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 05:07:50.000-0600 | Date Closed: | 2018-03-05 08:40:39.000-0600 | ||
Priority: | Major | Regression? | Yes | ||
Status: | Closed/Complete | Components: | Channels/chan_sip/General Core/General | ||
Versions: | SVN 13.1.0 | Frequency of Occurrence | Constant | ||
Related Issues: |
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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] |