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Summary:ASTERISK-24488: Wrong remote identity and target in dialog package XML in NOTIFY
Reporter:Alejandro Padilla (drazyck)Labels:pjsip
Date Opened:2014-11-04 05:07:50.000-0600Date Closed:2018-03-05 08:40:39.000-0600
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:Channels/chan_sip/General Core/General
Versions:SVN 13.1.0 Frequency of
Occurrence		Constant
Related
Issues:
is duplicated byASTERISK-26454 CallerID for BLF Not Displaying Number or Name
Environment:Attachments:( 0) issue_24488_full_log
( 1) Ngrep_Trace-notifycid_yes.txt
( 2) sip_casa.conf
( 3) sip.conf
Description:[Edit by Rusty - This issue is seen with devices that subscribe to the RFC4235 Dialog event package]

Hi, i had installed the new version asterisk 13.0.0 , and i found a issue on xml-info send to the phones on extensions monitoring.

In this example the extension 1080 are calling  to extension  1002

im monitoring the 1002 extension with blf and snom phone with other extension, and now i received this xml:
{noformat}
?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="9" state="full" entity="sip:1002@pbx.casa.local">
<dialog id="1002" call-id="pickup-080f0000be77-dq09hphbjmcr" local-tag="pndu3ilpwj" remote-tag="as03ca48d8" direction="recipient">
<remote>
<identity display="1002">sip:1002@pbx.casa.local</identity>
<target uri="sip:1002@pbx.casa.local"/>
</remote>
<local>
<identity display="1002">sip:1002@pbx.casa.local</identity>
<target uri="sip:1002@pbx.casa.local"/>
</local>
<state>early</state>
</dialog>
</dialog-info>
{noformat}
In before version of asterisk and same configuration im receiving this:
{noformat}
?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="13" state="full" entity="sip:1@pbx.casa.local">
<dialog id="1" call-id="pickup-0d100000a26a-r4fxlunt57hh" local-tag="we84z74osq" remote-tag="as78081848" direction="recipient">
<remote>
<identity display="tel760">sip:1080@pbx.casa.local</identity>
<target uri="sip:1080@pbx.casa.local"/>
</remote>
<local>
<identity display="tel870">sip:1002@pbx.casa.local</identity>
<target uri="sip:1002@pbx.casa.local"/>
</local>
<state>early</state>
</dialog>
</dialog-info>
{noformat}
I have this on sip.conf on both servers asterisk 11 and asterisk 13:
{noformat}
notifycid=ignore-context
trustrpid=no
sendrpid=yes
{noformat}

Regards
Comments:By: Matt Jordan (mjordan) 2014-11-04 08:13:16.413-0600

We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Please make sure 'sip set debug on' is enabled as well. Please also attach your sip.conf.
By: Alejandro Padilla (drazyck) 2014-11-04 11:04:59.004-0600

Ok, i wil send the debug as soon posible.

Thanks
By: Alejandro Padilla (drazyck) 2014-11-04 11:35:26.815-0600

Hi I attach the file debug and sip.conf.

Thanks
By: Alejandro Padilla (drazyck) 2014-11-11 05:49:53.686-0600

HI, any update of this?
By: Matt Jordan (mjordan) 2014-11-11 07:22:00.384-0600

You have to his "Send Back" after entering feedback, otherwise the issue doesn't end up back on the Triage queue.

There is no update at this time, since we just saw that you posted the debug information.
By: Alejandro Padilla (drazyck) 2014-11-11 09:28:28.894-0600

HI, i attach files before.. but i think that i doesn't the correct way...

