Hello everyone, I just cannot find a solution in this! The provider requires @ in the username ([email protected]
res_pjsip.c:3888 create_out_of_dialog_request: Unable to create outbound OPTIONS request to endpoint EEE–123456789 as URI ‘sip:[email protected] @www.xxx.com:5060’ is not validsip:[email protected] @www.xxx.com :5060 on AOR 123456
How can I use “@” in the username?
The provider gave me the instructions:[email protected]
Stewart1
October 12, 2020, 10:48pm
2
Finally, a post with good information.
Please try these trunk settings (replacing xxxx with the last 4 digits of your number):
Trunk Name: OTE–211402xxxx
Username: +30211402xxxx
Secret: (your password at OTE)
Authentication: Outbound
Registration: Send
SIP Server: ims.otenet.gr
SIP Server Port: 5060
Context: from-trunk
From Domain: ims.otenet.gr
From User: +30211402xxxx
Leave other settings blank or at their defaults.
Then, put these two lines into /etc/asterisk/pjsip.auth_custom_post.conf
[OTE-211402xxxx](+type=auth)
[email protected]
Restart Asterisk and test. If it won’t register, post the REGISTER request being sent, with Authorization header if available.
1 Like
Yeah, gave too much information though I think
Executing [s-INVALIDNMBR@macro-dialout-trunk:1] NoOp(“PJSIP/999-00000006”, “Dial failed due to trunk reporting Address Incomplete - giving up”) in new stack
Although, I used the same address for the sip trunk…
Stewart1
October 12, 2020, 11:19pm
4
Are you sending the destination number in the correct format? If OTE requires +30xxxxxxxxxx, your Outbound Route or trunk Manipulation Rules need to convert whatever you dialed to that format. Search the log for OUTNUM to see what it sent. If you still have trouble, compare the INVITE sent by your working chan_sip trunk with the one sent by pjsip.
What goes wrong on incoming? Possibly, the Contact header has the wrong address. Does anything get logged on an attempted call?
Pjsip sip log (with error):[email protected] :5060 SIP/2.0 sip:[email protected] ;tag=c13dea4d-32ef-4f15-a667-10c196cf2b0a sip:[email protected] sip:[email protected] :5060 sip:[email protected] ;tag=c13dea4d-32ef-4f15-a667-10c196cf2b0a sip:[email protected] sip:[email protected] ;tag=c13dea4d-32ef-4f15-a667-10c196cf2b0a sip:[email protected] ;tag=sbc0507f2stz0vf [email protected] :5060 SIP/2.0 sip:[email protected] ;tag=c13dea4d-32ef-4f15-a667-10c196cf2b0a sip:[email protected] ;tag=sbc0507f2stz0vf
CHAN SIP LOG (Correct)
INVITE sip:6********@ims.otenet.gr SIP/2.0 sip:[email protected] :5260 ;tag=as7e995a97 sip:6********@ims.otenet.gr sip:[email protected] :5260 [email protected] .164 [email protected] sip:[email protected] :5260 ;tag=as7e995a97 sip:6********@ims.otenet.gr .***.**.164 [email protected] sip:[email protected] :5260 ;tag=as7e995a97 sip:6********@ims.otenet.gr ;tag=sbc0504ncf61due-CC-79 sip:195.167.16.1:5060 sip:195.167.16.1:5060 .***.***.164 [email protected] sip:[email protected] :5260 ;tag=as7e995a97 sip:6********@ims.otenet.gr ;tag=sbc0504ncf61due-CC-79 sip:195.167.16.1:5060
Stewart1
October 12, 2020, 11:41pm
6
It appears that you don’t have From Domain set in your trunk. If you do, check for funny characters such as leading or trailing spaces.
Done!! Outbound is working!! But no inbound, and no logs. Tried enabling authentication in both inbound and outbound, but no luck…
Stewart1
October 13, 2020, 12:23am
8
Post the REGISTER request and the 200 OK response, for both the working chan_sip and the failing pjsip.
How can I capture the REGISTER request? When I restart asterisk, I cannot see the sip logs at same time and when I go back, it’s already registered…
Stewart1
October 13, 2020, 1:19am
10
Yes, it’s unfortunate that the pjsip logger option gets turned off by a reload.
For chan_sip, you can dosip set debug oncore reload
For pjsip, options include:
Turn logger on and wait for re-registration, possibly setting a shorter expiry if provider permits.
Add a config override debug=yes under [global]
Use sngrep, tcpdump or a similar tool to capture traffic independent of Asterisk.
Use an external traffic capture tool e.g. Wireshark with a managed switch.
After a lot of work, I finally have the logs:sip:[email protected] ;tag=as5a30053bsip:[email protected] [email protected] sip:[email protected] :5260
<— SIP read from UDP:195.167.16.1:5060 —>[email protected] sip:[email protected] ;tag=as5a30053bsip:[email protected] ;tag=sbc0508kzoom2y3sip:[email protected] :5260 ;expires=120
Pjsip (not working)sip:[email protected] ;tag=68dd4179-8e0a-4f5b-82b9-9788b7dbe008sip:[email protected] ;tag=s7yjgies
<— Transmitting SIP request (790 bytes) to UDP:195.167.16.1:5060 —>sip:[email protected] ;tag=68dd4179-8e0a-4f5b-82b9-9788b7dbe008sip:[email protected] sip:[email protected] :5060;line=jjjkmkg [email protected] ", realm=“ims.provider.gr”, nonce=“ZSpnLHh8dNCwF8p80QcLWg==”, uri=“sip:ims.provider.gr:5060”, response=“6d5dc9570a5dc4b5593a4dab8d8b349c”, algorithm=MD5
<— Received SIP response (539 bytes) from UDP:195.167.16.1:5060 —>sip:[email protected] ;tag=68dd4179-8e0a-4f5b-82b9-9788b7dbe008sip:[email protected] ;tag=yjh2uehstel:+3123456789 ,sip:[email protected] sip:[email protected] :5060;line=jjjkmkg ;q=1;expires=30
Stewart1
October 14, 2020, 1:10am
12
In opnsense, disable source port rewriting, restart Asterisk. With luck you’ll now see rport=5060. If this doesn’t work, please explain your network setup. Are you double NATted (opnsense + modem NAT)?
An alternative is to let the provider do NAT traversal. In Asterisk SIP Settings, add 195.167.16.0/24 to Local Networks, restart Asterisk.
system
November 14, 2020, 1:10am
13
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,had to dial it down a bit… It registers(!!!) but no outbound or inbound calls. Outbound - message: “The number you have dialed is not in service”- log: