Re: [asterisk-dev] [Code Review] 4413: Testsuite: Simulate phones and control from YAML.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4413/#review14450 --- /asterisk/trunk/lib/python/asterisk/phones.py https://reviewboard.asterisk.org/r/4413/#comment24971 I think this is a good case for using is instead of equality. I imagine that callers of this function are going to be passing a reference to the same account object that is stored on the phone_obj rather than a separate object with equivalent properties. Also, to avoid the continue statement, you can just use a positive comparison instead of a negative one: if account is phone_obj.account: return phone_obj /asterisk/trunk/lib/python/asterisk/phones.py https://reviewboard.asterisk.org/r/4413/#comment24970 A couple of PEP 8 points: 1) When catching exceptions, use the syntax except ErrorType as ErrorInstance instead of except ErrorType, ErrorInstance. So here, it would be except pj.Error as err 2) A PEP 8 checker will complain about the indentation of str(err)) on the final line I've highlighted. It will claim it should be lined up like this: raise Exception(Exception occurred while making call: '%s' % str(err)) See how str is lined up to show how it belongs to the parentheses? /asterisk/trunk/lib/python/asterisk/phones.py https://reviewboard.asterisk.org/r/4413/#comment24972 I suggest breaking this into two separate functions: blind_transfer and attended_transfer. 1) The nomenclature for the transfer_type has to be looked up in the source in order to know what values are accepted. 2) The transfer_uri is only useful for blind transfers. By having two separate functions, you can more clearly group which parameters are relevant and which are irrelevant for a specific type of transfer. So the attended transfer, as currently defined, would take no parameters and a blind transfer would take a transfer URI. If it became useful later to optionally specify that an attended transfer should go the other direction or should affect different calls, that could be added solely to the attended transfer function. /asterisk/trunk/lib/python/asterisk/phones.py https://reviewboard.asterisk.org/r/4413/#comment24973 self.current_call doesn't seem to be used for anything. /asterisk/trunk/lib/python/asterisk/phones.py https://reviewboard.asterisk.org/r/4413/#comment24974 Should this only be done if self.phone is None? Seems like this would be redundant after the initial setting. /asterisk/trunk/lib/python/asterisk/phones.py https://reviewboard.asterisk.org/r/4413/#comment24975 I suspect it's not an issue, but is there ever a chance that self.call could be None when the on_state() callback is called? /asterisk/trunk/lib/python/asterisk/phones.py https://reviewboard.asterisk.org/r/4413/#comment24976 This seems overly complicated for what you are trying to do here. Why not just try: self.phone.calls.remove(self.call) except ValueError: # Log an error or something If that doesn't work because equality of calls doesn't evaluate as you'd expect, then this is a situation where I expect that a single object reference is being passed around everywhere rather than separate call objects. You could take the approach of reconstructing the self.phone.calls list like this: self.phone.calls = [call for call in self.phone.calls if call is not self.call] /asterisk/trunk/lib/python/asterisk/phones.py https://reviewboard.asterisk.org/r/4413/#comment24977 I think this needs to be altered to always call self.call.hangup() if final is 1. If a transfer attempt fails, the phone is still responsible for hanging up the call in that situation. So if you perform a blind transfer to a non-existent extension in the dialplan, or you perform a blind transfer to an extension that exists but hangs up with Congestion() or Busy() for some reason, the phone still needs to hang up the call when receiving the final transfer response. /asterisk/trunk/lib/python/asterisk/phones.py https://reviewboard.asterisk.org/r/4413/#comment24978 This function's name is misleading because it does not actually log anything. It should either 1) Be renamed to something indicating that it is just gathering call information into a string or 2) Be altered to actually log the call info using LOGGER.debug() /asterisk/trunk/lib/python/asterisk/phones.py https://reviewboard.asterisk.org/r/4413/#comment24979 Can you add a comment about why the lock is being used here? /asterisk/trunk/lib/python/asterisk/phones.py https://reviewboard.asterisk.org/r/4413/#comment24980 If this exception occurs, does lock get properly unlocked? /asterisk/trunk/lib/python/asterisk/phones.py
[asterisk-dev] Reject incoming call
Hi Friends,
I am trying to implement a simple dial plan with asterisk.
1. Ring the inbound call
2. wait for 2 seconds
3. call agi script with cli
4. hangup
But when it gets hangup I see the Declined is passed from the asterisk.
But my PSTN provider keep waiting in dialing state with no noise until 60
seconds.
any idea what I have done wrong.
here is my dial plan
exten = 01,1,Ringing()
exten = 01,2,Wait(2)
exten =
01,3,Set(vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=${CALLERID(num)})
exten = 01,4,Agi(agi://127.0.0.1/url=${vxmlurl})
exten = 01,5,Hangup()
Please advice,
Best Regards,
Roy.
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Re: [asterisk-dev] Reject incoming call
Who is responsible to answer ??
On Feb 12, 2015 6:05 AM, Raj Roy Ghandhi roy.gan...@gmail.com wrote:
Hi Friends,
I am trying to implement a simple dial plan with asterisk.
1. Ring the inbound call
2. wait for 2 seconds
3. call agi script with cli
4. hangup
But when it gets hangup I see the Declined is passed from the asterisk.
But my PSTN provider keep waiting in dialing state with no noise until 60
seconds.
any idea what I have done wrong.
here is my dial plan
exten = 01,1,Ringing()
exten = 01,2,Wait(2)
exten =
01,3,Set(vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=${CALLERID(num)})
exten = 01,4,Agi(agi://127.0.0.1/url=${vxmlurl}
http://127.0.0.1/url=$%7Bvxmlurl%7D)
exten = 01,5,Hangup()
Please advice,
Best Regards,
Roy.