I attach the files again

By: Alejandro Padilla (drazyck) 2014-11-11 09:30:15.108-0600

there are the debug files and sip.conf
By: Rusty Newton (rnewton) 2014-11-17 14:19:12.301-0600

Alejandro can you attach the relevant dialplan as well? i.e. hints, etc.
By: Alejandro Padilla (drazyck) 2014-11-18 04:28:33.334-0600

Hi, mi hints.conf
{noformat}
[hints]

exten => 1054,hint,SIP/1054
exten => 1001,hint,SIP/1001
exten => 1000,hint,SIP/1000
exten => 1005,hint,SIP/1005
exten => 1002,hint,SIP/1002
exten => 1006,hint,SIP/1006
;exten => snom710,hint,SIP/1005
exten => 1080,hint,SIP/1080
exten => 1983,hint,SIP/1983
exten => 1003,hint,SIP/1003
exten => 1005,hint,SIP/1005
exten => line,hint,SIP/line


;exten => 4000,hint,Custom:4000

exten => b1054,hint,Custom:b1054

;exten => 1000,hint,Custom:01005&Custom:01054

exten => b1005,hint,Custom:b1005
exten => b1006,hint,Custom:b1006
exten => b1080,hint,Custom:b1080
{noformat}

;;;;;And macro of internal calls:
{noformat}
[macro-llamada_extensiones]

exten => s,1,GotoIf(${DB_EXISTS(DND/${ARG1})}?DND-ON)
exten => s,n,SipAddHeader(P-Asserted-Identity: <sip:${CALLERID(NUM)}>)
exten => s,n,NoOp(Se incrementa el contador)
exten => s,n,set(contador=$[${contador}+1])

exten => s,n,NoOP(${DEVICE_STATE(SIP/${ARG1})})
exten => s,n,GotoIf(${DB_EXISTS(buzon/${ARG1})}?unavail)


exten => s,n(llamar),Dial(SIP/${ARG1},30,trXx) ; DND is OFF, start dialing
exten => s,n,Hangup()
{noformat}

By: Marco Paland (mpaland) 2014-12-09 09:05:14.554-0600

Perhaps this needs an own topic/issue:

I'm having a similar MAJOR issue, which makes 13.0.0 unusable with snom phones.
We are using the PJSIP stack, not the chan_sip.

The NOTIFY header of a ringing extension (1005) looks like (at ext 1010):

{code}
<--- Transmitting SIP request (808 bytes) to UDP:192.168.1.155:36578 --->
NOTIFY sip:1010@192.168.1.155:36578 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;rport;branch=z9hG4bKPjbaab25f1-9a22-4e9d-95f0-0311eba3d937
From: <sip:1005@192.168.1.1>;tag=94d2b042-dfe5-4d56-bc1b-77b212dd4d0f
To: <sip:1010@192.168.1.1>;tag=54wyd8hmq4
Contact: <sip:192.168.1.1:5060>
Call-ID: 313431373934373935333433383134-n1tmcldgl41x
CSeq: 25798 NOTIFY
Event: dialog
Subscription-State: active;expires=3500
Allow-Events: message-summary, presence, dialog, refer
Max-Forwards: 70
User-Agent: Asterisk
Content-Type: application/dialog-info+xml
Content-Length:   228

<?xml version="1.0" encoding="UTF-8"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" state="full" entity="sip:1005@200.200.200.200:5060">
<dialog id="1005">
 <state>early</state>
</dialog>
</dialog-info>
{code}

Issues:
- <remote> and <local> tags are missing completely
- No (IMPORTANT) direction="recipient" attr in <dialog> tag
- entity="sip:1005@200.200.200.200:5060" is outside IP, but inside IP (192.168.1.1) is needed

By: Rusty Newton (rnewton) 2014-12-09 09:47:09.350-0600

[~mpaland] please file a new issue since it uses a different channel driver. There could be a relation. I'll link the two issues once you file it.
By: Rusty Newton (rnewton) 2014-12-24 15:39:44.069-0600

I reproduced this issue with a Snom300 (snom300-SIP 8.7.3.25) and SVN-branch-13-r429983.

I had not setup extension monitoring/presence/BLF with Snom phones before, so here is a quick guide for whoever works this issue.

* Setup your Snom phone to register to Asterisk like usual.
* Go to the Setup> Function Keys page of the Snom web GUI.
* Pick a line to use. I used P2(L2).
* In the first column, on the left, choose the identity you configured.
* In the second column choose BLF.
* In the third column enter the identifier for your hint configured in Asterisk. I used "6002" as my hint was setup as:
{noformat}
exten => 6002,hint,SIP/BOB
{noformat}
* Apply and save your configuration. The phone should immediately SUBSCRIBE to Asterisk for the *dialog* event package.

