similar to: PJSIP works on UDP but not TCP

Hi all, I have Asterisk 13 running and I'm currently trying to get PJSIP working on TCP. My transport looks like this. My box is not behind NAT. [transport-tcp] type=transport protocol=tcp bind=0.0.0.0:5061 My endpoint looks like this: [user1] type=endpoint transport=transport-tcp context=local_out disallow=all allow=alaw allow=ulaw allow=g722 auth=user1 aors=user1 direct_media=no

Joshua Colp wrote: > Chirag Desai wrote: > >* Joshua Colp wrote: *> >> >* <snip> *> >> >* > Remove "transport=transport-tcp" from your endpoints. *> >> >> >* Joshua...I did that but now my endpoints won't register. * > That should have no impact on things. Can you clarify what you mean by > it doesn't register?

Hi all, I was using 13.5 but upgraded today to 13.9 (13.10 came out a few hours after I upgraded). On both 13.5 and 13.9 asterisk seems to use 100% of the CPU. This usually happens a few hours after starting asterisk. A restart of asterisk gets the CPU back down, but only for a little while. There asterisk box has no call traffic flowing through it, just 15 or so registrations. I'm sure

> Joshua Colp wrote: > > There should be nothing different, except for how you configure things. > What is the full PJSIP configuration? What is the environment where > Asterisk is running? Is ICE actually in use on the other side? What is > the full SIP trace? > The full configuration is here: http://pastebin.com/XqZG1m5X I am connection over TLS / SRTP on port 5063. When

I'm dialling from the snom and every few calls asterisk sends media to the phones external IP and it works! And then now and again it sends the media to the phones internal IP and I hear nothing. I'm really at a loss. -------------- next part -------------- An HTML attachment was scrubbed... URL:

From: Matthew Jordan <mjordan at digium.com> > > > >> If the INVITE request is not shown in the CLI with 'pjsip set logger > >> on', then Asterisk is not actually receiving the request. > >> > >> Does a pcap show the message being sent to the correct IP/port? If you > >> change the transports to bind to port 5060, does that change

OK, it stopped working. It turns out the transport and endpoints in PJSIP are ok. I can send an invite from my unregistered snom phone and I can see some activity in the CLI. However, when I dial from my snom to Kamailio and have it pass the message to asterisk, PJSIP seems to ignore the sip messages even though they are there. Is there something wrong in the invite that I'm missing? U

Hi, I want to have Kamailio in front of one or more Asterisk boxes. I don't think it is necessary for Kamailio and Asterisk to register with one another. I'd like for PJSIP to recognise Kamailio by its IP address. I have two boxes, both have public IP addresses, they also have private IP addresses and can communicate with each other. I have a Snom phone accessing Kamailio via its

So I see: EXTERNAL_SNOM_IP -> EXTERNAL_ASTERISK_IP (UDP, length 218, src: 60798, dst 11128) EXTERNAL_ASTERISK_IP -> INTERNAL_SNOM_IP (UDP, length 218, src: 11128 dst 60478 So i see udp from the phone, but there's no audio. I do also see some packets :: EXTERNAL_ASTERISK_IP -> EXTERNAL_SNOM_IP (ICMP, length 246, Destination unreachable (Host administratively prohibited)

Hi all, Sorry if this has been asked before. I searched a lot, but found conflicting answers, so hoping for some clarification. My question is does Asterisk 13 support OPUS? If so which version exactly? If asterisk 13 requires a patch, which is the correct one and where do I get it? Kind regards, C -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I saw Matt Jordan's recent Kamailio world talk and was interested in the idea he proposed of stripping out authentication and registration from asterisk and letting Kamailio handle it. All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding registrations to asterisk. In order to do what Matt suggested would I be correct in assuming I would have to use the

Chirag Desai wrote: >I've tried explicitly setting the IP in bind and leaving it as above. >Nothing seems to come into asterisk. Although, as mentioned I can see the >SIP messages when I ngrep 5061. I got it working, I can see the sip traffic in the CLI now. I was trying to match on the IP of kamailio, when really I should have been matching on the domain name in the sip message

This question was asked by Chirag on March 4 2015 earlier, but I am following exactly the same procedure here and I cannot even get my clients to register on Asterisk. Here's my PJSIP.conf: [transport-tcp] type=transport protocol=tcp bind=0.0.0.0:5061 ... [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=!all,ulaw direct_media=no rtp_symmetric=yes

> Could you please describe what is the benefit of that? Currently there are two ways to provide DISPFlagDefinition, via bool and SPFlag, I would like to make it only via SPFlags, it will be NFC and it will make the changes in parser simpler for moving five flags from from DIFlags to DISPFlags. Currently parser checks the presence of SPFlags to see if the definition is present in bool or spflag

Yes, removing the support for isLocal, isDefinition fields completely from ll files, currently the LLParser still parses it. I want to remove it and update the all the ll files which still uses it. Also the metadata read will support old format, no changes in that. so if ll file has isLocal and isDefinition it will result in parser error. But the bitcode read will work as usual. - Chirag.

Hello, Thanks for prompt reply. Motivation, In case of debugging of bi-endian dwarf and value known at runtime some of the raw values needs to be byte swapped to change the endianity. We have a llvm based compiler which generates such dwarf, TAG_variable marked as big endian and debugging on little endian host using lldb. The listed operator on dwarf issues does that, one operator instead of

Hello Paul, Thanks for the reply. Yes, I am only looking for dwarf support at the moment and planning to support both PLI/COBOL decimal types. Also thanks for the suggestion, you are right as it is going to be rare cases, so it will be better to implement a separate subclass to avoid memory overhead for others. >> (Somebody is actively working on scaled binary operations, although they

Hi, I'm currently using Asterisk 16.4.0 cert version and working on webrtc. For configuration perspective, I'm pretty much done with it but here the real issue I'm currently facing i.e. when setting parameters max_audio_streams & max_video_streams to any positive greater than 0 integer value in realtime (DB) of any endpoints. After running command "pjsip show endpoint

Hi everyone, I have just upgraded to Asterisk 13.7.0-rc2 and noticed that when I type "pjsip show endpoints" at the CLI, I get "No Objects Found". However, if I request information on a specific endpoint, (for example: "pjsip show endpoint 101") then I get all of the information for that endpoint as expected. This seems to have started as soon as I upgraded to

Hello all, There is one enhancement request open for dwarfv5, http://dwarfstd.org/ShowIssue.php?issue=191107.1 The request is for addition of dwarf expression operator to swap the top of the dwarf stack, the response seems positive but it may take some time till v6. I like to add that operator as vendor extension but I am not sure how to go about it for llvm/lldb. Currently I am using it as