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serov.d.p at gmail.com Guest
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Posted: Wed Mar 11, 2015 2:27 am Post subject: [asterisk-users] PJSIP some AMI events is absent? |
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Hello.
Asterisk 13.2, PJSIP.
Problem: I do not get any AMI events when changing the status of the
contact.
When using chan_sip I got "peerstatus" event.
When using res_pjsip and devices (endpoint configuration) I got
"peerstatus" event.
When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND
AUTHENTICATION i got "registry" event.
When using "ENDPOINT CONFIGURED AS A TRUNK, OUTBOUND AUTHENTICATION" and
status on contact changed I do not get any AMI event.
I missed something?
Tell me how to determine the change in the status of the contact (or
endpoint/trunk) through AMI?
Simple config:
[srv_dev]
type=auth
auth_type=userpass
username=login
password=secret
[srv_dev]
type=aor
contact=sip:sip.example.com:5060
qualify_frequency=5
default_expiration=10
max_contacts=1
remove_existing=yes
[srv_dev]
type=endpoint
from_domain=example.com
aors=srv_dev
outbound_auth=srv_dev
rewrite_contact=yes
allow=!all,alaw
Dmitriy Serov
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jd.girard at sysnux.pf Guest
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Posted: Wed Mar 11, 2015 7:11 pm Post subject: [asterisk-users] PJSIP some AMI events is absent? |
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
I made some tests with asterisk-13.2.0 and chan_pjsip this weekend
myself, and came to the same conclusion: some peerstatus events are
missing (eg. when contacts become unreachable / unavailable, IIRC), and
I could not find a way to get contacts status through AMI.
It looks a bit similar to issues 23172, 23173: PJSip missing
functionalities.
Thanks,
- --
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
Le 10/03/2015 21:27, Dmitriy Serov a écrit :
Quote: | Hello.
Asterisk 13.2, PJSIP.
Problem: I do not get any AMI events when changing the status of the
contact.
When using chan_sip I got "peerstatus" event.
When using res_pjsip and devices (endpoint configuration) I got
"peerstatus" event.
When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND
AUTHENTICATION i got "registry" event.
When using "ENDPOINT CONFIGURED AS A TRUNK, OUTBOUND AUTHENTICATION" and
status on contact changed I do not get any AMI event.
I missed something?
Tell me how to determine the change in the status of the contact (or
endpoint/trunk) through AMI?
Simple config:
[srv_dev]
type=auth
auth_type=userpass
username=login
password=secret
[srv_dev]
type=aor
contact=sip:sip.example.com:5060
qualify_frequency=5
default_expiration=10
max_contacts=1
remove_existing=yes
[srv_dev]
type=endpoint
from_domain=example.com
aors=srv_dev
outbound_auth=srv_dev
rewrite_contact=yes
allow=!all,alaw
Dmitriy Serov
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=Db9Z
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--
_____________________________________________________________________
-- 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:
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mjordan at digium.com Guest
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Posted: Thu Mar 12, 2015 9:43 am Post subject: [asterisk-users] PJSIP some AMI events is absent? |
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On Wed, Mar 11, 2015 at 7:11 PM, Jean-Denis Girard <jd.girard@sysnux.pf> wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
I made some tests with asterisk-13.2.0 and chan_pjsip this weekend
myself, and came to the same conclusion: some peerstatus events are
missing (eg. when contacts become unreachable / unavailable, IIRC), and
I could not find a way to get contacts status through AMI.
It looks a bit similar to issues 23172, 23173: PJSip missing
functionalities.
|
Yup, it was an oversight in the implementation of 'qualify' for PJSIP endpoints.
Dmitriy was kind enough to open an issue for it:
https://issues.asterisk.org/jira/browse/ASTERISK-24863
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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