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[asterisk-dev] asterisk + voiceglue in Rasbery PI
Hi Friends, would like to know if anybody did install Asterisk + voiceglue into Raspberry PI B module. Is it possible? Please let em know. Best Regards, Roy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Reject incoming call
Hi Roy,
Hi Friends,
I am trying to implement a simple dial plan with asterisk.
1. Ring the inbound call
2. wait for 2 seconds
3. call agi script with cli
4. hangup
But when it gets hangup I see the Declined is passed from the asterisk.
But my PSTN provider keep waiting in dialing state with no noise until 60
seconds.
any idea what I have done wrong.
here is my dial plan
exten = 01,1,Ringing()
exten = 01,2,Wait(2)
exten =
01,3,Set(vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=${CALLERID(num)})
exten = 01,4,Agi(agi://127.0.0.1/url=${vxmlurl})
exten = 01,5,Hangup()
Did you try to put a specific Clear Cause code to the Hangup command ?
For example, Hangup(17) means User Busy, i.e. 486 Busy Here on SIP, or
Hangup(1) means Unallocated Number or 404 Not Found etc. If you want to
find more codes, search for ISDN Cause Codes. Your PSTN provider will
probably react better to other causes than the default one, which is
probably 16 Normal Call Clearing.
With regards,
Pavel
Please advice,
Best Regards,
Roy.
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Re: [asterisk-dev] [Code Review] 4352: testsuite: Add tests for ARI redirect; PJSIP Transfer
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4352/ --- (Updated Feb. 12, 2015, 2:47 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 6397 Bugs: ASTERISK-24015 and ASTERISK-24703 https://issues.asterisk.org/jira/browse/ASTERISK-24015 https://issues.asterisk.org/jira/browse/ASTERISK-24703 Repository: testsuite Description --- This patch adds tests for https://reviewboard.asterisk.org/r/4316/, which includes both tests for PJSIP's .transfer channel callback and the ARI /channels/[id]/redirect operation. *PJSIP Tests* - Test transferring an unanswered channel to a PJSIP endpoint, which responds to the initial INVITE request with a 302 - Test transferring an answered channel to a PJSIP endpoint, which sends a REFER request to the target - Test transferring an unanswered channel to a SIP URI via PJSIP, which responds to the initial INVITE request with a 302 - Test transferring an answered channel to a SIP URI via PJSIP, which sends a REFER request to the target *ARI Tests* - Off-nominal testing of the new operation, verifying that the various off nominal error response codes are returned as expected - Nominal testing of the operation. For fun, this spawns four Asterisk instances (one call generator, one load balancer, and two destinations) - and proceeds to load balance 'calls' between the two destination instances. As a pre-emptive note: (1) The off-nominal test makes use of the ARI event matcher, as it requires a PJSIP channel and tests off nominal error response codes. Along with needing to originate a second channel, the Python callback for this is relatively self contained and limited, both of which remove most of the benefit of driving the whole thing in YAML. (2) The nominal test is written in old style - a single run-test. I can't really see how anyone would re-use portions of it, but it was a fun test to write to show the power of the new operation - plus it does exercise the operation quite a lot. Diffs - /asterisk/trunk/tests/rest_api/channels/tests.yaml 6332 /asterisk/trunk/tests/rest_api/channels/redirect/tests.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/off-nominal/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/off-nominal/redirect_off_nominal.py PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/off-nominal/configs/ast1/pjsip.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/off-nominal/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/run-test PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast4/pjsip.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast4/http.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast4/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast3/pjsip.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast3/http.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast3/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast2/pjsip.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast2/http.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast2/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast1/pjsip.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast1/http.conf PRE-CREATION /asterisk/trunk/tests/rest_api/channels/redirect/nominal/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/transfers/tests.yaml 6332 /asterisk/trunk/tests/channels/pjsip/transfers/asterisk/uri/tests.yaml PRE-CREATION /asterisk/trunk/tests/channels/pjsip/transfers/asterisk/uri/refer/test-config.yaml PRE-CREATION /asterisk/trunk/tests/channels/pjsip/transfers/asterisk/uri/refer/sipp/alice.xml PRE-CREATION /asterisk/trunk/tests/channels/pjsip/transfers/asterisk/uri/refer/configs/ast1/pjsip.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/transfers/asterisk/uri/refer/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/transfers/asterisk/uri/redirect/test-config.yaml PRE-CREATION /asterisk/trunk/tests/channels/pjsip/transfers/asterisk/uri/redirect/sipp/alice.xml PRE-CREATION
Re: [asterisk-dev] [Code Review] 4316: ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis dialplan application to another system; improve and fix PJSIP's transfer ability
