Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?
On Fri, June 16, 2017 12:28, Tim S wrote: Whether it is intentional or not these messages railing against the list operators has a decided tone of condescension which is not warranted. The fact of the matter is that DMARC is broken by design and the unpleasant effects that adoption of it has on mailing-list traffic were well hashed out on the ITEF mailing lists before it was adopted anyway. What was predicted there has come to pass. DMARC conflicts with the existing SMTP RFCs in several ways, none of which I will elaborate here but all of which may be discovered by perusing the relevant threads on the ITEF mailing lists. Some mailing list management software, notably Mailman, since has been modified to 'work around' the problems with DMARC if so configured by the list owners. But only at the cost of violating the SMTP RFCs themselves. Do not take my word for it. Raise these issues on the Postfix mailing list and discover what response you get from Viktor and Wietse. The driving force behind DMARC was YAHOO's shoddy security of their own users' accounts. With Hotmail and similar ilk close behind. It is a completely inappropriate, and in my opinion ill-thought-out, technical solution to what is essentially an internal security problem at some email providers, albeit very large ones. In general it is an example of what is called 'externalising your costs'. The appropriate answer has been provided: lose the gmail/hotmail/yahoo/freemail account and administer your own domain for personal email. Configure the spf and dkim settings on your own domain as required to suit your needs and not those of someone else. -- *** e-Mail is NOT a SECURE channel *** Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom870 FW:8.7.5.35
Does anyone on this list know how to make the Snom870 with FW:8.7.5.35 display the Caller ID in the display field while the ringing either together with, or instead of, the topmost virtual key in the info column? I realise that the purpose of having the virtual key display the caller ID so as to allow selection of which incoming call to take. But the resulting display size is so small as to make that information unusable. -- *** e-Mail is NOT a SECURE channel *** Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: DMARC enabled domains on this list
On Mon, June 5, 2017 15:30, Daniel Tryba wrote: > > The reports are there to tell you something isn't right (like on this > mailing list). Disabling them is only hiding the problem, people might > be replying with the correct answer to a problem, but the OP might > never gets that message. > What DMARC reports is that somebody other than yourself is sending email claiming to be you. And there is absolutely nothing that you can do about it. So the question arises: What is the value in these reports? -- *** e-Mail is NOT a SECURE channel *** Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: DMARC enabled domains on this list
On Fri, June 2, 2017 16:30, Doug Lytle wrote: This is likely the issue surrounding mailing lists rewriting headers and/or modifying messages bodies or simply re-transmitting messages as the original sender from an unapproved domain. This was discussed at length on the ITEF mailing list. Without seeing your headers and those of a recipient it is impossible to be sure but my spidy sense tells me this is so. You can manage this in your DNS forward zone by turning off the DMARC reporting request. No, I no longer recall the details. Or you can simply direct the incoming reports to /dev/null. As I get the digest version of the list the message sender and domain match DMARC provisions, if any are set for digium.com. HTH. -- *** e-Mail is NOT a SECURE channel *** Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM870 provisioning BLF settings
We use Snom870s together with Asterisk 13.14.0 and FreePBX 13.0.191.11. I am having an ongoing problem with setting the BLF values on these phones from the configuration file generated from FreePBX. In FreePBX we employ the Commercial Endpoint Manager (CEM) to configure these phones. The resulting configuration file contains lines like these: blf sip:10@; blf sip:11@; blf sip:12@; However, on the test phone I am using (FW 8.7.5-35) I see this as a result: Context TypeNumber Short Text Active LineP1 Active LineP2 . . . Active LineP15 When instead I have expect to see: Context TypeNumber Short Text Active BLF <sip:10@addr;user=phone>|** Recep P1 Active BLF <sip:11@addr;user=phone>|** AKL P2 . . . Active BLF <sip:24@addr;user=phone>|** JillP15 I have rebooted this phone several times and the provisioning web server records the transfer of the settings file with a 200 result. "GET /snom870.htm HTTP/1.1" 200 224 "-" "Mozilla/4.0 (compatible; snom870-SIP 8.7.5.35 SPEAr300 SNOM 1.4 000413419A8A)" But the settings file appears to have no effect on the BLF configuration. I am at a loss at this point. Any suggestions? -- *** e-Mail is NOT a SECURE channel *** Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxModem pickup problem
On Thu, May 4, 2017 13:19, Telium Technical Support wrote: . . . > This design (FreePBX) makes Asterisk much more fragile than it has to > be. > It's a good idea to keep a backup astdb on the PBX in case of > corruption. > I have added a cron job to make a copy that file every day at midnight with a date timestamp in the file name. I also have daily scheduled backups of the entire FreePBX installation and databases through FreePBX itself but this approach seems a little more convenient. We have had similar incidents in the past which we could never determine the cause of before it was somehow rectified. I infer that on each occasion something we tried simply caused the astdb file to be rebuilt and thereby corrected the issue without us ever being aware that is what happened. -- *** e-Mail is NOT a SECURE channel *** Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxModem pickup problem
On Thu, May 4, 2017 11:38, Telium Technical Support wrote: > It depends a bit on your version of FreePBX, but here's a link to show > you how: > > http://telium.ca/pages/forums/viewtopic.php?f=7=19 > > Hopefully option 1 works for you (quick and easy). If not, you'll > have to try option 2. Ignore option 3 since that's only for users > of High Availability for Asterisk (HAAst). > > (I assume that if you had a full backup you would have already tried > to restore it) > No, I did not try to restore from backups; and yes I have daily backups to recover from if that is necessary. However, I have since corrected the damaged rows in astdb.sqlite and the fax service is now working again. If someone could explain what likely happens to damage astdb.sqlite when the system is suddenly powered off I would appreciate it. -- *** e-Mail is NOT a SECURE channel *** Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxModem pickup problem
On Thu, May 4, 2017 10:22, James B. Byrne wrote: I am advised that it may be possible thast the astdb.sqlite3 database may be corrupted. Are there procedures to rebuild or repair this? Where are they documented? If not then how does one repair such? -- *** e-Mail is NOT a SECURE channel *** Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxModem pickup problem
We run Asterisk 13 using the FreePBX 13.0.190.19 distro based on CentOS-6.4. We also run HylFAX+ 5.5.3 with iaxModem 1.2.0 on the same system with AdvantFAX as the web front-end. Our two fax lines are configured as iax2 DEVICES. These components have been working together through various versions since 2013. On Tuesday last our site was subjected to a prolonged power outage that drained our twin UPS set up flat resulting in a power down state for the Asterisk host. Upon power recovery the asterisk host system came up, the phone system works, but we are now unable to receive faxes through Asterisk. We can send faxes but not receive them. The fax line never picks up and a redirect to a voice recording informing voice callers of their mistake is triggered instead. I have a trace of the asterisk 'full' log that captures one of these failed calls and I would like some help in determining if a clue to what has happened is contained therein. I do not wish to simply post it to the list with getting permission since it is quiet long. But I do need to get this resolved and I cannot fathom why the lines are not picking up incoming faxes. Any help would be gratefully accepted. -- *** e-Mail is NOT a SECURE channel *** Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Script to Program Snom Phones
