Phones have stopped working? Getting error: PJSIP_EUNSUPTRANSPORT

all of a sudden phones are not longer able to register, they just stopped working. I’ve already tried rebooting 5 times and looking for any new updates and totally lost on this. Anybody come across this issue, below is a quick printout of the log. Currently using FreePBX v12.0.32

I can see that SIP trunk is up, but none of the phones work either making outbound or internal call.

[2015-01-15 11:35:05] WARNING[3351] res_pjsip_mwi.c: No contacts bound to AOR 114. Cannot send unsolicited MWI.
[2015-01-15 11:35:05] WARNING[3351] pjsip: tsx0x7f50d8005 .Failed to send Request msg OPTIONS/cseq=502 (tdta0x7f50d80041b0)! err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT))
[2015-01-15 11:35:05] ERROR[3351] res_pjsip.c: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending OPTIONS request to endpoint
[2015-01-15 11:35:05] WARNING[3351] pjsip: tsx0x7f50d800a .Failed to send Request msg OPTIONS/cseq=37852 (tdta0x7f50d80090d0)! err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT))
[2015-01-15 11:35:05] ERROR[3351] res_pjsip.c: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending OPTIONS request to endpoint
[2015-01-15 11:35:05] WARNING[1863] res_pjsip_mwi.c: No contacts bound to AOR 110. Cannot send unsolicited MWI.
[2015-01-15 11:35:05] WARNING[1863] pjsip: tsx0x7f50d800a .Failed to send Request msg NOTIFY/cseq=49669 (tdta0x7f50d80090d0)! err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT))
[2015-01-15 11:35:05] ERROR[1863] res_pjsip.c: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending NOTIFY request to endpoint 104
[2015-01-15 11:35:05] WARNING[1863] res_pjsip_mwi.c: No contacts bound to AOR 105. Cannot send unsolicited MWI.
[2015-01-15 11:35:05] VERBOSE[1974] netsock2.c: Using SIP TOS bits 96
[2015-01-15 11:35:05] VERBOSE[1974] netsock2.c: Using SIP CoS mark 4
[2015-01-15 11:35:05] WARNING[1974] chan_sip.c: Section ‘5163214111’ lacks type
[2015-01-15 11:35:05] VERBOSE[1974] config.c: Parsing ‘/etc/asterisk/sip_notify.conf’: Found
[2015-01-15 11:35:05] VERBOSE[1974] config.c: Parsing ‘/etc/asterisk/sip_notify_custom.conf’: Found
[2015-01-15 11:35:05] VERBOSE[1974] config.c: Parsing ‘/etc/asterisk/sip_notify_additional.conf’: Found
[2015-01-15 11:35:05] WARNING[1863] res_pjsip_mwi.c: No contacts bound to AOR 102. Cannot send unsolicited MWI.

hi,

I can confirm that everything is now broken on my PBX also with the same errors. This seems have been caused from updates that have been pushed out yesterday evening / today (my system regularly updates all modules till yesterday noon caused no issues, rolling back to 2 days ago renders pbx back operational but cannot upgrade as there seems to be a module that breaks everything at the moment.
Helpppp :slight_smile: we need pbx-guru to fix.
if i read logs correctly pbx no longer knows what pjsip is ? (res_pjsip.c: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending OPTIONS request to endpoint).
regards,

hello:
pjsip is a SIP stack that is used for SIP trunk or SIP phones. Maybe your pjsip conf files are broken. If you have a backup files for PJSIP, you can restore back and try.

@James : thanks but we know its a sip stack. No that doesn’t work what you suggest it is literally a module that schmooze/freepbx pushed out over the last day that broke everything. (rolling back to a serverimage of two days ago (with NOT updated modules works of course.)
There is a module that breaks everything and i hope schmooze lets us know what it is and when we can update again without breaking everything.

More info hoping @tm1000 will look at it.
Running freepbx shmz distro build 6.12.65-24 - asterisk 13

in short when updating to newest modules NO phones can connect and error es_pjsip.c: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending OPTIONS request to endpoint) is everywhere in the logfiles.

PBX no longer functional.

Regards,

I have already looked into it. Everything is working fine here.

