallee # === # sip.conf # Asterisk 1.8.15-cert1 # --- ; [general] ; ; - trucated ; register=accountnum...@server.carrier.tld:secret:acco...@proxy.carrier.tld/ DID ; ; - end # === # SIP REGISTER Dialog # --- IP 4.4.4.4.sip 8.8.8.8.sip: UDP, length 602 REGISTER sip:server.carrier.tld SIP/2.0 Via: SIP/2.0/UDP 4.4.4.4:5060;branch=aAaAaAaAaAaAaAa Max-Forwards: 70 From: sip:accountnum...@server.carrier.tld;tag=as6c2d23d4 To: sip:accountnum...@server.carrier.tld Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4 CSeq: 190 REGISTER Authorization: Digest username=account, realm=carrier.tld, algorithm=MD5, uri=sip:sip:8.8.8.8, nonce=1368595443265327, response=0b833bff6d83337f9f88f6fb53bbcef6 Expires: 1800 Contact: sip:DID@4.4.4.4:5060 Content-Length: 0 IP 8.8.8.8.sip 4.4.4.4.sip: UDP, length 469 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 4.4.4.4:5060;branch=aAaAaAaAaAaAaAa From: sip:accountnum...@server.carrier.tld;tag=as6c2d23d4 To: sip:accountnum...@server.carrier.tld;tag=3577586129 Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4 CSeq: 190 REGISTER Content-Length: 0 Date: Wed, 15 May 2013 05:55:29 GMT WWW-Authenticate: Digest realm=carrier.tld, domain=sip:8.8.8.8, nonce=1368597329273572, opaque=, stale=TRUE, algorithm=MD5 IP 4.4.4.4.sip 8.8.8.8.sip: UDP, length 602 REGISTER sip:server.carrier.tld SIP/2.0 Via: SIP/2.0/UDP 4.4.4.4:5060;branch=BbBbBbBbBbBbBbB Max-Forwards: 70 From: sip:accountnum...@server.carrier.tld;tag=as333ffab1 To: sip:accountnum...@server.carrier.tld Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4 CSeq: 191 REGISTER Authorization: Digest username=account, realm=carrier.tld, algorithm=MD5, uri=sip:sip:8.8.8.8, nonce=1368597329273572, response=097ee5b915cd39c1407c785fb3c06caf Expires: 1800 Contact: sip:DID@4.4.4.4:5060 Content-Length: 0 IP 8.8.8.8.sip 4.4.4.4.sip: UDP, length 373 SIP/2.0 200 OK Via: SIP/2.0/UDP 4.4.4.4:5060;branch=BbBbBbBbBbBbBbB From: sip:accountnum...@server.carrier.tld;tag=as333ffab1 To: sip:accountnum...@server.carrier.tld Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4 CSeq: 191 REGISTER Contact: sip:DID@4.4.4.4:5060;q=0;expires=1901 Content-Length: 0 Date: Wed, 15 May 2013 05:55:29 GMT # === -- _ -- 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] Initial REGISTER Request: Contains Credentials before 401: KDDI Japan
Hi Matthew, Thanks for the response. From: Matthew J. Roth mr...@imminc.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 15 May 2013 12:28:11 -0500 (CDT) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Initial REGISTER Request: Contains Credentials before 401 Brian LaVallee wrote: My SIP provider is not happy that credentials (in the Authorization header field) are provided in the initial REGISTER request. The SIP provider ONLY wants the credentials AFTER rejecting the message with a 401. I know it's dumb, because the RFC says that the the initial REGISTER message MAY include credentials. If it fails, the proper authentication method is included in the 401. I know there is nothing wrong, it is how SIP is supposed to work. Who is your SIP provider? They need to be called out so that other Asterisk users can avoid them. This tendency to flip the customer/vendor relationship on its head must be discouraged. The SIP provider is KDDI Japan. However I would like to keep my SIP provider from complaining. The only thing they should complain about is if you don't pay your bill on time. Asterisk is NOT SUPPORTED by the SIP provider. The REGISTER request was successful so, at least from a practical standpoint, the provider does support Asterisk. It would be ideal if all providers officially supported Asterisk, but this is just one example of how it's not worth trying to please everyone. I know that the SIP provider is being overly diligent to prevent toll-fraud, but some of their complaints about normal SIP communications have been outrageous. I don't want to go into it, but KDDI has actually complained that the REPLY to the 401 was too fast. Does anyone in the Asterisk community know how to avoid sending the credentials until AFTER receiving a 401? Edit the source. I'm sorry to be blunt, but I really can't see the developers adding another option to sip.conf just to satisfy such a pointless request. Any suggestions would be appreciated! Ask the provider what platforms are supported. Pick one of them and use it to configure the useragent and sdpsession options in sip.conf. Or look for another provider that doesn't waste your time complaining about RFC-compliant behavior. KDDI does provide a list of supported equipment and vendors. Specific hardware or license based software products that quickly become cost prohibitive. I doubt that Asterisk will find it's way on the list any time soon. Because KDDI follows the traditional big telco method of interoperability, which normally means licensing products for use on their network. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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 -- _ -- 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] Using PHPMyAdmin to remotely access Asterisk MySQL Database