On Thu, April 9, 2015 12:37, Tafadzwa Nyabasa wrote: Hi There, Does anyone know how to program Snom phones using a Mac addresses like what is done with the Ciscos. I have about 50 extensions to be programmed and I am hoping there is a way on Asterisk to program extensions on the snom phones. Please assist. Regards I do not think that this is specifically an Asterisk problem. The SNOM phones that we use (870s and 76s) have FW 8.7.3.25.5. On the Update tab of the Advanced setting page there are set the update policy and URI. In our case the settings are 'Never update, load settings only', from URL http://192.168.6.9:83, with a refresh interval of 600840. The phone will look at http://192.168.6.9:83 for a file called snom870-.htm where is the phone's MAC number. If that fails then it will look for snom870.htm instead. These files should contain the xml dialect for the SNOM phone configuration directives: ?xml version=1.0 encoding=utf-8? settings phone-settings language perm=RWEnglish/language dnd_on_code perm=*78/dnd_on_code . . . /phone-settings /settings You need to provide a service that will provide the file via URI. You must put files therein with names following the specific nomenclature employed buy the phones themselves. Finally you must also set the phones to read from that location and to apply the configurations retrieved therefrom. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Anonymous SIP calls
On Thu, March 26, 2015 22:29, Michelle Dupuis wrote: You have to consider whether you really want anonymous calls, or you just want to enable SIP calls from trusted companies/partners. The latter means setting up routes to these companies and (ideally) registration between peers. This is what I am trying to get a handle on. It seemed to me that the promise of VOIP was essentially that one could use the Internet as a replacement for the PSTN directly, providing that ones callers/callees were also directly connected via VOIP. SIP providers I had considered a necessary transition to act as gateways between PSTN dialing and VOIP until VOIP replaced PSTN virtually entirely if not completely. That is why we are on Asterisk. We had to replace our old keyed system and the thought was that we might as well get ready for VOIP even if we planned to stay on PSTN for the foreseeable future. However, the overwhelming evidence I find is that one simply does not employ VOIP in the same way that PSTN works. Actually, I have put that backwards. What I have discovered is that the most commonly recommended method is to switch from a Telco to A SIP provider and continue in a manner similar to the former set-up. External calls all have to travel through a third party provider. One does not accept incoming VOIP calls from just everyone, apparently. One only accepts VOIP calls from known correspondents. I am not clear why this is so other than vague warnings respecting (admittedly real and serious) security issues. Even limiting VOIP to known correspondents one is ultimately trusting that they themselves are secured sufficiently to prevent unauthorised access to your systems through theirs. And that seems a bit of a stretch by way of rationalisation to me. Also I do not understand is why the same issues do not exist from incoming calls via PSTN. I somewhat understand the process of getting devices to register and authenticate to obtain access to our outgoing routes. What is it about incoming SIP calls destined to our internal users that make those calls so dangerous? Why cannot incoming anonymous SIP calls not be treated exactly as incoming PSTN calls (other than PSTN have to go though DAHDI to turn them into digital VOIP calls). What is it that prevents them from being blocked from gatewaying through to our PSTN lines? Please forgive my abysmal ignorance on this matter. Perhaps I have been down in the weeds too long getting our internal FreePBX system working to see what is obvious to others. I have been going theough the Asticon Videos on security and have or already had implemented most of the suggestions: Outbound LD secured by pins and allowed only during work hours; IPTABLES rules and fail2ban checks; Separation of voice and data network segments and addresses; Private IP for VOIP desk-sets and internal provisioning; and so forth. However, I still have the sense that I am just not getting it. What am I missing? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Anonymous SIP calls
We have a FreePBX-12 / Asterisk-12 setup that supports about 24 extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence. The server host is a dedicated atom(tm) box using the FreePBX distro (CentOS-6.x) and is up-to-date. Registrations require very long random passwords and registrable devices are further restricted by netblock filters. We have the usual firewall and fail2ban intrusion prevention and detection set-ups in place. Our connection to the rest of the world is via PSTN. We do our own DNS, both forward and reverse. We have NAPTR and SRV RRs for SIP and SIPS. That is the environment. Now for the questions. Can I safely configure FreePBX/Asterisk to allow people to call us directly via SIP? In other words, sip://someth...@harte-lyne.ca would reach us and ring internally as if someone had called our main office number via PSTN. Does it make sense to do so? I am not talking about routing our main number through a SIP trunk provider. We will remain on PSTN for the foreseeable future. But I am curious as to whether or not it it worthwhile to allow others who have the capability to simply call us via SIP rather than over PSTN. And if we do allow it what are the caveats and how does one actually configure Asterisk to do it? I have read a number of blogs, sections of the Definitive Asterisk book and mailing list archived posts respecting anonymous SIP calls. But I have to say these leave me rather more confused than informed. Virtually all sources advise against accepting any anonymous incoming SIP calls whatsoever. The few that do not absolutely advise against do not give much guidance in how to handle incoming calls. And frankly, I have only a dim idea how an incoming SIP call should be handled from a theoretical point of view. Any guidance would be welcome. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Jitsi, SRTP and Asterisk 11
Does anyone here have a Jitsi softphone set up with Asterisk such that SRTP is enabled, TLS is used to pass the SRTP key, and it works? Anyone? If so then what are the settings required for Asterisk? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Guidence in DialPlan programming.
I am dealing with a FreePBX generated dialplan. I have been following
the processing traces attempting to make use of the advice I received
here respecting setting a custom ring tone. I have discovered that
the context I am using for incoming calls is not used at all during a
blind transfer. Thus setting a third ring tone for that situation
inside that context is an impossibility.
I know now what I need to do and possibly where I need put it. What I
wish is some guidance on how to properly return from my custom code
without damaging the dialplan elsewhere.
Here is the situation:
In extensions.conf I see this:
;-
;-
; Internal dialplan that most internal phones have access to
;
[from-internal]
include = from-internal-noxfer
include = from-internal-xfer
include = bad-number ; auto-generated
;-
;-
; from-internal-noxfer:
;
; Place to put internal dialplan that should not be accessible
; during a blind transfer, this context will not be visible
; during such.
;
[from-internal-noxfer]
include = from-internal-noxfer-custom
include = from-internal-noxfer-additional ; auto-generated
;-
;-
; from-internal-xfer:
;
; Place to put most internal dialplan, will be visible during
; normal calls and blind transfers.
;
[from-internal-xfer]
include = from-internal-custom
include = from-internal-additional ; auto-generated
exten = s,1,Macro(hangupcall)
exten = h,1,Macro(hangupcall)
;-
;-
If a call is placed by a local extension then context
[from-internal-noxfer] is used. If a blind transfer is performed the
context is [from-internal-xfer]. What I am considering is placing the
following code in extensions-custom.conf:
[from-internal-custom].
exten = _X,1,Noop()
exten = _X,n,Set(AlertSnom=http://www.notused.com\;info=)
exten = _X,n,Set(AlertInternalTransfer=alert_internal_transfer)
exten = _X,n,Set(__ALERT_INFO=${AlertSnom}${AlertInternalTransfer})
exten = s,1,Macro(hangupcall)
exten = h,1,Macro(hangupcall)
My question is: are the last two lines the correct method of returning
from this back to extensions.conf? Is there something else I should
use? At them moment I just want to know how to properly and safely
return to the original referring context ([from-internal-xfer]).
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?
On Thu, March 5, 2015 09:56, Ruben Rögels wrote:
Hi again,
I'm glad to hear that I provided a somehow useful answer.
Unfortunatelly, I don't know these details.
If you wasn't lucky consulting the snom docs, maybe the snom support
can be helpful with information about the exact implementation
details.