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4316/ --- (Updated Feb. 12, 2015, 2:33 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers and Joshua Colp. Changes --- Committed in revision 431717 Bugs: ASTERISK-24015 and ASTERISK-24703 https://issues.asterisk.org/jira/browse/ASTERISK-24015 https://issues.asterisk.org/jira/browse/ASTERISK-24703 Repository: Asterisk Description --- This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. - Preemptive question: why 'redirect', and not 'transfer'? Mostly because 'transfer' was always kind of a bad name. If the channel isn't answered, we aren't transferring, we're forwarding. If it is answered, the type of transfer being performed is somewhat vague - is it blind? Is it attended? 'redirect' - while also a slightly loaded term - is a bit more generic and yet descriptive of what is happening: we're redirecting the channel to somewhere else. Answered, not answered, it doesn't matter: your channel is no good here! *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Diffs - /branches/13/rest-api/api-docs/channels.json 430839 /branches/13/res/stasis/control.c 430839 /branches/13/res/res_pjsip_transport_websocket.c 430839 /branches/13/res/res_pjsip_nat.c 430839 /branches/13/res/res_pjsip_multihomed.c 430839 /branches/13/res/res_ari_channels.c 430839 /branches/13/res/ari/resource_channels.c 430839 /branches/13/res/ari/resource_channels.h 430839 /branches/13/include/asterisk/stasis_app.h 430839 /branches/13/channels/chan_pjsip.c 430839 Diff: https://reviewboard.asterisk.org/r/4316/diff/ Testing --- Tests were written both for the PJSIP stack as well as the new ARI operation. See https://reviewboard.asterisk.org/r/4352. Thanks, Matt Jordan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4414: res_pjsip_session: Fix double re-INVITE collision crash.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4414/#review14452 --- Ship it! Ship It! - Mark Michelson On Feb. 11, 2015, 10:37 p.m., rmudgett wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4414/ --- (Updated Feb. 11, 2015, 10:37 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24727 https://issues.asterisk.org/jira/browse/ASTERISK-24727 Repository: Asterisk Description --- A multi-asterisk box setup with direct media enabled would occasionally crash when two re-INVITE collisions on a call leg happen in a row. The re-INVITE logic only had one timer struct to defer the re-INVITE. When the second collision happens the timer struct is overwritten and put into the timer heap again. Resources for the first timer are leaked and the heap has two positions occupied by the same timer struct. Now the heap ordering is potentially corrupted, the timer will fire twice, and any resources allocated for the second timer will be released twice. * The solution is to put the collided re-INVITE into the delayed requests queue with all the other delayed requests and cherry pick the next request that can come off the queue when an event happens. * Changed to put delayed BYE requests at the head of the delayed queue. There is no sense in processing delayed UPDATEs and re-INVITEs when a BYE has been requested. * Made the start of a BYE request flush the delayed requests queue to prevent a delayed request from overlaping the BYE transaction. I saw a few cases were a delayed re-INVITE got started after the BYE transaction started. * Changed the delayed_request struct to use an enum instead of a string for the request method. Cherry picking the queue is easier with an enum than string comparisons and the compiler can warn if a switch statement does not cover all defined enum values. * Improved the debug output to give more information. It helps to know which channel is involved with an endpoint. Trunks can have many channels associated with the endpoint at the same time. Diffs - /branches/13/res/res_pjsip_session.c 431716 Diff: https://reviewboard.asterisk.org/r/4414/diff/ Testing --- Ran the full testsuite with the patch. All but one test passes and the failing test fails with or without the patch. Setup a test environment with two asterisk boxes and direct media enabled for all legs to perform this call repeatedly: PJSIP/100 -- box 1 -- box 2 -- box 1 -- PJSIP/200 1) 100 calls 200 2) 200 answers 3) 100 hangs up 4) repeat call A crash no longer happens on either Asterisk box with the patch. Thanks, rmudgett -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4415: Testsuite: Attended transfers from non-Stasis to a Stasis bridge
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4415/#review14453 --- Ship it! I had comments on the PJSUA phone controller code, but I believe none of the findings would have any effect on how this test would be written. So, once the PJSUA phone controller is committed, I say commit away on this as well. - Mark Michelson On Feb. 12, 2015, 6:34 a.m., jbigelow wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4415/ --- (Updated Feb. 12, 2015, 6:34 a.m.) Review request for Asterisk Developers and Mark Michelson. Bugs: ASTERISK-24578, ASTERISK-24755 and ASTERISK-24782 https://issues.asterisk.org/jira/browse/ASTERISK-24578 https://issues.asterisk.org/jira/browse/ASTERISK-24755 https://issues.asterisk.org/jira/browse/ASTERISK-24782 Repository: testsuite Description --- This adds the Attended Transfers from non-Stasis to a Stasis bridge tests 1.1 1.2 as described on the StasisStart/StasisEnd Test Plan at: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30279826 Notes: * These tests use the phones.py module that is up for review at https://reviewboard.asterisk.org/r/4413/ * The tests may occasionally fail due to bug ASTERISK-24755. * These tests constantly fail due to ASTERISK-24782. The YAML config 'expected-result: False' has been set for the tests. Once the bug has been resolved the config line should be removed from both. Diffs - /asterisk/trunk/tests/rest_api/external_interaction/attended_transfer/tests.yaml 6379 /asterisk/trunk/tests/rest_api/external_interaction/attended_transfer/non_stasis_bridge_to_stasis_bridge/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/attended_transfer/non_stasis_bridge_to_stasis_bridge/configs/ast1/pjsip.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/attended_transfer/non_stasis_bridge_to_stasis_bridge/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/attended_transfer/non_stasis_app_to_stasis_bridge/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/attended_transfer/non_stasis_app_to_stasis_bridge/configs/ast1/pjsip.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/attended_transfer/non_stasis_app_to_stasis_bridge/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4415/diff/ Testing --- * Because of bug ASTERISK-24782, a few ARI/AMI events aren't received and actions don't fire. Therefore those events/actions in the YAML haven't been tested. * Executed tests multiple times. * Reviewed logs to ensure the tests were executing as expected. Thanks, jbigelow -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
