Re: [asterisk-users] WEBRTC is no longer working with Firefox after upgrade to version 37
Thank you. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Wednesday, April 08, 2015 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] WEBRTC is no longer working with Firefox after upgrade to version 37 Toufic Khreish (Gmail) wrote: Hello, Webrtc stopped after upgrading firefox from version 36 to version37. I have been running webrtc with freepbx 12 and asterisk 13.2 or 13.3 and firefox version 36 without any issues until firefox was upgraded to version 37. Unfortunately Chrome works well in one direction (from chrome to any extension) but calling from an extension to a webrtc on chrome has one way voice. Could someone try to investigate the problem of firefox version37.0.1 with webrtc ? no voice in any direction. Should we try it with a computer that has not an updated version of firefox things work normally, also if we rollback (install version 36, it works well) Someone already filed an Asterisk issue[1] and there is also a Firefox issue[2]. It's also been fixed in Firefox 38 already. [1] https://issues.asterisk.org/jira/browse/ASTERISK-24911 [2] https://bugzilla.mozilla.org/show_bug.cgi?id=1147919 -- 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 -- _ -- 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] Asterisk 1.8.28-cert5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2, 13.3.2 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolves the following security vulnerability: * AST-2015-003: TLS Certificate Common name NULL byte exploit When Asterisk registers to a SIP TLS device and and verifies the server, Asterisk will accept signed certificates that match a common name other than the one Asterisk is expecting if the signed certificate has a common name containing a null byte after the portion of the common name that Asterisk expected. This potentially allows for a man in the middle attack. For more information about the details of this vulnerability, please read security advisory AST-2015-003, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3 http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2 http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2 The security advisory is available at: Â * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf Thank you for your continued support of Asterisk! -- _ -- 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] AST-2015-003: TLS Certificate Common name NULL byte exploit
Asterisk Project Security Advisory - AST-2015-003 ProductAsterisk SummaryTLS Certificate Common name NULL byte exploit Nature of Advisory Man in the Middle Attack SusceptibilityRemote Authenticated Sessions Severity Major Exploits KnownNone Reported On 12 January, 2015 Reported By Maciej Szmigiero Posted On March 04, 2015 Last Updated OnApril 8, 2015 Advisory Contact Jonathan Rose jrose AT digium DOT com CVE Name CVE-2015-3008 Description When Asterisk registers to a SIP TLS device and and verifies the server, Asterisk will accept signed certificates that match a common name other than the one Asterisk is expecting if the signed certificate has a common name containing a null byte after the portion of the common name that Asterisk expected. For example, if Asterisk is trying to register to www.domain.com, Asterisk will accept certificates of the form www.domain.com\x00www.someotherdomain.com - for more information on this exploit, see https://fotisl.com/blog/2009/10/the-null-certificate-prefix-bug/ Resolution Asterisk has been patched to verify that the common name length of the certificate matches the common name that Asterisk actually reads. Asterisk will not accept certificates with common names that contain null bytes. Affected Versions Product Release Series Asterisk Open Source 1.8.x All versions Asterisk Open Source 11.xAll versions Asterisk Open Source 12.xAll versions Asterisk Open Source 13.xAll versions Certified Asterisk 1.8.28 All versions Certified Asterisk 11.6All versions Certified Asterisk 13.1All versions Corrected In Product Release Asterisk Open Source 1.8.32.3, 11.17.1, 12.8.2 13.3.2 Certified Asterisk 1.8.28-cert5, 11.6-cert11, 13.1-cert2 Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2015-003-1.8.28.diff Certified Asterisk 1.8.28 http://downloads.asterisk.org/pub/security/AST-2015-003-11.6.diff Certified Asterisk 11.6 http://downloads.asterisk.org/pub/security/AST-2015-003-13.1.diff Certified Asterisk 13.1 http://downloads.asterisk.org/pub/security/AST-2015-003-1.8.diffAsterisk 1.8 http://downloads.asterisk.org/pub/security/AST-2015-003-11.diff Asterisk 11 http://downloads.asterisk.org/pub/security/AST-2015-003-12.diff Asterisk 12 http://downloads.asterisk.org/pub/security/AST-2015-003-13.diff Asterisk 13 Links https://issues.asterisk.org/jira/browse/ASTERISK-24847 Asterisk Project Security Advisories are posted at http://www.asterisk.org/security
[asterisk-users] dial out with channel variable; sub-string usage
I want to do something like:
exten = _NXXXNxx,1,Dial(${BABY}/${EXTEN})
exten = _Nxx,1,Dial(${BABY}/${EXTEN})
exten = _1NXXNxx,1,Dial(${BABY}/${EXTEN})
exten = _011.,1,Dial(Dial({TOLL}/${EXTEN})
exten = _9NXXXNxx,1,Dial(${BABY}/${EXTEN})
exten = _9Nxx,1,Dial(${BABY}/${EXTEN})
exten = _91NXXNxx,1,Dial(${BABY}/${EXTEN})
exten = _9011.,1,Dial(Dial({TOLL}/${EXTEN})
(adapted from the book)
but don't know where to put those lines. I have BABY defined as channel
variable:
BABY = SIP/babytel_out
but that seems circular, somehow.