You also could use sip debug on asterisk to check what's going on
when pressing the transfer button vs. what's happening when using
## via DTMF.
Are you forced to get the transfer information from the SIP
signalling, or can you use AMI events for example? I think
this would be possible if asterisk is configured to stay in
the media path, so re-inviting is handled over asterisk itself
and therefore detectable with AMI events.
I am working with a FreePBX12/Asterisk11 setup. Asterisk stays on the
path (B2B) and there are no peer-to-peer re-invites.
What I am trying to do is to get our Snom870s to use a distinctive
ring tone when external calls are transferred internally. I have an
extension context override that detects the origin of calls and
assigns a distinctive ring to each based on ${CallerIDNum}.
But when a call is transferred then the tone does not change since the
CallerIDNum does not. An external original call always rings as if it
were coming from the outside (which it is but transferred calls have a
different handling procedure than unanswered calls). I need some way
to distinguish when the call has already been answered at least once
without changing the CallerID.
I am not worried about attended transfers since then the internal ring
tone is what should be used and that is what happens now. I just need
to deal with blind transfers.
What I have now is:
1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring1 || ring2 (depending on 1 or 2)
What I want is:
1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring3 (regardless of 1 or 2)
If everything went though ## then that would be simple enough. The
trick is that most (all) users employ the transfer button and the
touch screen to forward calls using blind transfer. But whatever
method they use to transfer I want the transfer ring tone to be the
same, albeit different from the one used for a new incoming call.
If the transfer is done using a sip message then that should be doable
as well. I just have to discover what the message is. If someone
already knows and would care to share the information then that would
be helpful. Otherwise wireshark and debug will eventually reveal it.
I may not know what I am doing. But, at least I know that I do not
know what I am doing.
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
--
_
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Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?
On Thu, March 5, 2015 05:30, Ruben Rögels wrote: Am 05.03.2015 um 01:09 schrieb James B. Byrne: I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? Hi, I hope I understood your question correctly. AFAIK, the transfer button sends a SIP message. Entering ## on the handset is recognized via DTMF by asterisk. I hope that I understood what I was asking for. Sometimes I do not. Yes, that is what I wanted to know. Does the implementation of the transfer button feature on the Snomp-870 use exactly the same technique as the ## feature code entered through the dial pad and produce exactly the same SIP message that Asterisk produces when it gets the ## DTMF? The reason is that I wish to be able to detect a call transfer performed via either method (## or Transfer-Button) and process the result of both in the same fashion. If the button and DTMF transfers are not performed using the same switching techniques in Asterisk then I need to discover what those differences are. If both are totally equivalent from a SIP message signalling point of view then the issue is far easier to handle. I searched, in vain, in the Snom-870 docs for specifics on this and either could not find or did not recognize anything that applied. Do you know where I can locate these sorts of details. My knowledge of SIP/RTP/VOIP etc. is cursory but, given an adequate reference, I can usually sort things out. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] OT - How does the blind transfer function work on Snom-870?
I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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, SRTP, Asterisk11 and Snom870s
This seems to me to be getting down to some sort of problem with configuring the Snom-870. when I register the device 41712 (set up for transport=tls only) then I see this in the SIP trace: Sent to udp:192.168.6.9:5060 at 4/3/2015 09:07:36:813 (836 bytes): REGISTER sip:voinet09.internal.hamilton.harte-lyne.ca:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.112:5060;branch=z9hG4bK-udx92poqese6;rport From: James B Byrne sip:41...@voinet09.internal.hamilton.harte-lyne.ca:5061;tag=frgaimnglp To: James B Byrne sip:41...@voinet09.internal.hamilton.harte-lyne.ca:5061 Call-ID: 71004941-gk6y4evf6dci CSeq: 482 REGISTER Max-Forwards: 70 Contact: sip:41712@192.168.6.112:5060;line=0p8zx4sh;reg-id=1;q=1.0;+sip.instance=urn:uuid:ad1349a7-e08d-411b-83b0-000413281B56;audio;mobility=fixed;duplex=full;description=snom870;actor=principal;events=dialog;methods=INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO User-Agent: snom870/8.7.3.25.5 Allow-Events: dialog X-Real-IP: 192.168.6.112 Supported: path, gruu Expires: 3600 Content-Length: 0 The SNOM-870 is sending registration via UDP and not by TLS. Is that how things are supposed to work? If only TLS is enabled in Asterisk for that peer then evidently the peer cannot register. But is registration supposed to be done via TLS? If so then how does one configure the Snom to do so? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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, SRTP, Asterisk11 and Snom870s
CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten this set-up (Asterisk11 with Snom870s using TLS) to work and if so could you provide the details? I have this in Asterisk sip.conf (loaded through FreePBXs sip_general_additional.conf). tcpenable=yes tlsenable=yes tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt tlscafile=/etc/pki/tls/certs/ca-bundle.crt tlsdontverifyserver=yes tlscipher=ALL tlsclientmethod=tlsv1 And I have this for the test device context: [41712] deny=0.0.0.0/0.0.0.0 secret=NearlyANastyThat dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=no port=5060 qualify=yes qualifyfreq=60 transport=tls,udp,tcp avpf=no force_avp=no icesupport=no encryption=yes callgroup= pickupgroup= dial=SIP/41712 mailbox=41712@device permit=192.168.6.0/255.255.255.0 callerid=James B Byrne 41712 callcounter=yes faxdetect=no cc_monitor_policy=generic If I change the transport setting to TLS then I get this reported: [2015-03-03 11:10:08] ERROR[22244]: tcptls.c:875 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.6.112:5060: Connection refused I cannot seem to configure the Snom870 to listen for TCP on 5060. There is a setting for that on the phone but it seems to have no effect (it always returns to NO following a reboot). The Snom website says that the option is not available in FW8.5 and later. It does not inform one of whether that the phone listens by default or not on FW8.5+, only that the option has no effect. It also does not say, as far as I can find, whether Snom870s listen for TCP at all or on what port. One may infer that since these devices purport to support TLS that the answer is yes and that TCP5061 is a likely candidate. But they do not seem to come right out and say so anywhere. In a section devoted to the Snom370, which is a model that we do not employ, there is reference to DNS SRV RRs. The inference drawn from the examples given is that these will control what ports the Snom will listen on for which services. We have such records in our DNS zone. They look like this: ;# Configure sip/sips service records (VOIP) ;HOST TTL CLASS TYPEORDER PREF FLAGS SERVICE REGEXP REPLACEMENT 300 IN NAPTR 50 50 s SIPS+D2T_sips._tcp.harte-lyne.ca. 300 IN NAPTR 90 50 s SIP+D2T _sip._tcp.harte-lyne.ca. 300 IN NAPTR 100 50 s SIP+D2U _sip._udp.harte-lyne.ca. ;HOST TTL CLASS TYPEORDER PREF PORTTARGET _sips._tcp.harte-lyne.ca. 300 IN SRV 10 10 5061voinet09.hamilton.harte-lyne.ca. _sip._tcp.harte-lyne.ca.300 IN SRV 10 10 5060voinet09.hamilton.harte-lyne.ca. _sip._udp.harte-lyne.ca.300 IN SRV 10 10 5060voinet09.hamilton.harte-lyne.ca. However, our phones are configured to use SIP accounts having the form account@ipv4-addr. I doubt greatly that the Snom870s will perform a reverse DNS lookup on the provider's IPv4 to discover the forward zone domain and thus I do not believe that SRV RRs can help us in this instance. They certainly do not seem to have any effect. Asterisk seems not to distinguish between 5060 and 5061 regarless of protocol. I am not sure then how to proceed. Is there a way to force Asterisk to talk to port TCP5061 on a specific device? Is this an exclusive setting? This long background is by way of asking for help. If I have not provided specific information that is significant to this problem then I will do so if asked. What I am attempting has to be possible. Somehow. And somebody must have already accomplished this. Somewhere. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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
Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s
These are the sip settings on our installion. Global Settings: UDP Bindaddress:0.0.0.0:5060 TCP SIP Bindaddress:0.0.0.0:5060 TLS SIP Bindaddress:(null) Videosupport: No Textsupport:No Ignore SDP sess. ver.: No AutoCreate Peer:Off Match Auth Username:No Allow unknown access: Yes Allow subscriptions:Yes Allow overlap dialing: Yes Allow promisc. redir: No Enable call counters: No SIP domain support: No Realm. auth:No Our auth realm asterisk Use domains as realms: No Call to non-local dom.: Yes URI user is phone no: No Always auth rejects:Yes Direct RTP setup: No User Agent: FPBX-12.0.40(11.14.2) SDP Session Name: Asterisk PBX 11.14.2 SDP Owner Name: root Reg. context: (not set) Regexten on Qualify:No Trust RPID: No Send RPID: No Legacy userfield parse: No Send Diversion: Yes Caller ID: Unknown From: Domain: Record SIP history: Off Call Events:On Auth. Failure Events: Off T.38 support: No T.38 EC mode: Unknown T.38 MaxDtgrm: 4294967295 SIP realtime: Disabled Qualify Freq : 6 ms Q.850 Reason header:No Store SIP_CAUSE:No Network QoS Settings: --- IP ToS SIP: CS3 IP ToS RTP audio: EF IP ToS RTP video: AF41 IP ToS RTP text:CS0 802.1p CoS SIP: 4 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 6 802.1p CoS RTP text:5 Jitterbuffer enabled: No Network Settings: --- SIP address remapping: Enabled using externaddr Externhost: none Externaddr: 216.185.71.9:0 Externrefresh: 10 Localnet: 216.185.71.0/255.255.255.0 192.168.6.0/255.255.255.0 192.168.209.0/255.255.255.0 192.168.216.0/255.255.255.0 192.168.71.0/255.255.255.0 Global Signalling Settings: --- Codecs: (gsm|ulaw|alaw) Codec Order:ulaw:20,alaw:20,gsm:20 Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: Yes Compact SIP headers:No RTP Keepalive: 0 (Disabled) RTP Timeout:30 RTP Hold Timeout: 300 MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: No Pedantic SIP support: Yes Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Sub. min duration 60 secs Sub. max duration: 3600 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Outbound reg. retry 403:0 Notify ringing state: Yes Include CID: No Notify hold state: Yes SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Outb. proxy:not set Session Timers: Accept Session Refresher: uas Session Expires:1800 secs Session Min-SE: 90 secs Timer T1: 500 Timer T1 minimum: 100 Timer B:32000 No premature media: Yes Max forwards: 70 Default Settings: - Allowed transports: UDP Outbound transport: UDP Context:from-sip-external Record on feature: automon Record off feature: automon Force rport:Yes DTMF: rfc2833 Qualify:0 Keepalive: 0 Use ClientCode: No Progress inband:Never Language: Tone zone: Not set MOH Interpret: default MOH Suggest: Voice Mail Extension: *97 -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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, SRTP, Asterisk11 and Snom870s
On Tue, March 3, 2015 13:19, jg wrote: Forget about the reverse DNS stuff for the moment. Do simple SIP accounts (without SRTP/SRTP and deny/permit stuff) work? Enable SRTP, but you likely need the AES-80 fro SRTP Auth-tag. Then try the rest. jg The Snom870s and our Asterisk FreePBX are communicating with each other and have been for the past two years. The Snoms are configured for AES-80 and SRTP is enabled on the FreePBX device entry. We have a working PBX system. I am trying to secure it. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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, SRTP, Asterisk11 and Snom870s
I reconfigured sip.conf to have these settings: tcpenable=yes tlsenable=yes tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.pem tlscafile=/etc/pki/tls/certs/ca-bundle.crt tlsdontverifyserver=yes tlscipher=ALL tlsclientmethod=tlsv1 tlsprivatekey=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.key tcpbindaddr=0.0.0.0/0.0.0.0:5061 tlsbindaddr=0.0.0.0/0.0.0.0:5061 Following amportal a r I see this: [2015-03-03 16:26:48] ERROR[17130]: tcptls.c:875 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.6.112:5060: Connection refused This is what sip show settings reveals: Global Settings: UDP Bindaddress:0.0.0.0:5060 TCP SIP Bindaddress:0.0.0.0:5060 TLS SIP Bindaddress:0.0.0.0:5061 Is it just me or is there something odd about specifying a TCP port and then having it ignored? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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, SRTP, Asterisk11 and Snom870s
On Tue, March 3, 2015 16:34, James Cloos wrote: Other things to consider: The transport config, which can be in [general] or in a peer's [] block. if you want tls-only, use transport=tls it also accepts tcp, udp or a comma-separated list. if given a list, it tries them in order The specific device I am using to test this with has only transport=tls set. Which is why it cannot register because the default fall-back to udp is not permitted. If you need ast to register over tls, use something like this: register = tls://username:xxx...@sip-tls-proxy.example.org Does this go in the device context? In other words is it placed in the same context that the device's transport value is set? Would the following be valid? [device] register = tls://user:extension@192.168.6.112:5061 How would multiple users at a single device be handled? (copied from the example sip.conf). Set tlsbindaddr to the address to which to bind(2) the tls socket. tlsbindaddr=0.0.0.0 is typical in ipv4-only configs. -JimC Presumably this is equivalent to tlsbindaddr=0.0.0.0/0.0.0.0? Is the syntax tlsbindaddr=0.0.0.0/0.0.0.0:5061 is also correct? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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, SRTP, Asterisk11 and Snom870s
On Tue, March 3, 2015 13:37, James Cloos wrote: JBB == James B Byrne byrn...@harte-lyne.ca writes: JBB tcpenable=yes JBB tlsenable=yes JBB tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt JBB tlscafile=/etc/pki/tls/certs/ca-bundle.crt JBB tlsdontverifyserver=yes JBB tlscipher=ALL JBB tlsclientmethod=tlsv1 You are missing the tls key. The config name is tlsprivatekey; set that to the filename of your tls key, akin to how tlscertfile is set. -JimC Thank you. The settings in sip_general_additional.conf are now: tcpenable=yes tlsenable=yes tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.pem tlscafile=/etc/pki/tls/certs/ca-bundle.crt tlsdontverifyserver=yes tlscipher=ALL tlsclientmethod=tlsv1 tlsprivatekey=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.key However, issuing 'amportal a r' still results in this error: [2015-03-03 15:40:42] ERROR[13681]: tcptls.c:875 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.6.112:5060: Connection refused -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] IAXModem or T38Modem?