Hi,
@tm1000 After upgrading a 6.12.65-24 asterisk 13 to the latest modules and any phone using pjsip (please reboot the pbx) they should now fail to work.
I just tried again with a working system upgraded all the modules everything keeps working !!! until you REBOOT (make sure you reboot when testing) as of then :

VERBOSE[1806] asterisk.c: Asterisk Ready.
WARNING[1835] pjsip: tsx0x7f013c010 .Failed to send Request msg OPTIONS/cseq=12687 (tdta0x7f013c00f8d0)! err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT))
ERROR[1835] res_pjsip.c: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending OPTIONS request to endpoint
resulting in no longer working pbx.
This is a standard schmooze freepbx distro nothing changed on it a few commercial modules thats it.
Regards,

I reboot quite often. Still works fine. So I rebooted again. Still works fine.

How about showing off your pjsip.transports.conf and pjsip.transports_custom.conf file.

hi,

@tm1000 yes no problem pasting the files here. But isn’t it odd the build with modules untill monday this week updated works. If i then update to the latest modules (friday) everything breaks. Why is this i am not changing any of these conf file myself it really is as i mentioned everything works until i upgrade to the latest modules of now. Between monday and friday are changes that have apparently an impact on the pbx and pjsip.

pjsip.transport.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
’#'include pjsip.transports_custom.conf

[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

the other file pjsip.transports_custom.conf is empty. NO conf file at all has been manually edited.
(if you need logfiles let me know can send them)
regards,

(thank you for looking into it) (the ‘#’ i added here as it didnt come out correctly in the forumpage.)

Give me the name of the module you think is breaking it

Same thing here… after last update all phones (using PJSIP) can not register!

I restored an old backup from last week and everything is back online now

@tm1000 hi, wish i could it is a whole bunch of modules. It is one of the modules that have been updated between monday and friday but i really do not know which one it is.
I tried for a 5th time : working system modules need updating all phones work i upgrade the modules do not choose betatrack modules and then reboot after reboot no pjsip phone registers ahd log is full with
ERROR[1835] res_pjsip.c: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending OPTIONS request to endpoint.
This for me confirms it is one of the modules. Which of the modules can potentially impact pjsip ??
(just asking : when you tested this did you tested it with schmooze commercial modules enabled , i have 6?)

Regards,

I am testing this on a development machine. All modules are enabled on that machine. Furthermore the only module that changes pjsip at this time is core and sip settings. You can rollback modules through module admin. Click check online then click core and you will see the options for rollback. I suggest you rollback until you figure out what version it happened at.

Stop updating all modules. I have asked a few times for a specific module and you keep going ahead and updating them all. Why? Let’s figure out the problem through proper troubleshooting.

Again I can’t replicate so this is a very small problem. Not saying your issue is low priority just that I can’t get it to happen. Make sure the transport for each extension is set properly and actually please send all of your pjsip.conf files. Anything that has pjsip in it.

The fact that I am here responding means I believe you but if I can’t replicate it then I can do anything. If you’d like to provide a vm image or backups that I can test with then that’s great too. But without that I can’t do much and I use pjsip daily.

Greetings,

Any new on this. I updated (callrecording 12.0.1.4 (current: 12.0.1.2)
certman 12.0.4 (current: 12.0.2)
framework 12.0.32 (current: 12.0.31)
ringgroups 12.0.3.1 (current: 12.0.2)

After Update I lost pjsip connections. When the phone requests re-register Asterisks is sending a 590 destination unreachable.

Thanks,
Edward

Here are the Files:
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
#include pjsip_custom.conf
#include pjsip.transports.conf
#include pjsip.endpoint.conf
#include pjsip.aor.conf
#include pjsip.auth.conf
#include pjsip.registration.conf
#include pjsip.identify.conf

[global]
type=global
user_agent=FPBX-12.0.33(13.0.0)

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
#include pjsip.auth_custom.conf

[5102-auth]
type=auth
auth_type=userpass
password=Passw0rd
username=5102

[2101-auth]
type=auth
auth_type=userpass
password=Passw0rd
username=2101

[3101-auth]
type=auth
auth_type=userpass
password=Passw0rd
username=3101

[4101-auth]
type=auth
auth_type=userpass
password=Passw0rd
username=4101

[5101-auth]
type=auth
auth_type=userpass
password=Passw0rd
username=5101

[5103-auth]
type=auth
auth_type=userpass
password=Passw0rd
username=5103

[8101-auth]
type=auth
auth_type=userpass
password=Passw0rd
username=8101

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
#include pjsip.transports_custom.conf

[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
external_media_address=192.168.0.20
external_signaling_address=192.168.0.20

After some Funness: We finally Flushed the arp table and then rebuilt them. After that we rebuilt the pjsip files and then the soft phones started connecting.