inbound calls work fine:
[inbound-calls]
exten = 16046289850,1,Dial(SIP/200)
[local_200]
exten = _9x.,1,Set(CALLERID(all)=Ali Baba 123456789)
exten = _9x.,1,Dial(SIP/${EXTEN:1}@babytel_out)
exten = 201,1,Dial(SIP/201)
[local_201]
exten = 200,1,Dial(SIP/200)
in local_200, that just seems suspect. Yes, dial out, but shouldn't it
be using BABY? I don't understand why it's using sub-string with the 1.
thanks,
Thufir
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Re: [asterisk-users] Fidelio protocol and Mitel protocol
On Tue, Apr 7, 2015 at 3:57 PM, Bryant Zimmerman brya...@zktech.com wrote: Does anyone know anything about the Fidelio and Mitel protocol for hotel / motel? Are these industry standards or proprietary formats? Are there open standards for communication with Hotel management software's that could be used in conjunction with a custom asterisk deployment? Thanks Bryant You can check what Xorcom has done with PMS's and Hospitality: http://www.xorcom.com/complete-concierge-integrated-pbx-and-pms-certified-by-micros-fidelio Xorcom is an Asterisk-based solution for PBX's and more. You can also check PBILLX: http://www.pbillx.org/pbillxnew/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
what about
exten = s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
regards
2015-04-08 5:45 GMT+00:00 Dmitriy Serov serov@gmail.com:
Hi, Andrew.
You are trying to solve two tasks: definition through what line the call
came and a beautiful display of this information.
1. definition through what line the call came. If the username and
password for inbound and outbound registration the same, then try the
following:
a) delete register lines.
b) add option callbackextension=Company1 to Company1 friend section..
And in others with their names too.
or you can change /s to /Company1 in register line.
2. beautiful display of this information
a) add option setvar=fromCompany=Company1 to Company1 friend section..
b) In dialplan add
Set(CALLERID(name)=${fromCompany} ${CALLERID(name)})
Maybe this will help?
Dmitiy.
08.04.2015 2:48, Andrew Galdes пишет:
Hi Dmitriy and others and thanks for your help so far.
The option match_auth_username=yes seems to have had no effect. From
my reading, this option will try to match the username of the incoming SIP
account to a section heading. If that is how it must work then i can see a
big problem. I'm trying to present the receptionist with a nice display of
which line the call came in on. For example, the receptionist answers calls
for 8 different companies and would like the phone to display the company
name that she should announce to the caller.
Here is a more complete output of an incoming call. I've changed the SIP
numbers to Company1', etc, to hide the numbers.
Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267)
Verbosity is at least 12
asterisk*CLI
asterisk*CLI
asterisk*CLI
== Using SIP RTP CoS mark 5
-- Executing [s@incoming:1] *Set*(*SIP/Company1-0797*,
*thedid=NodePhonesip:compa...@sip.internode.on.net
sip%3acompa...@sip.internode.on.net*) in new stack
-- Executing [s@incoming:2] *Set*(*SIP/**Company1**-0797*,
*pseudodid=NodePhonesip:** sip:Company2**@sip.internode.on.net
http://sip.internode.on.net*) in new stack
-- Executing [s@incoming:3] *Set*(*SIP/**Company1**-0797*,
*pseudodid=NodePhonesip:** sip:Company2*) in new stack
-- Executing [s@incoming:4] *Set*(*SIP/**Company1**-0797*,
*pseudodid=** sip:Company2*) in new stack
-- Executing [s@incoming:5] *GotoIf*(*SIP/**Company1**-0797*,
*0?internal,33,1:6*) in new stack
-- Goto (incoming,s,6)
-- Executing [s@incoming:6] *GotoIf*(*SIP/**Company1**-0797*,
*0?internal,88,1:7*) in new stack
-- Goto (incoming,s,7)
-- Executing [s@incoming:7] *GotoIf*(*SIP/**Company1**-0797*,
*0?internal,36,1:8*) in new stack
-- Goto (incoming,s,8)
-- Executing [s@incoming:8] *GotoIf*(*SIP/**Company1**-0797*,
*1?internal,36,1:9*) in new stack
-- Goto (internal,36,1)
-- Executing [36@internal:1] *Set*(*SIP/**Company1**-0797*,
*CALLERID(name)=SIP/**Company1**-0797*) in new stack
-- Executing [36@internal:2] *Dial*(*SIP/**Company1**-0797*,
*SIP/36,20*) in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/36
-- SIP/36-0798 is ringing
== Spawn extension (internal, 36, 2) exited non-zero on
'SIP/Company1-0797'
asterisk*CLI exit
And here is the sip.conf:
[general]
match_auth_username=yes
register=081...:...@sip.internode.on.net/s
register=082...:...@sip.internode.on.net/s
register=083...:...@sip.internode.on.net:/s
register=084...:...@sip.internode.on.net:/s
register=085...:...@sip.internode.on.net/s
register=086...:...@sip.internode.on.net/s
register=087...:...@sip.internode.on.net/s
register=088...:...@sip.internode.on.net/s
[Company1]
username=081...
fromuser=081...
secret=...
canreinvite=no
qualify=yes
context=incoming
type=friend
insecure=invite,port
fromdomain=sip.internode.on.net
host=sip.internode.on.net
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
allow=g729
bindport=5060
bindaddr=0.0.0.0
nat=yes
registertimeout=5
allowoverlap=no
srvlookup=no
ubscribecontext=from-sip
callcounter=yes
[Company2]
...
[Company3]
...
[Company4]
...
And here is some of the extensions.conf file:
[incoming]
; Get the DID number from the TO header.
exten = s,1,Set(thedid=${SIP_HEADER(TO)})
exten = s,2,Set(pseudodid=${SIP_HEADER(To)})
exten = s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten = s,4,Set(pseudodid=${CUT(pseudodid,:,2)})
; Direct the DID accordingly.
exten = s,5,GotoIf($[${pseudodid} = 081]?internal,33,1:6)
exten = s,6,GotoIf($[${pseudodid} = 082]?internal,88,1:7)
exten = s,7,GotoIf($[${pseudodid} = 083]?internal,36,1:8)
exten = s,8,GotoIf($[${pseudodid} = 084]?internal,36,1:9)
exten = s,9,GotoIf($[${pseudodid} = 085]?internal,36,1:10)
exten = s,10,GotoIf($[${pseudodid} = 086]?internal,89,1:11)
exten = s,11,GotoIf($[${pseudodid} = 087]?internal,36,1:12)
exten = s,12,GotoIf($[${pseudodid} = 088]?internal,13,1:13)
-Andrew
[asterisk-users] Asterisk is moving to Git
Hello! For quite some time now, there's been a desire in the Asterisk project to move the project source control from Subversion to Git. After a lot of work and planning, we believe we are finally able to start that process. Starting on Monday, April 13th, the Asterisk project's Subversion repository will be set to read-only. New changes will no longer be made in any of the Subversion branches. A new Git repository for the Asterisk project will be set up using Gerrit [1] as the primary repository, with mirrors conveniently located at git.asterisk.org [2]. While we've done a lot of work to plan for this migration, things can (and probably will) happen during this process. As such, there may be some hiccups during the next week or two while we iron out the finer points that such a large change will have on the project. If you are used to pulling directly from the project's Subversion repository, please be patient as we make this leap. At the same time, if you do happen to encounter any issues, please don't hesitate to send an e-mail to the asterisk-dev mailing list [3] or talk with the developers in the #asterisk-dev IRC channel. As always, thanks for supporting the Asterisk project! Matt [1] https://gerrit.asterisk.org [2] https://git.asterisk.org [3] http://lists.digium.com/mailman/listinfo/asterisk-dev -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition
I have seen a similar problem occasionally. We will be doing maintenance on a customer's server and they will have one or two ghost channels on their machine hundreds of hours old but with no call associated with them. So far we have just been rebooting their server or issuing a hangup command to the channels. From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Alex Villacís Lasso a_villa...@palosanto.com Sent: 08 April 2015 00:33 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition El 07/04/15 a las 17:38, Alex Villacís Lasso escribió: I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses asterisk with FreePBX 2.8. In order to generate the calls, I wrote a program that connects to Asterisk using the AMI protocol. This program expects the SIP agent extensions to be assigned as members of queues, of which there are about 20, as shown below: 9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s talktime), W:0, C:581, A:260, SL:82.6% within 60s Members: SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls (last was 800 secs ago) SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls (last was 708 secs ago) SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls (last was 656 secs ago) SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls (last was 789 secs ago) No Callers The program runs queue show through AMI every few seconds. For each queue to be used in telemarketing, the program counts the number of members that are Not In Use. If at least one is found, it reads that many phone numbers from the database and uses the AMI Originate command on each one, as follows: Action: Originate Channel: Local/NN@from-internal Exten: Context: from-internal Priority: 1 Async: true ActionID: xxx Here, NN is