On Mon, March 24, 2014 01:41, Mike Diehl wrote: Hi all, I'm installing Hylafax on my Asterisk system. From what I've read, I can either use IAXModem or T38Modem to provide the virtual fax device. So at the risk of starting a religious war, which one should I use? I don't mind running IAX if I have to. I want as much flexibility and stability as I can get. So, what are your recommendations? Mike. We use IAXModem-1.2.0 built from source and packaged as an rpm using mock/rpmbuild together with Hylafax+-5.5.3 from epel. Since April 2013 this combination has been running two dedicated POTS lines through a TDM800-p8 on our Atom CentOS-6.3 based Asterisk-11.7.0 (current version) box without any reported difficulties (once I sorted out the upstart stanzas). As this is the only combination I have experience with it is the only one I can recommend. But it has proven very reliable so far as I am aware and I would be made aware pretty quickly if it was not. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Replying to Posts
On Thu, March 13, 2014 15:32, Kevin Larsen wrote: On 13/3/14 6:27 pm, Eric Wieling wrote: This is an example of why I top post. Who wrote what? +1-1 = 0 I do not care about where people put their replies so long as I can figure out who is answering what. What I do not like to read is this interminable religious dogma about the 'natural' order of writing. This is the second or third list this week in which this B.S. has shown up in my inbox. In written business communication, in contrast to tech-speak customarily found on mailing lists, ones answer always goes before any quoted context. Not because it has to, it is just that I have seldom, if ever, seen it done any other way. And regular business communication with non-technical folk comprises well over 75% of my daily written communication. And while I understand the cultural motivation behind the dogma of bottom posting I remain sceptical respecting its utility. Is there any objective evidence whatsoever that top or bottom posting makes any difference to the reader's understanding of the message? Does any rigorously determined data exist to support that contention? If not then this is simply a matter of trying to impose a set of arbitrary cultural values cloaked in the guise of technical superiority. Of course, if you use a mail client that's capable of quoting correctly, it all works beautifully. Outlook can quote correctly, but it is an all or nothing setting it would appear. Lotus Notes actually handles it better as there is a Reply option for normal email and a Reply With Internet-Style History that I use for this list. I don't have any problems following the rules of the list, but I am fully on the side of the Replies should go at the top group and would vote for a change in the rules. And do not even start on the Chevy vs. Ford debate respecting the technical superiority of Pine over Outlook. GAWD... Life its too short as it is. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Dial plan flow control
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4
FreePBX = 2.11.0.4
I am trying to understand flow control in Asterisk dial plans and not
having very much luck. I have read the Asterisk book from O'Rielly,
or at least those parts I believe might apply, but that has not helped
me much on this particular issue.
What I wish is to set three distinct ring tones on our Snom phones for
external, internal and transferred calls. The first is accomplished
simply enough by setting the ALERT_INFO setting on the inbound route
for all phones. Done.
The second was much more complicated but I found a recipe which
demonstrated how to do that using extensions_override_freepbx.conf and
eventually got it working. Done.
However, in my ignorance I believed that I could use the same
technique, indeed the same code, to check whether the call was
internal or a transfer. In this belief I appear sadly mistaken.
So, I am left with trying to understand the nature of flow control in
Asterisk dial plans and specifically those distributed with FreePBX.
In FreePBX I see this in extensions.conf:
;-
; from-internal:
;
; Internal dialplan that most internal phones have access to
;
[from-internal]
include = from-internal-noxfer
include = from-internal-xfer
include = bad-number ; auto-generated
;-
;-
; from-internal-noxfer:
;
; Place to put internal dialplan that should not be accessible during
; a blind transfer, this context will not be visible during such.
;
[from-internal-noxfer]
include = from-internal-noxfer-custom
include = from-internal-noxfer-additional ; auto-generated
;-
;-
; from-internal-xfer:
;
; Place to put most internal dialplan, will be visible during
: normal calls and blind transfers.
;
[from-internal-xfer]
include = from-internal-custom
include = from-internal-additional ; auto-generated
exten = s,1,Macro(hangupcall)
exten = h,1,Macro(hangupcall)
;-
What I would like to do is to add checks for whether or not a call is
internal or transferred between extensions in the
[from-internal-custom] context, which is presumably best placed in the
file named /etc/asterisk/extensions_custom.conf. To begin testing I
did this
[from-internal-custom]
include = set-snom-ringtone-variables
[set-snom-ringtone-variables]
exten = _X.,1,Noop(CALLERID_ALL=${CALLERID(all)})
exten = _X.,n,Set(CallerIDNum=${CALLERID(num)})
Which simply does not work at all. The effect is that the extensions
stop working. So, clearly I misunderstand something very basic about
flow control and thus my question. How do I return from my
from-internal-custom context back to the from-internal-xfer context at
the point following the include = from-internal-custom statement?
Thank you.
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
--
_
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[asterisk-users] What is my syntax error here?
I have thsi code in a dial plan. The purpose of which is to set
distinctive ring tones for internal and transferred calls.
exten = _.,1,Noop(CALLERID_ALL=${CALLERID(all)})
exten = _.,n,Set(CallerIDNum=${CALLERID(num)})
; This just shows a list of interesting variables and their values
; Comment it out when finished debugging
;include = macro-dumpvars
;exten = _.,n,Macro(dumpvars)
exten = _417XX,n,Set(AlertSnom=http://www.notused.com\;info=)
; alert-external, alert-group and alert-internal are
; Snom predefined values.
exten = _417XX,n,Set(AlertExternalCall=alert-external)
; alert_internal_call and alert_internal_transfer are
; locally customised values
exten = _417XX,n,Set(AlertInternalCall=alert_internal_call)
exten = _417XX,n,Set(AlertInternalTransfer=alert_internal_transfer)
exten = _417XX,n,Set(__ALERT_INFO=${AlertSnom}${AlertInternalTransfer})
exten = _417XX,n,GotoIf(
$[$[${CallerIDNum} 41799] |
$[${CallerIDNum} 41700]]?notfromlocal:)
exten = _417XX,n,Set(__ALERT_INFO=${AlertSnom}${AlertInternalCall})
This works for internal calls but not transfers and it only works at
all only because of the fall through structure. It contains this
error that I do not understand:
-- Executing [41720@from-internal:1] NoOp(SIP/41712-0548,
CALLERID_ALL=James B Byrne 41712) in new stack
-- Executing [41720@from-internal:2] Set(SIP/41712-0548,
CallerIDNum=41712) in new stack
-- Executing [41720@from-internal:3] Set(SIP/41712-0548,
AlertSnom=http://www.notused.com;info=) in new stack
-- Executing [41720@from-internal:4] Set(SIP/41712-0548,
AlertExternalCall=alert-external) in new stack
-- Executing [41720@from-internal:5] Set(SIP/41712-0548,
AlertInternalCall=alert_internal_call) in new stack
-- Executing [41720@from-internal:6] Set(SIP/41712-0548,
AlertInternalTransfer=alert_internal_transfer) in new stack
-- Executing [41720@from-internal:7] Set(SIP/41712-0548,
__ALERT_INFO=http://www.notused.com;info=alert_internal_transfer)
in new stack
-- Executing [41720@from-internal:8] GotoIf(SIP/41712-0548,
) in new stack
== Extension Changed 41712[ext-local] new state InUse for Notify
User 41714
[2013-07-24 09:50:42] WARNING[10630][C-6b44]: pbx.c:11544
pbx_builtin_gotoif: Ignoring, since there is no variable to check
[2013-07-24 09:50:42] WARNING[10630][C-6b44]: pbx.c:11544
pbx_builtin_gotoif: Ignoring, since there is no variable to check
-- Executing [41720@from-internal:9] Set(SIP/41712-0548,
__ALERT_INFO=http://www.notused.com;info=alert_internal_call)
in new stack
-- Executing [41720@from-internal:10] Goto(SIP/41712-0548,
from-internal-original-override,41720,1) in new stack
-- Goto (from-internal-original-override,41720,1)
So my question is simple. What error in syntax have I committed here?