On Jan. 29, 2015, 10:23 p.m., Mark Michelson wrote: /branches/13/configs/examples/super_awesome_company/pjsip.conf, line 41 https://reviewboard.asterisk.org/r/4379/diff/1/?file=71114#file71114line41 I'm curious why you elected to use MAC addresses as the endpoint names. I'd personally find things a lot easier to configure/maintain if the SIP endpoint/aor/auth name is the same as the voicemail box number is the same as the extension number, etc. This also means that if Lindsey does some crazy extreme stunt that smashes her phone, then when she replaces it with a new one, you're going to have to change config values everywhere to have the new MAC address of the phone. Matt Jordan wrote: Hm. I think that's usually one of those best practices. You generally don't want the auth user to be easily guessed. Of course, we could split the concept of the endpoint name from the auth user, which would then allow the endpoints to be named 107 (for example) and the auth user to be her MAC address. Joshua Colp wrote: I think in practice this would just cause problems. Not all devices allow those two things to be separate. It's annoying. Mark Michelson wrote: SAC uses Digium phones, and Digium phones allow separate user and authuser to be specified. Joshua Colp wrote: Your statement is true but it would be nice if we could err on the side of not falling into a trap of doing fundamental stuff which isn't applicable to the wide world. rnewton wrote: I used MAC addresses as that is what we use as an example in our security best practices document: http://svnview.digium.com/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt?view=markup Perhaps this is a moot point. SAC's Asterisk system is behind NAT and firewall, so we could change the spec to specify that IT has locked down traffic between Asterisk and the public internet to only allow inbound traffic from the ITSP addresses. Or, on Asterisk we can use ACL's to limit traffic allowed to the internal network and ITSP addresses. With either of those approaches we should be able to use the less secure extension numbered auth users. What would be the issues either of these approaches other than an attacker on the internal network? Matt Jordan wrote: I think we need to come to some concurrence here so that the diffs can get updated. I suspect there are going to be additional rounds of review. The purpose of this set of example configs is to provide a base for a recommended deployment. Regardless of the scheme chosen, the example absolutely should use the best practices so that people have a secure system. If someone wants to use 'alice' and 'bob' for their names, that may be suitable for some examples, but not suitable for a recommended deployment. I don't care if we use MAC address or something else that is suitably difficult to guess, but MAC address is what A:TDG recommends [1] as well as our README-SERIOUSLY [2], and that feels like a decent starting point. [1] http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html#DeviceConfig_id291081 [2] http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt Looks like the best compromise is splitting out the user ID and auth name. Therefore the endpoint,aor,auth objects will be the extension number and the auth usernames will be the MAC address. We still have security, things are easier to administrate and the only downside is the config won't work as well for some phones. That being said, even some cheap generic phones I had laying around were able to configure a separate user id/account and auth name. This also makes other parts of the dialplan easier, hints and no crazy variable mapping for the endpoint names. - rnewton --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14377 --- On Jan. 27, 2015, 7:15 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Jan. 27, 2015, 7:15 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Feb. 13, 2015, 12:46 a.m.) Review request for Asterisk Developers. Changes --- Basically, review it again from head to toe. Re-tested, same testing as described in initial post. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs (updated) - /branches/13/configs/examples/awesome/voicemail.conf PRE-CREATION /branches/13/configs/examples/awesome/pjsip.conf PRE-CREATION /branches/13/configs/examples/awesome/musiconhold.conf PRE-CREATION /branches/13/configs/examples/awesome/modules.conf PRE-CREATION /branches/13/configs/examples/awesome/logger.conf PRE-CREATION /branches/13/configs/examples/awesome/indications.conf PRE-CREATION /branches/13/configs/examples/awesome/extensions.conf PRE-CREATION /branches/13/configs/examples/awesome/asterisk.conf PRE-CREATION /branches/13/configs/examples/awesome/README PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4379/diff/ Testing --- Setup Asterisk with configuration, connected up three phones using the first three users. Made calls between them all, left voicemails and retrieved them with all users. Verified MWI working with all phones. Thanks, rnewton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4413: Testsuite: Simulate phones and control from YAML.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4413/ --- (Updated Feb. 12, 2015, 6:27 p.m.) Review request for Asterisk Developers and Mark Michelson. Changes --- Addressed all but one issue which I'm looking for feedback on. Bugs: ASTERISK-24578 https://issues.asterisk.org/jira/browse/ASTERISK-24578 Repository: testsuite Description --- Pluggable modules to place, receive, and transfer (blind/attended) calls to simulate phones using PJSUA and YAML configuration. Calls are placed and/or transferred using the new pluggable action module. This should allow many currrent and future tests to easily send/receive calls to/from Asterisk along with transferring calls within YAML configuration. The pluggable test module (phones.PjsuaPhoneController) initializes the PJSUA accounts and once all have registered, the account callbacks are setup and are ready to receive calls. The pluggable action module (pluggable_modules.PjsuaPhoneActionModule) provides the ability to place calls and transfer calls using the accounts from YAML and the action is referenced with 'pjsua_phone'. The only time a call is hung up by this is when a transfer is performed and a 200 OK sipfrag NOTIFY is received. None of the modules set a pass/fail result and are only for driving and manipulating calls. See attached file for YAML demo. Diffs (updated) - /asterisk/trunk/lib/python/asterisk/pluggable_modules.py 6379 /asterisk/trunk/lib/python/asterisk/pjsua_mod.py 6379 /asterisk/trunk/lib/python/asterisk/phones.