the number read from the database and is the queue extension in the FreePBX-created context that eventually runs the Queue() dialplan application for the corresponding queue. This causes the call to be connected between the outgoing number and the queue, and is then assigned to a queue member by Asterisk. The dialplan is configured to route NN through one of a series of SIP trunks using the outbound routes as configured by FreePBX. The issue is that although this strategy works correctly on the user's machine for a few days, we have been observing that eventually the application stops placing calls. The agents are all idle (all 90 to 100 of them), but the queue show command shows them to be In Use on all queues. Furthermore, in normal operation, the core show channels command shows at most one channel for each configured SIP client in the Up state, but when calls stop being placed, the same command reports multiple channels in the Up state, as follows (after sorting): Local/9757007441@from-internal-a447;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,!47740412!!!3!572!(None)!1428426012.169192 Local/9759315789@from-internal-a456;1ZOMBIE!from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing Line)!9759315789!!!3!500!(None)!1428426084.169326 Local/9759315789@from-internal-a456;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,!47740412!!!3!500!(None)!1428426084.169323 SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/5547740413-00014c62!1428426022.169224 SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing Line)!110!!!3!590!(None)!1428425994.169124 SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760 SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!1428426052.169273 SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384 SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!1428426574.169850 SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!1428425991.169113 SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436
Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition
Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villacís Lasso a_villa...@palosanto.com: El 07/04/15 a las 17:38, Alex Villacís Lasso escribió: I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses asterisk with FreePBX 2.8. In order to generate the calls, I wrote a program that connects to Asterisk using the AMI protocol. This program expects the SIP agent extensions to be assigned as members of queues, of which there are about 20, as shown below: 9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s talktime), W:0, C:581, A:260, SL:82.6% within 60s Members: SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls (last was 800 secs ago) SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls (last was 708 secs ago) SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls (last was 656 secs ago) SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls (last was 789 secs ago) No Callers The program runs queue show through AMI every few seconds. For each queue to be used in telemarketing, the program counts the number of members that are Not In Use. If at least one is found, it reads that many phone numbers from the database and uses the AMI Originate command on each one, as follows: Action: Originate Channel: Local/NN@from-internal Exten: Context: from-internal Priority: 1 Async: true ActionID: xxx Here, NN is the number read from the database and is the queue extension in the FreePBX-created context that eventually runs the Queue() dialplan application for the corresponding queue. This causes the call to be connected between the outgoing number and the queue, and is then assigned to a queue member by Asterisk. The dialplan is configured to route NN through one of a series of SIP trunks using the outbound routes as configured by FreePBX. The issue is that although this strategy works correctly on the user's machine for a few days, we have been observing that eventually the application stops placing calls. The agents are all idle (all 90 to 100 of them), but the queue show command shows them to be In Use on all queues. Furthermore, in normal operation, the core show channels command shows at most one channel for each configured SIP client in the Up state, but when calls stop being placed, the same command reports multiple channels in the Up state, as follows (after sorting): Local/9757007441@from-internal-a447;2!macro- dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300, !47740412!!!3!572!(None)!1428426012.169192 Local/9759315789@from-internal-a456;1ZOMBIE! from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing Line)!9759315789!!!3!500!(None)!1428426084.169326 Local/9759315789@from-internal-a456;2!macro- dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300, !47740412!!!3!500!(None)!1428426084.169323 SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial! SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/ 5547740413-00014c62!1428426022.169224 SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing Line)!110!!!3!590!(None)!1428425994.169124 SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial! SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760 SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial! SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)! 1428426052.169273 SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial! SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384 SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial! SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)! 1428426574.169850 SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial! SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)! 1428425991.169113 SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial! SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436 SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial! SIP/5547740413/5556802622,300,!47740413!!!3!327!(None)!1428426257.169493 SIP/120-00014d5e!macro-dial-one!s!37!Up!Dial!SIP/230,, trT!120!!!3!314!SIP/230-00014d5f!1428426270.169510 SIP/121-00014c24!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing Line)!121!!!3!596!(None)!1428425988.169111 SIP/122-00014b56!EjecutivoQUADS!93000!1!Up!AppQueue!(Outgoing Line)!122!!!3!677!(None)!1428425906.168693 SIP/123-00014d53!macro-dialout-trunk!s!19!Up!Dial!
