Re: [asterisk-users] Issue with PRI connection
Hi, I have tried disabling and enabling crc4 before but that did not help. I have not defined any signalling value under chan_dahdi.conf Also, with respect to cabling we tried switching tx and rx but in that case we see alarm on the dahdhi status. -- _ -- 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] Issue with PRI connection
put signalling=euroisdn in chan_dahdi.conf and restart asterisk. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 On Mon, Sep 24, 2012 at 11:33 AM, Ashish Agarwal ashisha...@gmail.comwrote: Hi, I have tried disabling and enabling crc4 before but that did not help. I have not defined any signalling value under chan_dahdi.conf Also, with respect to cabling we tried switching tx and rx but in that case we see alarm on the dahdhi status. -- _ -- 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] CDR Unanswered calls
Dear friends, I am not able to capture the CDR records for unanswered calls. Only one record per call is coming in the CDR table. In the cdr.conf, I have enabled it by setting : unanswered = yes From asterisk CLI also, I am getting this : centos55server*CLI cdr show status Call Detail Record (CDR) settings -- Logging: Enabled Mode: Simple Log unanswered calls: Yes * Registered Backends --- mysql cdr-custom But still its not coming in the table. I am using My SQL CDR. All other records are coming properly. We need these records, because its a call center and the management requires details of the extnesions which were not answering calls in the queue. Is there any other setting to be done ? In Asterisk 1.4 it was working fine, but now I am using asterisk-1.8.12.1 Kindly help. Regards Shanavaz.-- _ -- 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] accept email and make phone call?
Already implemented Email to Fax in ICTFAX http://www.ictfax.org using both sendmail and drupal mail handler module , you need to modify Fax part with Voice call Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation. On Fri, Sep 21, 2012 at 3:06 PM, Joseph Acquisto j...@j4computers.comwrote: On 9/21/2012 at 4:00 AM, Jeremy Kister asterisk...@jeremykister.com wrote: On 9/20/2012 1:31 PM, Joseph Acquisto wrote: Any ideas on how asterisk could accept an email (such as an email to SMS or num...@mybox.org sort of thing) and make a phone call to a specific number and make an announcement? that's actually what my jkSMS package does. i don't know if it'd be useful out of the box, depending on what you're trying to do. http://jeremy.kister.net/code/asterisk/jkSMS Jeremy Kister http://jeremy.kister.net./ I will take a look at it and certainly look at all the other suggestions as well. Thanks to all for your response. joe a. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with PRI connection
On Sunday 23 September 2012, Ashish Agarwal wrote: For some reason pri show spans does not show up. Can someone assist me to fix this issue. DAHDI is fussy about its configuration files. A single misconfiguration anywhere can break the whole thing, so get it working with just one span first. Make sure you set the jumpers on the card correctly for E1 operation (they are often set as T1 on delivery) and specify the correct channel to use as a D-channel (usually 16 on span 1; 1-15 and 17-31 are the B-channels. Add 31 for span 2, and so forth) and the correct signalling system. Ask someone at your telco if not certain. Another thing that can go wrong is the cabling from the NTE to the Asterisk box. Sometimes you need a straight-through cable, other times you need a crossover. Usually the active pairs are 4 and 5, and 2 and 1; so you need to cross over 1 with 5 and 2 with 4. Sometimes the polarity on pins 4 and 5 is wrong (this is the only pair with the lower-numbered pin coloured and the higher-numbered pin white) and sometimes the equipment is bothered about this, so try also a crossover from 1-4 and 2-5. Again, ask someone from your telco if not certain. Once you've got span 1 working, bring up the others one by one; and don't be afraid to grow your configs with comments where you have commented out lines and/or added explanations of what you have changed. It's always easier to take extraneous stuff out later than to try to remember what you were thinking before something interesting happened. -- AJS Answers come *after* questions. -- _ -- 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] Peculiar problem with failover provision.
I have noticed a peculiar problem recently with the way that the failover operates in my dialplan. I normally have: 1,Dial(SIP/provider-1/extension) n,Dial(SIP/provider-2/extension) (or something similar). This has up until now worked flawlessly. If there is an error with the first provider, the call is completed with the second one. Now, what is happening is, if the remote party hags up first, then the call progresses to the next priority and re-dials them. Is this a change in default behaviour? Do I need to add a particular flag / config directive to my dialplan I am running Asterisk 10.6.0. Thanks for any help in solving this. -- _ -- 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] Peculiar problem with failover provision.