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4413/diff/ Testing --- * Tested placing calls, receiving calls, transfering via blind attended. * Pylint score of 9.40/10 for phones.py * See attached test-config.yaml for a demonstration. File Attachments Demonstration https://reviewboard.asterisk.org/media/uploaded/files/2015/02/11/659ab31f-8401-4f24-be5e-da1db0be3156__test-config.yaml Thanks, jbigelow -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/#review14456 --- /branches/13/configs/examples/awesome/extensions.conf https://reviewboard.asterisk.org/r/4379/#comment24995 Re-phrase this and remove trailing whitespace. - rnewton On Feb. 13, 2015, 12:46 a.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4379/ --- (Updated Feb. 13, 2015, 12:46 a.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- One of things discussed at the last AstriDevCon was better documentation (for everything!), but in particular, we mentioned needing some example configurations that pertain to a real-world scenario. That is, as opposed to the current sample files which are sort of all over the place at this point. This patch proposes a basic and minimal configuration of Asterisk to satisfy the requirements for the first phase of Super Awesome Company's implementation of Asterisk. I will submit four separate patches for the first phase, so that we don't have to review the entire thing all at once. This review is for the first patch. Who is Super Awesome Company? See https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company For the first patch, I am attempting to satisfy the below requirements. The patch does not include a new make target, as I believe Matt Jordan offered to handle that. SAC requires: * PJSIP connectivity for all employee desk phones. * The ability for employees to call one another inside of the office. * Voicemail boxes for each of the employees. Basic configuration We want SAC to have a clean system. That means: * No 'autoload' in modules.conf. Explicitly load a basic configuration. If SAC doesn't need the module, don't load it. * Every module loaded should have a configuration file that is appropriate for it. This includes all the 'core' things that need configuration. pjsip.conf * A PJSIP configuration for their desk phones. Assume every endpoint that is a phone has: * A voicemail mailbox that they can subscribe to * A hint for their device * Note that the PJSIP configuration should adhere to best practices. That means MAC addresses for device names, etc. extensions.conf * A safe dialplan for intra-company communication. This should be templated out so that it is trivial to add additional devices (use pattern matching/pattern matching hints, etc.) * Receiving a Busy/Unavailable should result in going to VoiceMail * A user should be able to dial something and get to their VoiceMailMain without having to enter in their extension number * Note that mapping of MAC address endpoints to extension numbers should be done in some fashion that is easily extensible. voicemail.conf * Set up mailboxes for every person in SAC. Assign 'default' pins. Create reasonable basic settings. * Do not set up e-mail or pager addresses. REVIEW? Please, if possible look at this from a few angles: * Use the configuration, configure a couple phones and call between them. Leave voicemails and retrieve them. * Have I created any security issues? * Is my dialplan easy to understand? * Could anything be done more efficiently without making it over-complicated? * Have I over-complicated anything? * Are there any critical settings I'm missing from any of the files? A couple, more specific questions: * We have sample configs in /configs/samples; what directory do we want these configurations in? (I used /configs/examples for now, but I don't really like it) * We have the make target make samples for the current samples; what do we want for these new configs? Diffs - /branches/13/configs/examples/awesome/voicemail.conf PRE-CREATION /branches/13/configs/examples/awesome/pjsip.conf PRE-CREATION /branches/13/configs/examples/awesome/musiconhold.conf PRE-CREATION /branches/13/configs/examples/awesome/modules.conf PRE-CREATION /branches/13/configs/examples/awesome/logger.conf PRE-CREATION /branches/13/configs/examples/awesome/indications.conf PRE-CREATION /branches/13/configs/examples/awesome/extensions.conf PRE-CREATION /branches/13/configs/examples/awesome/asterisk.conf PRE-CREATION /branches/13/configs/examples/awesome/README PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4379/diff/ Testing --- Setup Asterisk with configuration, connected up three phones using the first three users. Made calls between them all, left voicemails and retrieved them with
Re: [asterisk-dev] [Code Review] 4413: Testsuite: Simulate phones and control from YAML.
On Feb. 12, 2015, 12:42 p.m., Mark Michelson wrote: /asterisk/trunk/lib/python/asterisk/phones.py, lines 86-88 https://reviewboard.asterisk.org/r/4413/diff/1/?file=71348#file71348line86 I think this is a good case for using is instead of equality. I imagine that callers of this function are going to be passing a reference to the same account object that is stored on the phone_obj rather than a separate object with equivalent properties. Also, to avoid the continue statement, you can just use a positive comparison instead of a negative one: if account is phone_obj.account: return phone_obj Yup, that's what I meant to do :) On Feb. 12, 2015, 12:42 p.m., Mark Michelson wrote: /asterisk/trunk/lib/python/asterisk/phones.py, lines 115-117 https://reviewboard.asterisk.org/r/4413/diff/1/?file=71348#file71348line115 A couple of PEP 8 points: 1) When catching exceptions, use the syntax except ErrorType as ErrorInstance instead of except ErrorType, ErrorInstance. So here, it would be except pj.Error as err 2) A PEP 8 checker will complain about the indentation of str(err)) on the final line I've highlighted. It will claim it should be lined up like this: raise Exception(Exception occurred while making call: '%s' % str(err)) See how str is lined up to show how it belongs to the parentheses? I blame my python plugin for vim which just shifts twice. I installed the pep8 utility which pointed out a couple others which I have also fixed. On Feb. 12, 2015, 12:42 p.m., Mark Michelson wrote: /asterisk/trunk/lib/python/asterisk/phones.py, line 156 https://reviewboard.asterisk.org/r/4413/diff/1/?file=71348#file71348line156 self.current_call doesn't seem to be used for anything. Removed. On Feb. 12, 2015, 12:42 p.m., Mark Michelson wrote: /asterisk/trunk/lib/python/asterisk/phones.py, lines 190-191 https://reviewboard.asterisk.org/r/4413/diff/1/?file=71348#file71348line190 Should this only be done if self.phone is None? Seems like this would be redundant after the initial setting. That would make sense. On Feb. 12, 2015, 12:42 