[asterisk-users] use of EC2
Curious if anyone has any stats on max concurrent calls on different EC2 instance sizes. A client has a proof of concept running on a medium compute instance now, and we are curious at what point we might experience issues. All calls are SIP, no transcoding, using SPEEX. I'd love to hear if anyone has a small or medium compute instance doing 100 simultaneous calls. Cheers, j -- _ -- 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] Asterisk Inbound calls, multiple SIP accounts, calledID
Andrew,
Instead of your SET and GOTO blocks I'd recommend using the Asterisk DB to
make things easier to maintain.
You could make two database entries for each of your DID's
database put 4259981810 name JohnPersonal
database put 4259981810 target kiniston-extern,john-personal,1
Then you could do a single block that would do the lookup and call routing:
Set(DESTINATION=${CUT(PASSTHRU(${SIP_HEADER(TO):5}),@,1)})
Set(CALLERID(name)=${DB(${DESTINATION}/name)})
Goto(${DB(${DESTINATION}/target)})
On Tue, Apr 7, 2015 at 6:06 PM, Andrew Galdes andrew.gal...@agix.com.au
wrote:
Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but
it does work. For prosperity, the SIP service is through Internode.
Here is my extensions.conf file:
exten = s,5,Set(callersname=${IF($[ ${pseudodid} =
081...]?Company1:${callersname})})
exten = s,6,Set(callersname=${IF($[ ${pseudodid}
= 082...]?Company2:${callersname})})
exten = s,13,GotoIf($[${callersname} = Company1]?internal,36,1:14);
to reception
exten = s,14,GotoIf($[${callersname} = Company2]?internal,88,1:15);
to department1
And later in same file:
; Phone 36 reception
*exten = 36,1,Set(CALLERID(name)=${callersname})*
exten = 36,n,Dial(SIP/36,20)
exten = 36,n,VoiceMail(36,u)
exten = 36,n,Hangup
--
_
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[asterisk-users] WEBRTC is no longer working with Firefox after upgrade to version 37
Hello, Webrtc stopped after upgrading firefox from version 36 to version37. I have been running webrtc with freepbx 12 and asterisk 13.2 or 13.3 and firefox version 36 without any issues until firefox was upgraded to version 37. Unfortunately Chrome works well in one direction (from chrome to any extension) but calling from an extension to a webrtc on chrome has one way voice. Could someone try to investigate the problem of firefox version37.0.1 with webrtc ? no voice in any direction. Should we try it with a computer that has not an updated version of firefox things work normally, also if we rollback (install version 36, it works well) Thank you and best regards Toufic KHREISH -- _ -- 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] WEBRTC is no longer working with Firefox after upgrade to version 37
Toufic Khreish (Gmail) wrote: Hello, Webrtc stopped after upgrading firefox from version 36 to version37. I have been running webrtc with freepbx 12 and asterisk 13.2 or 13.3 and firefox version 36 without any issues until firefox was upgraded to version 37. Unfortunately Chrome works well in one direction (from chrome to any extension) but calling from an extension to a webrtc on chrome has one way voice. Could someone try to investigate the problem of firefox version37.0.1 with webrtc ? no voice in any direction. Should we try it with a computer that has not an updated version of firefox things work normally, also if we rollback (install version 36, it works well) Someone already filed an Asterisk issue[1] and there is also a Firefox issue[2]. It's also been fixed in Firefox 38 already. [1] https://issues.asterisk.org/jira/browse/ASTERISK-24911 [2] https://bugzilla.mozilla.org/show_bug.cgi?id=1147919 -- 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