Why not use the DIALSTATUS channel variable to determine if a fail over is necessary? - Logan On Sep 24, 2012 6:00 AM, Thomas Kenyon dig...@sanguinarius.co.uk wrote: I have noticed a peculiar problem recently with the way that the failover operates in my dialplan. I normally have: 1,Dial(SIP/provider-1/**extension) n,Dial(SIP/provider-2/**extension) (or something similar). This has up until now worked flawlessly. If there is an error with the first provider, the call is completed with the second one. Now, what is happening is, if the remote party hags up first, then the call progresses to the next priority and re-dials them. Is this a change in default behaviour? Do I need to add a particular flag / config directive to my dialplan I am running Asterisk 10.6.0. Thanks for any help in solving this. -- __**__**_ -- 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-**usershttp://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] Help Required IVR
Hello everyone, I stuck in problem I have creating a time based IVR and its working fine. If my IVR playing in office hour it would standard IVR and if not they we have play a greeting message and place that call to voice mail of a extension. My problem is this I am able to transfer the call on voice mail but how to play greeting message first. I am using trixbox 2.2.8 anyone help is this regard would great full. -- Thanks Farooq Hussain -- _ -- 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] R: SIP CANCEL, Reason
Hi Jordan,
Thanks for all, but i found this bug in Asterisk :
https://issues.asterisk.org/jira/browse/ASTERISK-16465
Attached the patch to fix the problem, if the online site does not work.
Thanks for all
Best Regards
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Matthew Jordan
Inviato: giovedì 20 settembre 2012 13:42
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] SIP CANCEL, Reason
- Original Message -
From: Marco Colombo mcolo...@enter.it
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 19, 2012 10:51:43 AM
Subject: [asterisk-users] SIP CANCEL, Reason
Hi All!
i have a problem with asterisk 1.8.11.
I must have in the SIP cancel message, the line “Reason”
Example : Reason : SIP;cause=16;text=”Normal Call Clearing”
I have already enable “use_q850_reason=yes”, but this not work.
In my dialplan I have already add : exten =
_X.,n,Hangup(${HANGUPCAUSE})
Can anyone help me?
I don’t know what to do
The use_q850_reason settings applies globally. If you execute sip show
settings, what is the value of the Q.850 Reason header?
If you enable 'sip set debug on', what is the actual CANCEL request sent to the
UA?
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com http://asterisk.org
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Index: chan_sip.c
===
--- chan_sip.c (revision 280339)
+++ chan_sip.c (working copy)
@@ -12514,8 +12514,19 @@
}
reqprep(resp, p, sipmethod, seqno, newbranch);
- if (sipmethod == SIP_CANCEL p-answered_elsewhere) {
- add_header(resp, Reason, SIP;cause=200;text=\Call
completed elsewhere\);
+ if (sipmethod == SIP_CANCEL) {
+ if (p-answered_elsewhere) {
+ if (ast_test_flag(p-flags[1], SIP_PAGE2_Q850_REASON))
+ add_header(resp, Reason,
Q.850;cause=200;text=\Call completed elsewhere\);
+ else
+ add_header(resp, Reason,
SIP;cause=200;text=\Call completed elsewhere\);
+ }
+ else if (ast_test_flag(p-flags[1], SIP_PAGE2_Q850_REASON)
p-hangupcause) {
+ char buf[50];
+
+ sprintf(buf, Q.850;cause=%i, p-hangupcause 0x7f);
+ add_header(resp, Reason, buf);
+ }
}
return send_request(p, resp, reliable, seqno ? seqno : p-ocseq);
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[asterisk-users] Real time voice quality impairments detection
Dear List memebers, We as voice quality testing sotware vendor are very interested in your opinion on demands (Asterisk owners, call centers, etc) for a voice quality impairments detection library we plan to release. Currently the library can detect: - clicking - clipping - stuck - SNR - echo The library can work in real time supporting approximately 100-2000 ports on average server. Your feedback is greatly appreciated and will have a significant impact on our further research. Best regards, Sevana Oy, Finland Vendor of AQuA (Audio Quality Analyzer) http://www.sevana.fi -- _ -- 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 order of operations
Hello all,
I inherited an Asterisk 1.2 machine and I have a question about the order
of operations.
I want to give people the ability to dial specifics and block others. For
example, lets say NYC
[allowed]
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
exten = _1212.,s,Goto(denied,s,1)
[denied]
exten = s,1,Playback(num-outside-area)
exten = s,2,Hangup
What I would like to do it allow a specific and deny the rest. Mind you the
allowed will be everything EXCEPT what is allowed. My question is, will the
above work? Please don't comment on upgrading, this is an inherited system
which I cannot update.
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Re: [asterisk-users] Dial plan order of operations
On Monday 24 September 2012, Asterisk Newb wrote:
Hello all,
I inherited an Asterisk 1.2 machine and I have a question about the order
of operations.