It sounds like phpMyAdmin is NOT on the same server as the Asterisk DB. You will run into a couple possible issues when allowing remote MySQL access on the Asterisk server, You will need to set the MySQL user privileges to a specific host or a wildcard (%). Most common issue is the firewall, likely iptables on the Asterisk server. Allow: tcp/3306 To test the privileges, stop iptables and try this from the local machine: mysql --host=asterisk_server --user=myuser --password=mypass If you can access the MySQL on the Asterisk server, adjust and restart iptables. Then try again with the firewall in place. To add the Asterisk server to phpMyAdmin, see the following: http://goo.gl/J1Py5 Brian From: Lobna Hegazy lobna.heg...@gmail.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 14 May 2013 22:57:34 +0200 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using PHPMyAdmin to remotely access Asterisk MySQL Database I'm successfully configured with mysql. After research, accessing Astersik MySQL database requires 2 major steps which are: (1) Allow remote access. (2) Add the new server to my local php configuration files. I'm just asking for the right steps to do these two major steps. Thank you in advance On Tue, May 14, 2013 at 10:22 PM, Gertjan Baarda gertjan.baa...@gmail.com wrote: On Tuesday, May 14, 2013, Lobna Hegazy wrote: I'm running XAMMP on Mac OS which includes Apache, MySQL, PHP, and Perl. phpMyAdmin now connected to the localhost server. What I'm willing to do is adding asterisk database server in order to access cdr database. Thank you for any help On Tue, May 14, 2013 at 10:00 PM, Gertjan Baarda gertjan.baa...@gmail.com wrote: On Tuesday, May 14, 2013, Lobna Hegazy wrote: Dear All, I'm trying to connect to Asterisk CDR database using PHPMyAdmin but unfortunately all my trials and searches failed. So I'd be more than grateful if someone helped me with right steps to do this. Kindly note that I'm working on a remore server that I can connect to as a root using ssh. Asterisk Version: 11.3.0 MySQL Version: mysql-server.x86_64 0:5.1.69-1.el6_4 Please ask me for any specifications you need, thank you in advance. -- Best Regards, Lobna Hegazy Can you be (a lot) more specific? Are you running Apache? Distro? Firewall and such. -- Sent from Gmail Mobile -- _ -- 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 -- Best Regards, Lobna Hegazy Public Relations Committee Bdaya l we love charity German University in Cairo Asterisk as such does not have a database server, it can use mysql as a backend for cdr. Is your cdr succesfully configured with mysql? Ergo: does is write the cdr entries into mysql? -- Sent from Gmail Mobile -- _ -- 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 -- Best Regards, Lobna Hegazy Public Relations Committee Bdaya l we love charity German University in Cairo -- _ -- 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 -- _ -- 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] qualify=yes: OPTIONS: How to Change?: `From: asterisk`
My Google-Fu skills have failed me, I have not been able to find a solution to the problem I am facing. asterisk + from + asterisk + options + qualify != what I am looking for -- When qualify is enabled on a trunk, the From line shows asterisk. See the SIP message below. I would like to keep qualify enabled without sending the other end any reference to asterisk. Can anyone point me to a setting that will change or remove `²asterisk²` from `FROM:` in the OPTIONS message? Thanks, Brian LaVallee -- /etc/asterisk/sip.conf (Asterisk 1.8.15-cert1) [general] ; - Truncated [TRUNK] ; - Truncated qualify=yes ; ; end -- IP 4.4.4.4.sip 3.3.3.3.sip: UDP, length 573 OPTIONS sip:server.carrier.tld SIP/2.0 Via: SIP/2.0/UDP 4.4.4.4:5060;branch=aBcDeFgHiJkLmNo;rport Max-Forwards: 70 From: asterisk sip:accountid@4.4.4.4;tag=as1832334c To: sip:server.carrier.tld Contact: sip:accountid@4.4.4.4:5060 Call-ID: f80a4ad87fee7c9fdc19b7769495fdb5@4.4.4.4:5060 CSeq: 102 OPTIONS Date: Thu, 09 May 2013 07:22:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -- _ -- 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] Thanks! qualify=yes: OPTIONS: How to Change?: `From: asterisk`
Thanks Jeremy! On 5/9/13 8:21 PM, Brian LaVallee wrote: When qualify is enabled on a trunk, the From line shows asterisk. See the SIP message below. I had the same annoyance/issue. fixed it in https://issues.asterisk.org/jira/browse/ASTERISK-17616 That's looks like the problem I was seeing. the patch was included in 1.8.9 rc1. I've been trying to stick to the current AsteriskNow as my base standard, hopefully the fix was applied to the latest version. Thanks again! Brian LaVallee -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 -- _ -- 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
