Yes, but after disabling ALG you need to put the external IP adres in the configs. Otherwise the server sends the wrong IP (unreachable) adres in the SIP messages.
I also find it very strange you have it working with proxy, ALG disabled and your local IP in the headers. That shouldnât work so I still suspect there is a ALG in there somewhere thatâs messing things up.
I understand that the provider is using it in the wrong way, but it will be great to be added in asterisk and/or pjsip for 0.00001% of the users This will not be against the RFC however probably not so much use it, but in the same time can help some. If someone knows how to make a little patch I will apply to my asterisk. Sorry but I had an average knoledge of asterisk more than 10 years ago (I believe it was <=1.6) know I only transfer the configuration
I havenât said such a thing canât go in or that it wouldnât, just that it doesnât currently work that way, and noone has written anything to do it. Itâs not something Iâm working on.
Isnât this up to the pjproject at http://www.pjsip.org ?
Although pjproject is bundled with asterisk, the source is downloaded from there during compilation with some patches (unless I totally misunderstood the process).
(Without any knowledge of C and pjproject my guess is that you could need to adjust it in sip_dialog.c > create_uas_dialog.)
Haha, just tried something similar but my change was this:
â<%s:dummy@%s%.*s%s:%d%s%s>â,
I donât think it would matter which âuserâ you pass and it can even be something nonsense (like âdummyâ) because it doesnât get used anyway. Sending a real user would even be a security issue. (Iâm not sure what your endpoint->contact_user containt at the moment)
BTW. Seeing the Contact header from the INVITE from the calling server, it also doesnât have the a âuserâ included, so why would it demand it from the other side. Really buggy server.
And also⌠âŚ/1/asterisk-16.30.1/res/res_pjsip.c ???
I thought you where on 20.4.0 ?
In one of my test, it was to go back to 16 (in the previous server I was using 16 with chan_sip), but it is the same mod and I didnât try to pass a wrong contact_user, if they test it I believe/hope they need it
Thank all to have pushed me to look in the code
I can confirm, I modified the contact_user to use a wrong one, and my 200 was not ack
Therefore, it is the confirmation that it is used, and it must be the right one
Sorry but I cannot contact the provider since they donât offer any kind of support
Mmm, thatâs weird. Iâm not sure to what (user)info they would check it against. 180/183 RINGING would have the same dummy contact. Unless itâs their login-username which is used there in endpoint->contact_user.
But Iâm glad it works now.
Itâs up to the developers to see if they make an option for this.
(although Iâm not sure if you would need to file it in the issue section for this to happen)
Everything goes through Github, and patches need to be under a contributor license agreement to be included. We wonât just take patches from this forum for example and include them.