PJSIP still has bugs

Don’t use PJSIP yet is is very buggy

Mod edit: this was taken from New installation reload failed still

What does that have to do with anything in this thread?

Another issue: PJSIP extensions disappeared from extension menu. I am sure I installed PJSIP and it disappeared after reboot today.

That has nothing to do with his issue and it’s not very buggy at all

Chan sip is community supported only. Digium doesn’t work on it. Its probably more buggy.

I would suggest that many here would disagree with that contention, especially while trying to connect to trunks

Many have no issue with chan_sip, quite a few have had a problem with chan_pjsip with quite a few providers

There are numerous threads here on these fora that support my contention, no?

Chan sip is also unsupported by digium.
Pjsip is supported by digium.

It’s easy to say this when many providers have had support docs on how to configure chan sip but they have none for pjsip. That doesn’t mean pjsip is very buggy. Thats making a very broad statement with no actual facts and if it is that buggy then why aren’t you reporting bugs to digium.

The error on line 11 is unable to connect to asterisk manager. The manager password in the database doesn’t match the password in manager.conf.

Pjsip extensions are disappearing from the interface because freepbx can’t determine the version of asterisk. So it defaults to chan sip.

Sure. But a broad claim of “very buggy” should be backed up by actual facts. Because others in the future will reference this thread and not use pjsip “because dicko said so”. We should be presenting facts. Not opinions. Does it have bugs. Sure. Is it “very buggy”. Not at all.

I have issues with these statements because they are so broad and it appears the person (could be anyone claiming ‘very buggy’) who says it isn’t actually doing anything to help digium out other than spreading rumors and false information about technologies.

If someone feels it’s so buggy then go out and report some bugs to digium.

Therefore I will always sidetrack or derail a conversation if I see a broad statement made with no facts behind it. Have facts? Fine. Did you report them to digium???

Last year at the digium Astricon developers conference I stated to the developers that the community (freepbx community) thinks pjsip is “very buggy”. The asterisk team was shocked. They hadn’t judged that from their own bug tracker. Here we are a year later and I see false claims like this and it’s aggravating. I will be going back in October to report the same thing and digium will tell me “Where’s the bugs”

My reply. “Sorry my community is lazy?”

You tell me


Because the VSP’s are often not supporting Digium’s version of PJSIP and Digiums PJSIP is not always concordant with those VSP’s , I agree, yes you can bitch to the VSP, you can bitch to Digium, so FreePBX is not really a factor here , by default you choose 5060 for PJSIP, so unless clever you will be stuck in that catch 22 under such in-congruent circumstances, but pragmatically you CAN stay with chan_sip and arrange for it to use 5060 while all this sh*t goes down.

If you have a rock solid solution to make PJSIP work on all Trunks, please post it, until then, . . . .

Challenge accepted.

Name a provider and I shall provide. If I can’t I will open a bug with Digium.

1 Like

I don’t have a problem with any of my providers on chan_sip, why rock the boat yet as it all works?

This is what you said. I asked you which providers and you can’t even give me a list. You just want to tell me that chan sip is rock solid. Way to help the community. Or asterisk. Or digium. You’re just trolling now. You don’t want a “Real” solution for pjsip. You just want to tell people it’s “very buggy”. If that’s your agenda then I will always be there to rebut it because it’s unsubstantiated.

I await proof.

You will get no proof from me, quite a while ago I sampled PJSIP, there where too many bugs then, and until I see an absolute need for PJSIP on a trunk, I just don’t need it, it as all experiential, so quite reasonably I shared it, if you believe that I am trolling then that would be you, I personally believe you are wrong in suggesting that. But ultimately to all the other users here, “do your trunks work on PJSIP?”

Please don’t take it personally, without doubt Digium’s PJSIP does NOT work yet with many VSP’s, that is NOT your fault nor Sangoma’s and I never suggested otherwise:-)


At least with Sangoma phones @tonyclewis has suggested very recently that PJSIP still has weird issues, see

These two posts were in the last 3 weeks…

To me it sounds like it’s not fully stable yet…

Chan_SIP does have issues mind you but they are well known…

Have a nice day!


1 Like

We are talking about providers. Not phones. Also the two issues you liked to from Tony was Tony making an assumption to help the user out. Never said it was 100% PJSIP

Are not both just considered “Endpoints” in the SIP milieu , The signalling should be symetrical and ideally be identical, the responses, maybe not so much. Each connection will be parochial, but some without doubt are failing.

Never mind, I am out of here, I can offer no real solution apart from the original one I suggested ( try chan_sip if chan_pjsip fails you)

Good Luck

Thanks. Then perhaps because your “very buggy” comment is a thought you had from over a year ago and you apparently didn’t document any bugs about what you experienced to anyone and you aren’t even willing to give me the name of a provider who are you to say it’s “very buggy” a year later. You don’t even know what has been fixed in a year and you aren’t even willing to try. I don’t think anyone should take any comment you have about pjsip seriously.

Probably so, but in those last few months, nothing has failed me , so as I say, it is all pragmatic, at some point in the future, when there is an essential reason to use PJSIP, I will perhaps sample it again, please go in peace and stop "ragging on me " :slight_smile:

The problem is that if something’s not working, it’s not always easy to get to the bottom of it so you can actually present it as a bug. It probably is one, but nobody is gonna confirm that for you if you haven’t been able to do it yourself.

Nobody wants to end up with a full fledged PJSIP system in production, then have problems and then have to revert everything back to chan_SIP.

If it’s smaller bugs we could live with and wait for a fix, ok, but most of what I read recently seems to be more substantial, things that really affect call quality and that a normal user would not want to live with for long.
E.g. the thread about Sangoma phones randomly unregistering, etc. Those kind of problems can become nightmarish.

Is there a PJSIP forum? If not, seems like one would be useful to support the transition and track issues. Appears like PJSIP is a big enough issue to require special attention.

I agree with Andrew, just opening a thread to say PJSIP is buggy and having no specific issue with PJSIP is not the right thing to do. What was going on in dicko’s mind?

I did start using PJSIP very early (about two years ago) and I did face a few issues many months ago. Some of these issues were due to PJSIP being different to chan_sip but that did not mean PJSIP was broken (example: assyncronous codec issue with some phone brands) and ONE issue was due to a bug that ended up fixed in Asterisk 13.10.2.

I run two PBXs that are 100% PJSIP (trunk and all) and these boxes (virtual machines actually) are rock solid. I consider PJSIP quite stable already and someone coming here should not feel discouraged by this thread.

1 Like