I want to give people the ability to dial specifics and block others. For
example, lets say NYC
[allowed]
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
exten = _1212.,s,Goto(denied,s,1)
[denied]
exten = s,1,Playback(num-outside-area)
exten = s,2,Hangup
What I would like to do it allow a specific and deny the rest. Mind you the
allowed will be everything EXCEPT what is allowed. My question is, will the
above work? Please don't comment on upgrading, this is an inherited system
which I cannot update.
Asterisk always tests against the most specific (= hardest-to-match)
wildcarded extensions first, regardless of the actual order in the dialplan.
Since _1212555. is harder to match than _1212., the former will be tested
first.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] Dial plan order of operations
On Mon, Sep 24, 2012 at 12:43 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
Asterisk always tests against the most specific (= hardest-to-match)
wildcarded extensions first, regardless of the actual order in the
dialplan.
Since _1212555. is harder to match than _1212., the former will be tested
first.
--
AJS
So then the proper way would be
[allowed]
exten = _1212321.,s,Goto(denied,s,1)
exten = _1212333.,s,Goto(denied,s,1)
exten = _1212456.,s,Goto(denied,s,1)
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
[denied]
exten = s,1,Playback(num-outside-area)
exten = s,2,Hangup
?
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Re: [asterisk-users] Peculiar problem with failover provision.
You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on the internet do it the same way. It is still wrong. When you do a Dial on the dialplan you need check the value of DIALSTATUS or HANGUPCAUSE before dialing again. Both variables will give you some indication of why the first call ended. Then your dialplan logic can decide how to proceed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon Sent: Monday, September 24, 2012 7:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peculiar problem with failover provision. I have noticed a peculiar problem recently with the way that the failover operates in my dialplan. I normally have: 1,Dial(SIP/provider-1/extension) n,Dial(SIP/provider-2/extension) (or something similar). This has up until now worked flawlessly. If there is an error with the first provider, the call is completed with the second one. Now, what is happening is, if the remote party hags up first, then the call progresses to the next priority and re-dials them. Is this a change in default behaviour? Do I need to add a particular flag / config directive to my dialplan I am running Asterisk 10.6.0. Thanks for any help in solving this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peculiar problem with failover provision.
I think a lot of people leave it out in examples for simplicity's sake. It doesn't instil proper practices in folks' heads. - Logan On Sep 24, 2012 12:06 PM, Eric Wieling ewiel...@nyigc.com wrote: You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on the internet do it the same way. It is still wrong. When you do a Dial on the dialplan you need check the value of DIALSTATUS or HANGUPCAUSE before dialing again. Both variables will give you some indication of why the first call ended. Then your dialplan logic can decide how to proceed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon Sent: Monday, September 24, 2012 7:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peculiar problem with failover provision. I have noticed a peculiar problem recently with the way that the failover operates in my dialplan. I normally have: 1,Dial(SIP/provider-1/extension) n,Dial(SIP/provider-2/extension) (or something similar). This has up until now worked flawlessly. If there is an error with the first provider, the call is completed with the second one. Now, what is happening is, if the remote party hags up first, then the call progresses to the next priority and re-dials them. Is this a change in default behaviour? Do I need to add a particular flag / config directive to my dialplan I am running Asterisk 10.6.0. Thanks for any help in solving this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with PRI connection
On Mon, Sep 24, 2012 at 09:02:47AM +0100, A J Stiles wrote: Make sure you set the jumpers on the card correctly for E1 operation (they are often set as T1 on delivery)... A minor FYI not directly related to this discussion: Since DAHDI-Linux 2.6.0 the jumpers can be overridden in the wcte12xp [1] and wct4xxp [2] drivers with the default_linemode module parameter. Before that it could be set with the t1e1override module parameter [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10243 [2] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10269 The code in the drivers is present to set the linemode as part of the normal DAHDI configuration files (i.e. processed with dahdi_cfg) but the method of expressing that settings is not there yet. The vision is that dahdi_genconf will be able to pick a default line mode based on the locale information on the server. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] Dial plan order of operations
[allowed]
exten = _1212321.,s,Goto(denied,s,1)
exten = _1212333.,s,Goto(denied,s,1)
exten = _1212456.,s,Goto(denied,s,1)
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
[denied]
exten = s,1,Playback(num-outside-area)
exten = s,2,Hangup
?
My final question to the list (hopefully) will be, why doesn't this work as
documented (voip-wiki, etc):
My dialplan (Asterisk 1.2 line)
exten = _212555.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212444.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212444.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212333.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212333.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = 102,1,hangup
When I try calling, I am prompted for the pin, I enter it 3 time and rather
than it go to n+101 it allows the call through.