I expect that CallerIDNum == 41712 in the check:
exten = _417XX,n,GotoIf(
$[$[${CallerIDNum} 41799] |
$[${CallerIDNum} 41700]]?notfromlocal:)
But I am getting a message say there is no variable to check. So what
I have done that is wrong?
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
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Re: [asterisk-users] What is my syntax error here?
Additional data:
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4
FreePBX = 2.11.0.4
-- Original Message --
Subject: What is my syntax error here?
From:James B. Byrne byrn...@harte-lyne.ca
Date:Wed, July 24, 2013 10:08
To: asterisk-users@lists.digium.com
--
I have thsi code in a dial plan. The purpose of which is to set
distinctive ring tones for internal and transferred calls.
exten = _.,1,Noop(CALLERID_ALL=${CALLERID(all)})
exten = _.,n,Set(CallerIDNum=${CALLERID(num)})
; This just shows a list of interesting variables and their values
; Comment it out when finished debugging
;include = macro-dumpvars
;exten = _.,n,Macro(dumpvars)
exten = _417XX,n,Set(AlertSnom=http://www.notused.com\;info=)
; alert-external, alert-group and alert-internal are
; Snom predefined values.
exten = _417XX,n,Set(AlertExternalCall=alert-external)
; alert_internal_call and alert_internal_transfer are
; locally customised values
exten = _417XX,n,Set(AlertInternalCall=alert_internal_call)
exten = _417XX,n,Set(AlertInternalTransfer=alert_internal_transfer)
exten = _417XX,n,Set(__ALERT_INFO=${AlertSnom}${AlertInternalTransfer})
exten = _417XX,n,GotoIf(
$[$[${CallerIDNum} 41799] |
$[${CallerIDNum} 41700]]?notfromlocal:)
exten = _417XX,n,Set(__ALERT_INFO=${AlertSnom}${AlertInternalCall})
This works for internal calls but not transfers and it only works at
all only because of the fall through structure. It contains this
error that I do not understand:
-- Executing [41720@from-internal:1] NoOp(SIP/41712-0548,
CALLERID_ALL=James B Byrne 41712) in new stack
-- Executing [41720@from-internal:2] Set(SIP/41712-0548,
CallerIDNum=41712) in new stack
-- Executing [41720@from-internal:3] Set(SIP/41712-0548,
AlertSnom=http://www.notused.com;info=) in new stack
-- Executing [41720@from-internal:4] Set(SIP/41712-0548,
AlertExternalCall=alert-external) in new stack
-- Executing [41720@from-internal:5] Set(SIP/41712-0548,
AlertInternalCall=alert_internal_call) in new stack
-- Executing [41720@from-internal:6] Set(SIP/41712-0548,
AlertInternalTransfer=alert_internal_transfer) in new stack
-- Executing [41720@from-internal:7] Set(SIP/41712-0548,
__ALERT_INFO=http://www.notused.com;info=alert_internal_transfer)
in new stack
-- Executing [41720@from-internal:8] GotoIf(SIP/41712-0548,
) in new stack
== Extension Changed 41712[ext-local] new state InUse for Notify
User 41714
[2013-07-24 09:50:42] WARNING[10630][C-6b44]: pbx.c:11544
pbx_builtin_gotoif: Ignoring, since there is no variable to check
[2013-07-24 09:50:42] WARNING[10630][C-6b44]: pbx.c:11544
pbx_builtin_gotoif: Ignoring, since there is no variable to check
-- Executing [41720@from-internal:9] Set(SIP/41712-0548,
__ALERT_INFO=http://www.notused.com;info=alert_internal_call)
in new stack
-- Executing [41720@from-internal:10] Goto(SIP/41712-0548,
from-internal-original-override,41720,1) in new stack
-- Goto (from-internal-original-override,41720,1)
So my question is simple. What error in syntax have I committed here?
I expect that CallerIDNum == 41712 in the check:
exten = _417XX,n,GotoIf(
$[$[${CallerIDNum} 41799] |
$[${CallerIDNum} 41700]]?notfromlocal:)
But I am getting a message say there is no variable to check. So what
I have done that is wrong?
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
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_
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Re: [asterisk-users] What is my syntax error here?
On Wed, July 24, 2013 10:33, James B. Byrne wrote:
Additional data:
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4
FreePBX = 2.11.0.4
. . .
So my question is simple. What error in syntax have I committed here?
I expect that CallerIDNum == 41712 in the check:
exten = _417XX,n,GotoIf(
$[$[${CallerIDNum} 41799] |
$[${CallerIDNum} 41700]]?notfromlocal:)
But I am getting a message say there is no variable to check. So what
I have done that is wrong?
As suggested I made these additions to the dial plan:
; Line 8
exten = _417XX,n,NoOp($[${CallerIDNum} 41799])
; Line 9
exten = _417XX,n,NoOp($[${CallerIDNum} 41700])
; Line 10
exten = _417XX,n,NoOp($[${CallerIDNum} 41799] |
$[${CallerIDNum} 41700])
; Line 11
exten = _417XX,n,NoOp($[${CallerIDNum} 41799] |
$[${CallerIDNum} 41700])
; Line 12
exten = _417XX,n,NoOp($[$[${CallerIDNum} 41799] ||
$[${CallerIDNum} 41700]])
; Line 13
exten = _417XX,n,NoOp($[$[${CallerIDNum} 41799] ||
$[${CallerIDNum} 41700]])
; Line 14 - original
exten = _417XX,n,GotoIf(
$[$[${CallerIDNum} 41799] ||
$[${CallerIDNum} 41700]]?notfromlocal:)
Which changed nothing but the results did provide a clue. Taking the
earlier suggestion I ensured that my original line did not contain
line breaks, which I cannot reproduce in this email because of its
length. However, putting everything on one line caused the missing
variable error to disappear.
exten = _417XX,n,GotoIf($[$[${CallerIDNum} 41799] ||
$[${CallerIDNum} 41700]]?notfromlocal:)
Thank you both for the help. I much appreciate it.
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
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[asterisk-users] Help with decyphering DND status
@192.168.6.9:5060,response=802e69ee4fca8e75636fcf72b6167069,algorithm=MD5 Expires: 3600 Content-Length: 0 Received from udp:192.168.6.9:5060 at 16/7/2013 11:56:16:683 (510 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.120:3072;branch=z9hG4bK-6lft658u13gn;received=192.168.6.120;rport=3072 From: sip:41720@192.168.6.9;tag=se3w15c5fb To: sip:41710@192.168.6.9;;tag=as6723ebb5 Call-ID: 455FE4519A6B01B916F8CAE0BE485B25-x08c2euu13zz CSeq: 64 SUBSCRIBE Server: FPBX-2.11.0(11.4.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: sip:41710@192.168.6.9:5060;expires=3600 Content-Length: 0 Received from udp:192.168.6.9:5060 at 16/7/2013 11:56:16:686 (694 bytes): NOTIFY sip:41720@192.168.6.120:3072;line=d241fk25 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK6c7fbe14;rport Max-Forwards: 70 From: sip:41710@192.168.6.9;;tag=as6723ebb5 To: sip:41720@192.168.6.9;tag=se3w15c5fb Contact: sip:41710@192.168.6.9:5060 Call-ID: 455FE4519A6B01B916F8CAE0BE485B25-x08c2euu13zz CSeq: 187 NOTIFY User-Agent: FPBX-2.11.0(11.4.0) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 207 ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=85 state=full entity=sip:41710@192.168.6.9 dialog id=41710 stateterminated/state /dialog /dialog-info Sent to udp:192.168.6.9:5060 at 16/7/2013 11:56:16:690 (300 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK6c7fbe14;rport=5060 From: sip:41710@192.168.6.9;;tag=as6723ebb5 To: sip:41720@192.168.6.9;tag=se3w15c5fb Call-ID: 455FE4519A6B01B916F8CAE0BE485B25-x08c2euu13zz CSeq: 187 NOTIFY User-Agent: snom870/8.7.4.8 Content-Length: 0 I see neither DND not CONNECTED an any of this. Either these phone logs are incomplete or what is passed over the wire differs from what is displayed in the Asterisk trace. I lack the knowledge to determine which is the case. In Asterisk -rvvv I see this (among musch else): == Extension Changed 41712[ext-local] new state InUse for Notify User 41720 -- Executing [*78@from-internal-original:2] Wait(SIP/41712-09e4, 1) in new stack -- Executing [*78@from-internal-original:3] Macro(SIP/41712-09e4, user-callerid,) in new stack -- Executing [s@macro-user-callerid:1] Set(SIP/41712-09e4, TOUCH_MONITOR=1373990250.7222) in new stack -- Executing [s@macro-user-callerid:2] Set(SIP/41712-09e4, AMPUSER=41712) in new stack . . . Set(SIP/41712-09e4, DB(DND/41712)=YES) in new stack -- Executing [*78@from-internal-original:5] Set(SIP/41712-09e4, STATE=BUSY) in new stack -- Executing [*78@from-internal-original:6] Gosub(SIP/41712-09e4, app-dnd-on,sstate,1()) in new stack -- Executing [sstate@app-dnd-on:1] Set(SIP/41712-09e4, DEVICE_STATE(Custom:DND41712)=BUSY) in new stack So, I am guessing that DB(DND/41712)=YES is telling the asterisk database to change the status but I am not sure where STATE=BUSY is going, perhaps BUSY=CONNECTED; and I cannot see where extension 41720 is being passed anything that resembles what I see in that unit's own trace log. Can somebody guide me through what it is I am seeing here? How do I pass DND to the other extensions? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] PoE L3 Switches
On Sun, July 14, 2013 18:36, bilal ghayyad wrote: Hello; Anyone used PoE L2 network switches other than cisco and recommend this for us? We need it to be stable and costly effective. Regards Bilal We use multiple Cisco SG100D-08P eight-port POE unmanaged switches with each located close to the end users, often on their desks. These devices are relatively cheap, seem to work well, and provide four POE ports plus four standard Ethernet ports. We find that this arrangement works for two or three employee work centres on each switch. So, it is a Cisco solution which you deprecate. However, it does work and it is both inexpensive and flexible. If distributing power from a central location over Ethernet is an absolute requirement then these probably are not what you want. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Setting the vkey background colour on Snom870
Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.4 Snom870 FW = 8.7.3.19 /8.4.8beta I would like to change the background colours on the BLF vkey field based on the station status. I posted the following to the Snom support forums some days back and have had no response so I am asking here in the hope than one or more of you have done something like this: I have the following code in a provisioning file vkey_red perm=BUSY IN_A_CALL IN_A_MEETING HOLDING/vkey_red vkey_orange perm=AWAY INACTIVE/vkey_orange vkey_green perm=AVAILABLE/vkey_green vkey_blue perm=DND/vkey_blue goto_virtual_keys_state_on_activity perm=on/goto_virtual_keys_state_on_activity I have a test phone with f/w 8.7.4.8 and loaded into it the provisioning file that include the instructions given above. When I place that phone on DND the BLF key for that extension does not show any colour change at all but the text changes to [talking]. As well, ringing, and talking actions on other extensions displayed in the BLF do not show any change in background colour at all although the text changes. On phones using 8.7.3.19 using the same provisioning file the background colour for ringing shows green and for talking shows orange but nothing else appears to change any colour and neither red nor blue appears used for anything. Is there an additional setting that I have to turn on in order to use vkey_colours on the BLF in 8.4.8? Is there any way to control the BLF background colours in Snom870s with f/w 8.3.7.19? Is there any way to set the hues to colours other than red/green/orange/blue? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Question on AEL2 string comparisons
On Thu, July 4, 2013 02:14, Satish Barot wrote:
${CALLERID(num)} should give you only number and not technology i.e.
41712.
Give this a shot,
exten = _417XX,n,Noop(CALLERIDNUM=${CALLERID(num)})
exten = _417XX,n,GotoIf($[$[${CALLERID(num)} 41799] |
$[${CALLERID(num)} 41700]]?notfromlocal:)
--Satish Barot
Ahmedabad, India
That works. Thank you.
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
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[asterisk-users] Custom dial plan for internal transfers of external calls
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4.0
FreePBX = 2.11.0.2
We use Snom870 handsets with firmware v.8.7.3.19.
I am trying to develop a custom dial plan to invoke a distinctive
ring-tone when an external call is transferred internally. Based on
an earlier solution I discovered I am attempting this:
[from-internal]
include = set-alert-if-local
[from-internal-original]
include = from-internal-xfer
include = bad-number
[set-alert-if-local]
. . .
exten = _417XX,n,GotoIf($[${CALLERID(num)}
SIP/41799]?notfromlocal)
exten = _417XX,n,GotoIf($[${CALLERID(num)}
SIP/41710]?notfromlocal)
;If we reach here then the caller is within the upper and lower bounds
exten = _417XX,n,Set(__ALERT_INFO=${AlertSnom}${AlertInternalTransfer})
exten = _417XX,n(notfromlocal),Goto(from-internal-original,${EXTEN},1)
;The following three lines must not be changed!
exten = _.,1,Goto(from-internal-original,${EXTEN},1)
exten = s,1,Goto(from-internal-original,s,1)
exten = h,1,Macro(hangupcall)
This context appears to be entered only when the call originates from
another extension. When a transfer of an external call is attempted
theis context does not seem to be entered.
The following abstracted asterisk trace log shows this for an incoming
call answered on 41712 and then transferred to 41720.
-- SIP/41711-0165 is ringing
-- SIP/41713-0167 is ringing
-- SIP/41712-0166 is ringing
-- SIP/41720-0169 is ringing
-- SIP/41718-0168 is ringing
== Extension Changed 41712[ext-local] new state InUse for Notify
User 41710
== Extension Changed 41712[ext-local] new state InUse for Notify
User 41711
-- SIP/41712-0166 connected line has changed. Saving it until
answer for DAHDI/1-1
-- SIP/41712-0166 answered DAHDI/1-1
== Extension Changed 41712[ext-local] new state InUse for Notify
User 41715
== Extension Changed 41712[ext-local] new state InUse for Notify
User 41717
== Extension Changed 41712[ext-local] new state InUse for Notify
User 41718
-- Executing [s@macro-auto-blkvm:1] Set(SIP/41712-0166,
__MACRO_RESULT=) in new stack
-- Executing [s@macro-auto-blkvm:2] Macro(SIP/41712-0166,
blkvm-clr,) in new stack
At this point the original incoming call is answered.
. . .