Upon dialing the phone being monitored by the Snom, you should see a NOTIFY get sent to the Snom:

{noformat}
   -- Executing [6002@from-internal:1] Dial("SIP/ALICE-00000002", "SIP/BOB,30,Tt") in new stack
 == Using SIP RTP CoS mark 5
   -- Called SIP/BOB
set_destination: Parsing <sip:CHARLIE@10.24.18.21:2048;line=5tonr3op> for address/port to send to
set_destination: set destination to 10.24.18.21:2048
Reliably Transmitting (no NAT) to 10.24.18.21:2048:
NOTIFY sip:CHARLIE@10.24.18.21:2048;line=5tonr3op SIP/2.0
Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK18c214f6;rport
Max-Forwards: 70
From: <sip:6002@10.24.18.124;user=phone>;tag=as5911d088
To: <sip:CHARLIE@10.24.18.124>;tag=nabty9bj3g
Contact: <sip:6002@10.24.18.124:5060>
Call-ID: 549b2e0b787e-u3khy899l7uz
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX SVN-branch-13-r429983
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 540

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="4" state="full" entity="sip:6002@10.24.18.124">
<dialog id="6002" call-id="pickup-549b2e0b787e-u3khy899l7uz" local-tag="nabty9bj3g" remote-tag="as5911d088" direction="recipient">
<remote>
<identity display="6002">sip:6002@10.24.18.124</identity>
<target uri="sip:6002@10.24.18.124"/>
</remote>
<local>
<identity display="6002">sip:6002@10.24.18.124</identity>
<target uri="sip:6002@10.24.18.124"/>
</local>
<state>early</state>
</dialog>
</dialog-info>
{noformat}



By: Alejandro Padilla (drazyck) 2014-12-26 03:09:49.811-0600

Hi, thanks for the support.
for many years I am configuring monitoring extension with snom . This problem is not new , it has happened to me on several versions of Asterisk , I mean the xml property dialog info on snom phones, introduced from Asterisk 1.8. This has not worked properly until Asterisk 1.8.22 , I remember ..
and has returned to do bad in Asterisk 13.0.0 , do you have tested on this version? I have configured exactly as you say, but as you see in the logs that I commanded asterisk information is incorrect . Is it fixed in version 13.1 asterisk ?

Many thanks
By: Alejandro Padilla (drazyck) 2015-02-04 01:53:25.787-0600

Hi, any update of this? is solved?
By: Alejandro Padilla (drazyck) 2015-08-03 04:53:01.768-0500

What happend with this issue? are solved now? there are any workaround?
By: Tanguy CHAPRON (thera) 2015-12-07 02:33:28.952-0600

There is still the issue in 13.6.0.
By: Frank Mehrtens (fme) 2016-02-10 15:29:33.022-0600

Not fixed in 13.7.2. Is there a workaround or patch?
By: Alejandro Padilla (drazyck) 2016-02-11 03:19:38.703-0600

1 year open and not solved... I think that wont be solved never...
By: Abderrahim Laab (Abdo) 2016-07-04 17:24:56.349-0500

This bug cannot be solved?
By: LTCtech (LTCtech) 2016-10-31 19:08:54.281-0500

This is still an issue in 13.12.1. It seems that it can't find the ringing channel, callee is null, and the fields never get populated. The bug is probably somewhere else entirely thus tough to find. I did a diff against 11.6cert15 and neither find_ringer_channel nor state_notify_build_xml were modified.

{code}
callee = find_ringing_channel(data->device_state_info, p);
if (callee) {
static char *anonymous = "anonymous";
static char *invalid = "anonymous.invalid";
char *cid_num;
char *connected_num;
int need;
int cid_num_restricted, connected_num_restricted;

ast_channel_lock(callee);

cid_num_restricted = (ast_channel_caller(callee)->id.number.presentation &
  AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED;
cid_num = S_COR(ast_channel_caller(callee)->id.number.valid,
S_COR(cid_num_restricted, anonymous,
     ast_channel_caller(callee)->id.number.str), "");