p.m., Mark Michelson wrote: /asterisk/trunk/lib/python/asterisk/phones.py, line 192 https://reviewboard.asterisk.org/r/4413/diff/1/?file=71348#file71348line192 I suspect it's not an issue, but is there ever a chance that self.call could be None when the on_state() callback is called? As far as I know self.call would never be None when a pjsua callback is called such as on_state(). On Feb. 12, 2015, 12:42 p.m., Mark Michelson wrote: /asterisk/trunk/lib/python/asterisk/phones.py, lines 197-204 https://reviewboard.asterisk.org/r/4413/diff/1/?file=71348#file71348line197 This seems overly complicated for what you are trying to do here. Why not just try: self.phone.calls.remove(self.call) except ValueError: # Log an error or something If that doesn't work because equality of calls doesn't evaluate as you'd expect, then this is a situation where I expect that a single object reference is being passed around everywhere rather than separate call objects. You could take the approach of reconstructing the self.phone.calls list like this: self.phone.calls = [call for call in self.phone.calls if call is not self.call] The first suggestion is actually what I initially was using. However I found that the call object wasn't always being removed from the list. Debugging showed the id() of self.call not matching any in the list which I believe is because self.call is set to weakref.proxy(call) in pjsua.CallCallback(). Comparing the SIP Call-ID of the call objects should be unique enough to find the one to remove from the list. On Feb. 12, 2015, 12:42 p.m., Mark Michelson wrote: /asterisk/trunk/lib/python/asterisk/phones.py, line 246 https://reviewboard.asterisk.org/r/4413/diff/1/?file=71348#file71348line246 This function's name is misleading because it does not actually log anything. It should either 1) Be renamed to something indicating that it is just gathering call information into a string or 2) Be altered to actually log the call info using LOGGER.debug() Renamed. On Feb. 12, 2015, 12:42 p.m., Mark Michelson wrote: /asterisk/trunk/lib/python/asterisk/phones.py, line 270 https://reviewboard.asterisk.org/r/4413/diff/1/?file=71348#file71348line270 Can you add a comment about why the lock is being used here? Fixed this by removing the locking :) Apparently a lock isn't needed when making call. Unless perhaps a callback isn't passed and then set later using call.set_callback() but that's not the case here. On Feb. 12, 2015, 12:42 p.m., Mark Michelson wrote: /asterisk/trunk/lib/python/asterisk/phones.py, lines 272-275
Re: [asterisk-dev] Advanced feature query
On 12 Feb 2015, at 09:55, David Radcliffe david.radcli...@clockworkit.co.uk wrote: Hi, Does anyone know if Asterix has the ability to send a network message (TCP/IP packet) indicating that a call is in progress through the PBX? This is not really a question about asterisk development - so you are on the wrong mailing list. This mailing list is for developers of the asterisk source code. I need a message to say “pbx has at least one call in progress”, and another to say “pbx has no calls in progress” when it is idle. Please check the asterisk management interface where you get events like this. Regards /Olle If it can do this, can someone please let me know how to configure Asterix to do this. Thanks David W. Radcliffe Senior Developer image001.jpg Clockwork IT Ltd tel: 08448 040950 mob: 07764 155251 image002.jpg WWW.SUPPORTDESKPRO.CO.UK image003.png image004.pngimage005.jpg image006.gif Clockwork IT Ltd, Saturn Business Centre, 101 Lockhurst Lane, Coventry, CV6 5SF NOTICE-This message contains information intended only for the use of the addressee named above. It may also be confidential and/or privileged. If you are not the addressee, you should not disclose, copy, distribute, take any action or rely on this E-mail and should notify us immediately by return E-mail to i...@clockworkit.co.uk The views expressed in this communication may not necessarily be the views held by Clockwork IT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Advanced feature query
Hi, Does anyone know if Asterix has the ability to send a network message (TCP/IP packet) indicating that a call is in progress through the PBX? I need a message to say pbx has at least one call in progress, and another to say pbx has no calls in progress when it is idle. If it can do this, can someone please let me know how to configure Asterix to do this. Thanks David W. Radcliffe Senior Developer Clockwork IT Ltd tel: 08448 040950 mob: 07764 155251 http://www.supportdeskpro.co.uk/ WWW.SUPPORTDESKPRO.CO.UK http://www.linkedin.com/company/clockwork-it-ltd?trk=fc_badge%22%3e%3cimg%2 0src=%22http://static01.linkedin.com/scds/common/u/img/webpromo/btn_cofollow _badge.png%22%20locale=%22en%22%20alt=%22Clockwork%20IT%20Ltd%20on%20LinkedI n%22%3e%3c/ahttp://twitter.com/ClockworkIT http://facebook.com/pages/Clockwork-It/104091779694342 callto://david.radcliffe/ Clockwork IT Ltd, Saturn Business Centre, 101 Lockhurst Lane, Coventry, CV6 5SF NOTICE-This message contains information intended only for the use of the addressee named above. It may also be confidential and/or privileged. If you are not the addressee, you should not disclose, copy, distribute, take any action or rely on this E-mail and should notify us immediately by return E-mail to mailto:i...@clockworkit.co.uk i...@clockworkit.co.uk The views expressed in this communication may not necessarily be the views held by Clockwork IT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Reject incoming call
Hi,
Thanks for the response.
no one going to answer this call. We treat it as miss call.
we catch the CLI, call AGI to execute .sjp and hangup.
Regrads,
Roy
On Thu, Feb 12, 2015 at 5:14 PM, bala murugan fightwit...@gmail.com wrote:
Who is responsible to answer ??
On Feb 12, 2015 6:05 AM, Raj Roy Ghandhi roy.gan...@gmail.com wrote:
Hi Friends,
I am trying to implement a simple dial plan with asterisk.
1. Ring the inbound call
2. wait for 2 seconds
3. call agi script with cli
4. hangup
But when it gets hangup I see the Declined is passed from the asterisk.
But my PSTN provider keep waiting in dialing state with no noise until 60
seconds.
any idea what I have done wrong.
here is my dial plan
exten = 01,1,Ringing()
exten = 01,2,Wait(2)
exten =
01,3,Set(vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=${CALLERID(num)})
exten = 01,4,Agi(agi://127.0.0.1/url=${vxmlurl}
http://127.0.0.1/url=$%7Bvxmlurl%7D)
exten = 01,5,Hangup()
Please advice,
Best Regards,
Roy.