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Re: [asterisk-users] Dial plan order of operations
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Newb
Sent: Monday, September 24, 2012 1:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial plan order of operations
[allowed]
exten = _1212321.,s,Goto(denied,s,1)
exten = _1212333.,s,Goto(denied,s,1)
exten = _1212456.,s,Goto(denied,s,1)
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
[denied]
exten = s,1,Playback(num-outside-area)
exten = s,2,Hangup
?
My final question to the list (hopefully) will be, why doesn't this work as
documented (voip-wiki, etc):
My dialplan (Asterisk 1.2 line)
exten = _212555.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212444.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212444.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212333.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212333.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = 102,1,hangup
When I try calling, I am prompted for the pin, I enter it 3 time and rather
than it go to n+101 it allows the call through.
Not to be harsh, but the wiki information is buyer beware unless it is by
one of the developers or a select group of contributors. Lots of things in
the wikis work as written, but may fail if the wrong tweak is applied.
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[asterisk-users] Asterisk as a E1/T1 FXO ?
Hi, Does anybody here have experience using Asterisk as an FXO to emulate a E1/T1/PRI line for test purpose? Gateway or PCI card? Thank you! HB -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan order of operations
Going to n+101 was deprecated in Asterisk 1.2 or 1.4. Don't use it.. Read the
docs for Authenticate and see what diaplan variables you can check.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Newb
Sent: Monday, September 24, 2012 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial plan order of operations
[allowed]
exten = _1212321.,s,Goto(denied,s,1)
exten = _1212333.,s,Goto(denied,s,1)
exten = _1212456.,s,Goto(denied,s,1)
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
[denied]
exten = s,1,Playback(num-outside-area)
exten = s,2,Hangup
?
My final question to the list (hopefully) will be, why doesn't this work as
documented (voip-wiki, etc):
My dialplan (Asterisk 1.2 line)
exten = _212555.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212444.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212444.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212333.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212333.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = 102,1,hangup
When I try calling, I am prompted for the pin, I enter it 3 time and rather
than it go to n+101 it allows the call through.
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Re: [asterisk-users] Dial plan order of operations
On Mon, Sep 24, 2012 at 2:55 PM, Eric Wieling ewiel...@nyigc.com wrote:
Going to n+101 was deprecated in Asterisk 1.2 or 1.4. Don't use it..
Read the docs for Authenticate and see what diaplan variables you can
check.
Thanks, situated the problem with the following:
exten = _212555.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212444.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212444.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212333.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212333.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _X.,102,hangup
; exten = 102,1,hangup --- my screw up
So when someone dials a number to a dest (212555{444{333}}) they're asked
for a PIN 3x, if it fails now it hangs up
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[asterisk-users] confbridge command not found
Currently running version 1.8.16.0 and trying to manage confbridge rooms and users. When I try to use the confbridge cli command I get a command not found error. CLI confbridge No such command 'confbridge' (type 'core show help confbridge' for other possible commands) I've tried googling this but did not get anywhere. How can I enable the confbridge commands? Thanks!-- _ -- 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] confbridge command not found
Gary Carr wrote: Currently running version 1.8.16.0 and trying to manage confbridge rooms and users. When I try to use the confbridge cli command I get a command not found error. CLI confbridge No such command 'confbridge' (type 'core show help confbridge' for other possible commands) I've tried googling this but did not get anywhere. How can I enable the confbridge commands? Thanks! Well, first of all, Asterisk 1.8's version of confbridge doesn't register any CLI commands for confbridge. You need to be using version 10 or higher to get those commands. Aside from that, it's worth noting that even if you have the right version of Asterisk, just 'confbridge' is still not a CLI command and you can see the various confbridge commands by attempting to tab complete confbridge (type 'confbridge', don't hit enter but do hit tab once or twice and see what gets listed). I hope that helps. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 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] Dial plan order of operations
On Mon, 24 Sep 2012, Asterisk Newb wrote:
Thanks, situated the problem with the following:
exten = _212555.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
Two suggestions:
1) Using the 'n' priority will make your dialplans more maintainable.
2) Using a more 'explicit' pattern like '_212555' will result in a
more responsive dialplan* because Asterisk 'knows' it is looking for a 10
digit number and won't have to 'wait' to see if it has the whole number.
(For circuits that don't deliver the DID/DNIS all at once.)