need = strlen(cid_num) + (cid_num_restricted ? strlen(invalid) :
 strlen(p->fromdomain)) + sizeof("sip:@");
local_target = ast_alloca(need);

snprintf(local_target, need, "sip:%s@%s", cid_num,
cid_num_restricted ? invalid : p->fromdomain);

ast_xml_escape(S_COR(ast_channel_caller(callee)->id.name.valid,
    S_COR((ast_channel_caller(callee)->id.name.presentation &
    AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED, anonymous,
  ast_channel_caller(callee)->id.name.str), ""),
      local_display, sizeof(local_display));

connected_num_restricted = (ast_channel_connected(callee)->id.number.presentation &
   AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED;
connected_num = S_COR(ast_channel_connected(callee)->id.number.valid,
     S_COR(connected_num_restricted, anonymous,
   ast_channel_connected(callee)->id.number.str), "");

need = strlen(connected_num) + (connected_num_restricted ? strlen(invalid) :
strlen(p->fromdomain)) + sizeof("sip:@");
remote_target = ast_alloca(need);

snprintf(remote_target, need, "sip:%s@%s", connected_num,
connected_num_restricted ? invalid : p->fromdomain);

ast_xml_escape(S_COR(ast_channel_connected(callee)->id.name.valid,
    S_COR((ast_channel_connected(callee)->id.name.presentation &
    AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED, anonymous,
   ast_channel_connected(callee)->id.name.str), ""),
      remote_display, sizeof(remote_display));

ast_channel_unlock(callee);
callee = ast_channel_unref(callee);
}
{code}
By: Miguel Sanz (miguelsanzpardo) 2016-11-04 07:40:25.478-0500

I am having the same issue using Asterisk 11.21.2.

Im my example I am using Grandstream 21XX phones:
- Extension 1004 --> GXP 2130
- Extension 1001 --> GXP 2160
- Extension 1005 --> GXP 2170
- Ext.1004 calls to Ext.1001 and Ext.1005 has monitorized Ext.1001.

I have tried with notifycid=no, notifycid=yes, and notifycid=ignore-context.
With notifycid=no:
I cant see anything on BLF where I have monitored 1001 (I think that this is ok because notificid=no)
With notifycid=yes or notifycid=ignore-context :
I can see on BLF where I have monitored 1001 that 1001 is calling to 1001 (When the correct result should be "1004 is calling to 1001")

*extensions.conf*:
{noformat}
[from-internal-gs]
exten => 1001,1,NoOp()
same => n,Dial(SIP/${EXTEN},60,)
same => n,Hangup()

exten => 1004,1,NoOp()
same => n,Dial(SIP/${EXTEN},60,)
same => n,Hangup()

exten => 1005,1,NoOp()
same => n,Dial(SIP/${EXTEN},60,)
same => n,Hangup()

exten => 1001,hint,SIP/1001
exten => 1004,hint,SIP/1004
exten => 1005,hint,SIP/1005

exten => _**1XXX,1,Pickup(${EXTEN:2}@from-internal-gs)
same => n,Hangup()
{noformat}

*sip.conf*:
{noformat}
[general]
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-12.0.76.4(11.21.2)
disallow=all
allow=alaw
tonezone=es
callevents=no
rtpend=20000
rtpstart=10000
language=es
bindport=5060
jbenable=no
registertimeout=20
registerattempts=0
allowguest=yes
rtpholdtimeout=300
rtpkeepalive=0
rtptimeout=30
srvlookup=no
canreinvite=no
checkmwi=10
defaultexpiry=120
videosupport=no
g726nonstandard=no
maxcallbitrate=384
maxexpiry=3600
minexpiry=60
nat=never
ALLOW_SIP_ANON=no
externip=62.14.255.158
localnet=192.168.1.0/255.255.255.0
localnet=192.168.6.0/255.255.255.0
subscribecontext=from-internal-gs
trustrpid=yes
sendrpid=pai
notifycid=yes

[1001]
deny=0.0.0.0/0.0.0.0
disallow=all
secret=
dtmfmode=rfc2833
canreinvite=no
context=from-internal-gs
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=pai
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=31
pickupgroup=31
allow=alaw
dial=SIP/1001
mailbox=1001@default
permit=0.0.0.0/0.0.0.0
callerid=Edi <1001>
callcounter=yes
faxdetect=no
cc_monitor_policy=
subscribecontext=from-internal-gs