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Re: [asterisk-dev] Reject incoming call
Hi Pavel,
Thanks for the reply,
I tried with Hangup(16) but same result
what I get in console is
-- Executing [01XXX@public:1] Ringing(SIP/10.2.2.75-0004, ) in
new stack
-- Executing [01XXX@public:2] Wait(SIP/10.2.2.75-0004, 1)
in new stack
-- Executing [01XXX@public:3] Set(SIP/10.2.2.75-0004,
vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XX) in
new stack
-- Executing [01XXX@public:4] AGI(SIP/10.2.2.75-0004, agi://
127.0.0.1/url=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XX)
in new stack
-- SIP/10.2.2.75-0004AGI Script agi://
127.0.0.1/url=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XX
completed,
returning 4
[Feb 13 10:25:24] ERROR[10111][C-0004]: utils.c:1321 ast_carefulwrite:
write() returned error: Broken pipe
--- Reliably Transmitting (no NAT) to 11.200.1.53:9131 ---
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 11.200.1.53:9131
;branch=z9hG4bKkcw3bikyw336f07x6xt6wbbc6;received=10.200.1.53
From: sip:07XX@11.2.2.75;user=phone;tag=sbc0403xbxkbx6w-CC-22
To: sip:01XXX@11.2.2.75;user=phone;tag=as02deb6c4
Call-ID: isbcyc60z60kyw17fk6e1y36i7ey7tz33y3i@SoftX3000
CSeq: 1 INVITE
Server: Asterisk PBX 11.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
--- SIP read from UDP:11.200.1.53:9131 ---
ACK sip:01XXX@11.2.2.75:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 11.200.1.53:9131
;branch=z9hG4bKkcw3bikyw336f07x6xt6wbbc6;received=11.200.1.53
Call-ID: isbcyc60z60kyw17fk6e1y36i7ey7tz33y3i@SoftX3000
From: sip:07XX@11.2.2.75;user=phone;tag=sbc0403xbxkbx6w-CC-22
To: sip:01XXX@11.2.2.75;user=phone;tag=as02deb6c4
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
I am running Asterisk 11.8.0 with voiceglue 0.14
When I try the .jsp page with wget, it works fine without any error.
Regards,
Roy.
On Thu, Feb 12, 2015 at 4:58 PM, Pavel Troller pat...@sinus.cz wrote:
Hi Roy,
Hi Friends,
I am trying to implement a simple dial plan with asterisk.
1. Ring the inbound call
2. wait for 2 seconds
3. call agi script with cli
4. hangup
But when it gets hangup I see the Declined is passed from the asterisk.
But my PSTN provider keep waiting in dialing state with no noise until 60
seconds.
any idea what I have done wrong.
here is my dial plan
exten = 01,1,Ringing()
exten = 01,2,Wait(2)
exten =
01,3,Set(vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=${CALLERID(num)})
exten = 01,4,Agi(agi://127.0.0.1/url=${vxmlurl})
exten = 01,5,Hangup()
Did you try to put a specific Clear Cause code to the Hangup command ?
For example, Hangup(17) means User Busy, i.e. 486 Busy Here on SIP, or
Hangup(1) means Unallocated Number or 404 Not Found etc. If you want to
find more codes, search for ISDN Cause Codes. Your PSTN provider will
probably react better to other causes than the default one, which is
probably 16 Normal Call Clearing.
With regards,
Pavel
Please advice,
Best Regards,
Roy.
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Re: [asterisk-dev] [Code Review] 4413: Testsuite: Simulate phones and control from YAML.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4413/ --- (Updated Feb. 12, 2015, 10:25 a.m.) Review request for Asterisk Developers and Mark Michelson. Bugs: ASTERISK-24578 https://issues.asterisk.org/jira/browse/ASTERISK-24578 Repository: testsuite Description (updated) --- Pluggable modules to place, receive, and transfer (blind/attended) calls to simulate phones using PJSUA and YAML configuration. Calls are placed and/or transferred using the new pluggable action module. This should allow many currrent and future tests to easily send/receive calls to/from Asterisk along with transferring calls within YAML configuration. The pluggable test module (phones.PjsuaPhoneController) initializes the PJSUA accounts and once all have registered, the account callbacks are setup and are ready to receive calls. The pluggable action module (pluggable_modules.PjsuaPhoneActionModule) provides the ability to place calls and transfer calls using the accounts from YAML and the action is referenced with 'pjsua_phone'. The only time a call is hung up by this is when a transfer is performed and a 200 OK sipfrag NOTIFY is received. None of the modules set a pass/fail result and are only for driving and manipulating calls. See attached file for YAML demo. Diffs - /asterisk/trunk/lib/python/asterisk/pluggable_modules.py 6379 /asterisk/trunk/lib/python/asterisk/pjsua_mod.py 6379 /asterisk/trunk/lib/python/asterisk/phones.py PRE-CREATION Diff: https://reviewboard.asterisk.org/r/4413/diff/ Testing --- * Tested placing calls, receiving calls, transfering via blind attended. * Pylint score of 9.40/10 for phones.py * See attached test-config.yaml for a demonstration. File Attachments Demonstration https://reviewboard.asterisk.org/media/uploaded/files/2015/02/11/659ab31f-8401-4f24-be5e-da1db0be3156__test-config.yaml Thanks, jbigelow -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] DNS Support in Asterisk
Looking at this, I'm inclined to say that libunbound is the better of the two options in spite of it being somewhat more difficult to consume DNS records than it would be with c-ares. In my estimation a (seemingly?) more-active community and the inclusion of a cache are more important. DNSSEC isn't a huge deal, at least not for me at this time, but is a nice bonus as well. - Brad On Thu, Feb 12, 2015 at 10:01 AM, Joshua Colp jc...@digium.com wrote: Greetings all, I've extended the sections of my wiki page for c-ares[1] and libunbound[2] to include further information about documentation, general usage experience, and other aspects. Personally I lean towards libunbound because it was straight forward to experiment with, supports DNSSEC, and has a cache. Cheers, [1] https://wiki.asterisk.org/wiki/display/~jcolp/DNS+Support+in+Asterisk#DNSSupportinAsterisk-c-ares [2] https://wiki.asterisk.org/wiki/display/~jcolp/DNS+Support+in+Asterisk#DNSSupportinAsterisk-libunbound -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] DNS Support in Asterisk