-- Executing [s@macro-auto-blkvm:3] ExecIf(SIP/41712-0166,
0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=41712)) in new stack
And then transferred to 41720
[2013-07-03 13:43:03] WARNING[7954][C-4685]: res_srtp.c:406
ast_srtp_unprotect: SRTP unprotect failed with: authentication failure
110
-- Stopped music on hold on DAHDI/1-1
== Extension Changed 41712[ext-local] new state Idle for Notify User
41710
== Extension Changed 41712[ext-local] new state Idle for Notify User
41711
== Extension Changed 41712[ext-local] new state Idle for Notify User
41715
== Spawn extension (from-internal-xfer, 41720, 1) exited non-zero on
'DAHDI/1-1' in macro 'dial'
== Spawn extension (from-internal-xfer, 41720, 1) exited non-zero on
'DAHDI/1-1'
-- Executing [41720@from-internal-xfer:1] Set(DAHDI/1-1,
__RINGTIMER=20) in new stack
And finally answered on 41720
[2013-07-03 13:43:16] DEBUG[29747][C-4685]: sip/sdp_crypto.c:310
sdp_crypto_offer: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:5D038u88tI6PLyruDovyQIku9PH7exEAL3Qolc9m
== Extension Changed 41720[ext-local] new state InUse for Notify
User 41711
== Extension Changed 41720[ext-local] new state InUse for Notify
User 41715
-- SIP/41720-016a answered DAHDI/1-1
It is evident from the trace that the context [set-alert-if-local] is
not entered on internal transfers and I lack the experience to
understand why. Can someone here enlighten me as to what is going on
in this instance and how I should change my contexts in order to check
for internal transfers of external calls?
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
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Re: [asterisk-users] Asterisk-11 loop behaviour
On Tue, June 25, 2013 09:57, Matthew Jordan wrote: On Mon, Jun 24, 2013 at 2:37 PM, James B. Byrne byrn...@harte-lyne.cawrote: It is not an infinite loop but it does go on for an inordinately long time. Does anyone here recognize what is happening and can provide me with an explanation? Since it is pbx_spool doing the processing, you probably have something creating a callfile in /var/spool/asterisk/outgoing on startup (or periodically). I did a quick Google search and found out that this particular context is used by FreePBX 2.9's Time Conditions feature - see http://www.freepbx.org/forum/freepbx-distro/distro-discussion-help/odd-failed-calls-in-logs for more information. Thank you. Could I ask what search term you used for google? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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-11 loop behaviour
Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvv': Channel Local/s@tc-maint-02a4;1 was answered. Launching NoCDR() on Local/s@tc-maint-02a4;1 [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 attempt_thread: Call completed to Local/s@tc-maint [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 attempt_thread: Call completed to Local/s@tc-maint == Spawn extension (tc-maint, s, 5) exited non-zero on 'Local/s@tc-maint-02a4;2' -- Attempting call on Local/s@tc-maint for application NoCDR() (Retry 1) -- Executing [s@tc-maint:1] NoCDR(Local/s@tc-maint-02a6;2, ) in new stack -- Executing [s@tc-maint:2] Set(Local/s@tc-maint-02a6;2, TCMAINT=RETURN) in new stack -- Executing [s@tc-maint:3] Gosub(Local/s@tc-maint-02a6;2, timeconditions,1,1()) in new stack -- Executing [1@timeconditions:1] GotoIfTime(Local/s@tc-maint-02a6;2, 08:00-17:00,mon-fri,*,*?truestate) in new stack -- Goto (timeconditions,1,9) -- Executing [1@timeconditions:9] GotoIf(Local/s@tc-maint-02a6;2, 0?falsegoto) in new stack -- Executing [1@timeconditions:10] ExecIf(Local/s@tc-maint-02a6;2, 0?Set(DB(TC/1)=)) in new stack -- Executing [1@timeconditions:11] Set(Local/s@tc-maint-02a6;2, DEVICE_STATE(Custom:TC1)=NOT_INUSE) in new stack -- Executing [1@timeconditions:12] ExecIf(Local/s@tc-maint-02a6;2, 0?Set(DEVICE_STATE(Custom:TCSTICKY)=INUSE)) in new stack -- Executing [1@timeconditions:13] GotoIf(Local/s@tc-maint-02a6;2, 0?ext-group,417,1) in new stack -- Executing [1@timeconditions:14] Set(Local/s@tc-maint-02a6;2, TCSTATE=true) in new stack -- Executing [1@timeconditions:15] Return(Local/s@tc-maint-02a6;2, ) in new stack -- Executing [s@tc-maint:4] System(Local/s@tc-maint-02a6;2, /var/lib/asterisk/bin/schedtc.php 60 /var/spool/asterisk/outgoing 1) in new stack -- Executing [s@tc-maint:5] Answer(Local/s@tc-maint-02a5;2, ) in new stack Channel Local/s@tc-maint-02a5;1 was answered. Launching NoCDR() on Local/s@tc-maint-02a5;1 [2013-06-24 15:22:32] NOTICE[32681]: pbx_spool.c:402 attempt_thread: Call completed to Local/s@tc-maint [2013-06-24 15:22:32] NOTICE[32681]: pbx_spool.c:402 attempt_thread: Call completed to Local/s@tc-maint It is not an infinite loop but it does go on for an inordinately long time. Does anyone here recognize what is happening and can provide me with an explanation? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] FreePBX, Asterisk and Twinkle - Testing a new setup
I am experimenting with Asterisk having downloaded and installed the FreePBX i386 CentOS-6.3 based distro and updated it. The current package level on this system is: asterisk11-11.3.0-49_centos6 freepbx-2.11.0beta2-112 I am using twinkle-1.4.2-7.el6 as a softphone testing tool. There is no firewall on the asterisk host and SELinux is disabled on it. Fail2Ban is installed but I have made no alterations to the default configuration, whatever it is. The asterisk host is configured as 192.168.6.122. The softphone is configured on a separate host with a routable IP on our 216.xxx.xxx/24 netblock. Both networks pass though an internal switch and are firewalled from the outside world by a centos-6.4 based gateway host using IPTables. I have no difficulty in connecting to the asterisk host either by ssh or by https. I have initialised the FreePBX config and have selected the user/device approach as this seems to fit our firm's employee requirements more closely than the extension based configuration. We have several employees who frequently telecommute. For the purposes of testing I have created two users, 11 and 12. I have configured a twinkle user profile for user 12. I can place a call to user 11 from twinkle and I get the IVR message for 'the number you have called is not in service'. I have tried to register Twinkle and this always fails. If I do : # asterisk -r CLI sip show peers Name/username HostDyn Forcerport ACL Port Status Description 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] CLI sip show users Username Secret Accountcode Def.Context ACL ForcerPort Which seems to say to me that I have nothing configured albeit I have tried to through FreePBX. At this point I am not trying to get a call out to our PTSN, although I have the FXO port plugged into a live analogue line. What I am trying to understand is the relationship between asterisk devices and users. The twinkle softphone has two lines (1 and 2). It seems to me that I should be able to configure each line as a separate extension and to call one from the other. What I cannot seem to discover is how to do it. Is it possible to do this? How is it done? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Which card to get?
We are investigating the possibility of using Asterisk in a KVM based virtual machine to handle connections to and from our HylaFax service. Our current set up uses a dedicated host with external fax modems. What I wish to know is what interface card would the list members recommend for a proof of concept trial? We currently have two incoming fax lines and five vox lines all POTS. Our physical internet connection is fiber but I could not tell you exactly what type of service it presently carries. It is upgradable to a considerable extent in any case. We are planning to move to VOIP as an adjunct to this project. This is secondary to getting the fax system moved but we would like to avoid having to install additional hardware for VOIP once the fax portion of project is complete and the service transferred. What are our options? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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