[1004]
deny=0.0.0.0/0.0.0.0
disallow=all
secret=
dtmfmode=rfc2833
canreinvite=no
context=from-internal-gs
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=pai
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=31
pickupgroup=31
allow=alaw
dial=SIP/1004
mailbox=1004@default
permit=0.0.0.0/0.0.0.0
callerid=Albert <1004>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
subscribecontext=from-internal-gs

[1005]
deny=0.0.0.0/0.0.0.0
disallow=all
secret=
dtmfmode=rfc2833
canreinvite=no
context=from-internal-gs
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=pai
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=31
pickupgroup=31
allow=alaw
dial=SIP/1005
permit=0.0.0.0/0.0.0.0
callerid=Oscar <1005>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
subscribecontext=from-internal-gs
{noformat}

*Notify dialog-info+xml send by Asterisk to the Ext.1005*:
{noformat}
NOTIFY sip:1005@192.168.1.169:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK2d1bcc18;rport
Max-Forwards: 70
From: <sip:1001@192.168.1.40>;tag=as392d3a2d
To: <sip:1005@192.168.1.40>;tag=60618332
Contact: <sip:1001@192.168.1.41:5060>
Call-ID: 801912290-5060-4@BJC.BGI.B.BGJ
CSeq: 130 NOTIFY
User-Agent: FPBX-12.0.76.4(11.21.2)
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 544

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="28" state="full" entity="sip:1001@192.168.1.40">
<dialog id="1001" call-id="pickup-801912290-5060-4@BJC.BGI.B.BGJ" local-tag="60618332" remote-tag="as392d3a2d" direction="recipient">
<remote>
<identity display="1001">sip:1001@192.168.1.40</identity>
<target uri="sip:1001@192.168.1.40"/>
</remote>
<local>
<identity display="1001">sip:1001@192.168.1.40</identity>
<target uri="sip:1001@192.168.1.40"/>
</local>
<state>early</state>
</dialog>
</dialog-info>
{noformat}

I understand that the correct dialog-info+xml shoul be something like:
{noformat}
NOTIFY sip:1005@192.168.1.169:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK2d1bcc18;rport
Max-Forwards: 70
From: <sip:1001@192.168.1.40>;tag=as392d3a2d
To: <sip:1005@192.168.1.40>;tag=60618332
Contact: <sip:1001@192.168.1.41:5060>
Call-ID: 801912290-5060-4@BJC.BGI.B.BGJ
CSeq: 130 NOTIFY
User-Agent: FPBX-12.0.76.4(11.21.2)
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 544

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="28" state="full" entity="sip:1001@192.168.1.40">
<dialog id="1001" call-id="pickup-801912290-5060-4@BJC.BGI.B.BGJ" local-tag="60618332" remote-tag="as392d3a2d" direction="recipient">
<remote>
<identity display="1001">sip:1001@192.168.1.40</identity>
<target uri="sip:1001@192.168.1.40"/>
</remote>
<local>
<identity display="1004">sip:1004@192.168.1.40</identity>
<target uri="sip:1004@192.168.1.40"/>
</local>
<state>early</state>
</dialog>
</dialog-info>
{noformat}

I am going to upload a trace made with ngrep  (notifycid=yes) when 1004 calls to 1001 and 1005 try to do a pickup  --> Ngrep_Trace-notifycid_yes.txt

Is there somebody that doesnt have this problem using Asterisk 11.X could say to me what revision is using exactly? With 11.21.2 this problem is appearing and the person that creates this issue doesnt have the problem using Asterisk 11.X, he says that has this problem using Asterisk 13.X
By: LTCtech (LTCtech) 2016-11-04 15:31:52.408-0500

@miguelsanzpardo I was using 11.6 certified 15 when it worked. It'd be interesting to find out in which exact version it broke and do a diff of the changes.
By: Miguel Sanz (miguelsanzpardo) 2016-11-04 18:32:59.971-0500

@LTCtech thanks by the info, if anyone is using an Asterisk between 11.6 and 11.21 it would be interesting to know if this problem appears or not with the corresponding Asterisk 11.X revision. The more information we have it will be more easy to find in which version was broken this functionality.