Greetings all, I've extended the sections of my wiki page for c-ares[1] and libunbound[2] to include further information about documentation, general usage experience, and other aspects. Personally I lean towards libunbound because it was straight forward to experiment with, supports DNSSEC, and has a cache. Cheers, [1] https://wiki.asterisk.org/wiki/display/~jcolp/DNS+Support+in+Asterisk#DNSSupportinAsterisk-c-ares [2] https://wiki.asterisk.org/wiki/display/~jcolp/DNS+Support+in+Asterisk#DNSSupportinAsterisk-libunbound -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] DNS Support in Asterisk
There is one version of c-ares in resiprocate as well. C-ares has been in use for a long time and is in use every single day for you as part of most curl installs. I am not sure there is much to do there. Libunbound adds a lot if that is what we want. Why is a cache a good thing? You surely have a caching resolver running on your system, right? DNSsec is a huge deal - and the foundation for a lot of security things coming up. Someone wrote an IETF draft about that and SIP. https://tools.ietf.org/html/draft-johansson-sipcore-dane-sip-00 I got a patch sent to me that implements that in Asterisk with unbound, but haven't gotten time to go through it and test it. /O On 12 Feb 2015, at 16:25, Brad Watkins marqui...@gmail.com wrote: Looking at this, I'm inclined to say that libunbound is the better of the two options in spite of it being somewhat more difficult to consume DNS records than it would be with c-ares. In my estimation a (seemingly?) more-active community and the inclusion of a cache are more important. DNSSEC isn't a huge deal, at least not for me at this time, but is a nice bonus as well. - Brad On Thu, Feb 12, 2015 at 10:01 AM, Joshua Colp jc...@digium.com wrote: Greetings all, I've extended the sections of my wiki page for c-ares[1] and libunbound[2] to include further information about documentation, general usage experience, and other aspects. Personally I lean towards libunbound because it was straight forward to experiment with, supports DNSSEC, and has a cache. Cheers, [1] https://wiki.asterisk.org/wiki/display/~jcolp/DNS+Support+in+Asterisk#DNSSupportinAsterisk-c-ares [2] https://wiki.asterisk.org/wiki/display/~jcolp/DNS+Support+in+Asterisk#DNSSupportinAsterisk-libunbound -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Reject incoming call
On Thu, Feb 12, 2015 at 5:44 AM, bala murugan fightwit...@gmail.com wrote:
Who is responsible to answer ??
On Feb 12, 2015 6:05 AM, Raj Roy Ghandhi roy.gan...@gmail.com wrote:
Hi Friends,
I am trying to implement a simple dial plan with asterisk.
1. Ring the inbound call
2. wait for 2 seconds
3. call agi script with cli
4. hangup
But when it gets hangup I see the Declined is passed from the asterisk.
But my PSTN provider keep waiting in dialing state with no noise until 60
seconds.
any idea what I have done wrong.
here is my dial plan
exten = 01,1,Ringing()
exten = 01,2,Wait(2)
exten =
01,3,Set(vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=${CALLERID(num)})
exten = 01,4,Agi(agi://127.0.0.1/url=${vxmlurl}
http://127.0.0.1/url=$%7Bvxmlurl%7D)
exten = 01,5,Hangup()
Please advice,
Best Regards,
Roy.
This question belongs on asterisk-users [1]. You're far more likely to get
help there than on the asterisk-dev mailing list, which is used to discuss
the actual source code and development of Asterisk.
[1] http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-dev] [Code Review] 4408: Testsuite: Add external bridging tests for Stasis (two channel) interactions
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4408/ --- (Updated Feb. 12, 2015, 10:48 a.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers and Mark Michelson. Changes --- Committed in revision 6389 Bugs: ASTERISK-24611 https://issues.asterisk.org/jira/browse/ASTERISK-24611 Repository: testsuite Description --- This adds external bridging tests for Stasis (two channel) interactions as defined on the StasisStart/StasisEnd Test Plan (tests 2.5, 2.6, 2.7, and 2.8) at: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30279826#StasisStart/StasisEndTestplan-ExternalBridging This also renames (move to sub directory) the test 'tests/rest_api/external_interaction/ami_bridge/stasis_app/' to 'tests/rest_api/external_interaction/ami_bridge/stasis_app/non_stasis_app/'. NOTE: The files for the renamed test don't appear just because of how things work. Diffs - /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/tests.yaml 6377 /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_bridge/two_channel_same_stasis_app/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_bridge/two_channel_same_stasis_app/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_bridge/two_channel_different_stasis_app/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_bridge/two_channel_different_stasis_app/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_bridge/tests.yaml 6377 /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/two_channel_same_stasis_app/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/two_channel_same_stasis_app/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/two_channel_different_stasis_app/test-config.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/two_channel_different_stasis_app/configs/ast1/extensions.conf PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/tests.yaml PRE-CREATION /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/test-config.yaml 6377 /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_app/configs/ast1/extensions.conf 6377 Diff: https://reviewboard.asterisk.org/r/4408/diff/ Testing --- * Executed each test in a loop of 100 iterations with no failures. * Reviewed logs to ensure the tests were executing as expected. Thanks, jbigelow -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