Using an 'open ended' pattern could also expose you to unexpected
outcomes.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Realtime Hints
Hello, I'd like to start using realtime hints in my asterisk 1.8 dialplan, but I am unable. I haven't understood if they have to be put inside the extensions realtime table (with priority -1) or if a dedicated realtime hints table can be made. Neither ways seem to work. Have you any working example? Leandro -- _ -- 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] Realtime Hints
On Mon, 24 Sep 2012 23:37:32 +0200 Leandro Dardini ldard...@gmail.com wrote: I'd like to start using realtime hints in my asterisk 1.8 dialplan, but I am unable. I haven't understood if they have to be put inside the extensions realtime table (with priority -1) or if a dedicated realtime hints table can be made. Neither ways seem to work. Have you any working example? Did you try priority 'hint'? -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] Peculiar problem with failover provision.
Eric Wieling wrote: You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on the internet do it the same way. It is still wrong. When you do a Dial on the dialplan you need check the value of DIALSTATUS or HANGUPCAUSE before dialing again. Both variables will give you some indication of why the first call ended. Then your dialplan logic can decide how to proceed. Thanks for your help. In previous versions of asterisk it worked, and iirc after the called party hung up, the dialplan only progressed if there was a particular flag used with Dial (g?). It's going to cause a heck of a headache but I'll look into doing this properly in the week. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon Sent: Monday, September 24, 2012 7:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peculiar problem with failover provision. I have noticed a peculiar problem recently with the way that the failover operates in my dialplan. I normally have: 1,Dial(SIP/provider-1/extension) n,Dial(SIP/provider-2/extension) (or something similar). This has up until now worked flawlessly. If there is an error with the first provider, the call is completed with the second one. Now, what is happening is, if the remote party hags up first, then the call progresses to the next priority and re-dials them. Is this a change in default behaviour? Do I need to add a particular flag / config directive to my dialplan I am running Asterisk 10.6.0. Thanks for any help in solving this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Hints
We use something like below
[blf]
exten =_ZXX!,hint,SIP/${ODBC_FINDEXTN(sd.name,${EXTEN})}
This uses an odbc call to create the hint when the phone asks for it.
Using snom 760 and 821
Cheers
Stephen
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[asterisk-users] T.38 gateway ATA
Hoping for some clarification. I would like to setup a NORMAL (not T.38) fax machine on an ATA, and have the ATA be a T.38 gateway to a remote asterisk (1.8) server, which is doing T.38 relay (passthru) to a provider. Some amount of googling today seems to imply that most ATAs are just T.38 passthru devices, and expect a T.38 capable fax machine, otherwise just fallback to ulaw (and mostly fail, in my experience so far). So does anyone use an ATA that actually does the gateway transcoding to a normal fax machine? Or am I barking up the wrong tree? Thanks! 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] Need to Help for setup Video IVRS on Asterisk
Hi Mitul, Thanks for reply. If you can possible then please let me know download location and steps or document for setup Video IVRS on Asterisk. Thanks in advance. -- Best Regards, Rajni Vanza Consultant Technology --- Working On Linux,C/C++,VoIP Technology On Thu, Sep 20, 2012 at 2:42 PM, Mitul Limbani mi...@enterux.in wrote: this is quite complicated to be setup. however you can try using : asterisk 1.4.11 with libpri patch for h324m and app_h324m. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 On Thu, Sep 20, 2012 at 2:19 PM, RAJNI VANZA rajniva...@gmail.com wrote: Hi All; I was wondering if anyone has any experience for Video IVRS on Asterisk. 3G User dial some number and he is able to see Video IVRS on his mobile. I need to setup Video IVRS on Asterisk PBX. So, please suggest me best solutions or way for achieve this requirement. If its possible on Asterisk then which version is use for that. Thanks in Advance. -- Best Regards, Rajni Vanza Software Engineer --- Working On Linux,C/C++,VoIP Technology -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenVox A400P+asterisk=problem with dial recall tone
Hello, all! My name is Andrew. I have OpenVox TDM400P card with 2 modules (FXO and FXS). It plugged to server asterisk (1.8.13.1) under gentoo (3.3.8-gentoo) via dahdi (2.6.1 from gentoo portages). All works, but I have a problem: When I whant to transfer call from FXS to other phone after press flash in asterisk logs I see WARNING[21899]: sig_analog.c:3227 __analog_handle_event: Unable to start dial recall tone on channel 4. And there no recall tone in phone. Nevertheless, if I enter number of other phone and hang up - call transfers success. That is, problem is only: no recall tone. I wrote to support openvox, but there is no result - they advised me change gentoo+asterisk to elastics Please help me. What I do wrong? Thanks for advance Andrew -- Написано в почтовом клиенте браузера Opera: http://www.opera.com/mail/ -- _ -- 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