By: Alejandro Padilla (drazyck) 2016-11-07 04:28:29.023-0600

Hi guys, i have this working on asterisk 11.21.2 , with snom telephones, i only haf this options on sip.conf:

trustrpid=yes
sendrpid=pai

and hints configured also..

The bug start on asterik 13.0.0 version, i dont know if the problem persist on actual lastest version 13.12.1.

Regards
By: Sinisa (siny) 2016-11-10 02:03:10.466-0600

The same problem is present in 14.1 too...


By: Loic Didelot | https://www.mixvoip.com/asterisk-bounties (mixvoip) 2016-12-09 04:22:07.708-0600

I confirm the issue on latest asterisk 13.13.1 and the issue is not present in asterisk 11.

I would like to add the following information. Asterisk sends 2 notify packets. The first one is wrong as stated in this bug report but the second packet which is sent 5-10 seconds later is correct. So if you let it ring for a longer time the phones show the correct remote id.
By: Volker Kettenbach (vsauer) 2017-03-03 15:41:23.557-0600

Can confirm this with Asterisk 14.3.0
By: Loic Didelot | https://www.mixvoip.com/asterisk-bounties (mixvoip) 2017-04-19 15:54:03.644-0500

I would like to offer a bounty of 1000 euro if someone can fix this and get the patch upstream into the official asterisk branch.
By: Volker Kettenbach (vsauer) 2017-04-20 01:31:52.978-0500

I developed a workaround:

https://github.com/kettenbach-it/asterisk-snom-pickup-info-xml-agi
By: Kamik (Kamik) 2017-07-25 14:12:41.591-0500

yealink has the same problem with v11 . and support does not respond. - http://forum.yealink.com/forum/showthread.php?tid=40767

Paying tribute to Siemens/Unify - Openstage on the same asterisk 11.25.1 works without problem.
Someone will fix NOTIFY or you can forget it?
By: Thierry Magnien (tmagnien) 2017-08-10 07:25:21.795-0500

Hi. I proposed a patch for code review. It does not really fixes the bug but may be an acceptable workaround.

In fact, XML is built when device reaches ringing state, but at this stage, the underlying channel is not in ringing state and is not yet bound to the monitored extension. So remote id can't be determined.

I did not find the perfect solution but result is that several (3 in my tests) NOTIFY are sent to the SNOM phone, first and second being empty, the latest containing the correct values. SNOM phone then displays the correct values.
By: Daniel Wagner (ipefongmbh) 2017-11-22 07:51:00.558-0600

hey core team, are you interested in resolving this bug? is it on any roadmap? thanks
By: Joshua C. Colp (jcolp) 2017-11-22 07:57:37.949-0600

The chan_sip channel driver is in extended support. This means that it is up to the community to fix this issue. I can't speak for anyone else as to whether they are actively working on.
By: Friendly Automation (friendly-automation) 2018-03-05 08:40:40.851-0600

Change 8299 merged by Jenkins2:
chan_sip: Emit a second ringing event to ensure channel is found.

[https://gerrit.asterisk.org/8299|https://gerrit.asterisk.org/8299]
By: Friendly Automation (friendly-automation) 2018-03-05 09:04:31.655-0600

Change 8300 merged by Jenkins2:
chan_sip: Emit a second ringing event to ensure channel is found.

[https://gerrit.asterisk.org/8300|https://gerrit.asterisk.org/8300]
By: Friendly Automation (friendly-automation) 2018-03-05 09:08:52.938-0600

Change 8302 merged by Jenkins2:
chan_sip: Emit a second ringing event to ensure channel is found.

[https://gerrit.asterisk.org/8302|https://gerrit.asterisk.org/8302]
By: Friendly Automation (friendly-automation) 2018-03-05 09:11:00.920-0600

Change 8301 merged by Jenkins2:
chan_sip: Emit a second ringing event to ensure channel is found.

[https://gerrit.asterisk.org/8301|https://gerrit.asterisk.org/8301]