similar to: PJSIP works on UDP but not TCP

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

Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP. tcpdump on asterisk server showing UDP packet bound for my remote endpoints internal IP: 17:07:57.130212 IP

Ok Dan, try this... I was able to get this to work behind a NAT and with ip address authentication. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp *local_net=<yourlocalnet I.E. 10.10.10.10/24 <http://10.10.10.10/24>>external_media_address=<your public ip address>external_signaling_address=<your public address>*

Joshua Colp wrote: <snip> > Remove "transport=transport-tcp" from your endpoints. Joshua...I did that but now my endpoints won't register. Kind Regards, Chirag -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150304/ea40a0bd/attachment.html>

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

Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed to make it work, all my terminals spa Cisco 5XX look my cli [Jul 8 11:09:16] ERROR[14733]: pjsip:0 <?>: tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:16] WARNING[14733]: pjsip:0 <?>: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)!

Hello! Oh, wise ones, ponder with me over two of the surprises that populate the universe! I have a phone, that I sometimes cannot reach, connected via pjsip. It can call other extensions just fine, it can call out over a trunk to my cell, all is well, but getting a call? Forget it most of the time. Here is all the config relevant to that phone: [murftest12] type=aor qualify_frequency=1992

Here's an update... My network admin would not turn off the ALG because it would cause several other problems to other phone systems we have. He looked at the sip trace. What he found is the PJSIP trace showed a different IP address than the older chan_sip so he had me change the aor contact to outbound.vitelity.net At this point, it seems to be working (and this is going through a Cisco

Tom Eastep wrote: > | Have you applied the ipsec+netfilter patches ? Without them, packets > are > | only seen encrypted in the OUTPUT chain. > | > Yes -- the ipsec+netfilter patches are applied. Here is the same test > with the bridge removed and the local ip address transfered to one of > the network cards: The problem is ipv4_sabotage_out in the briding code. It

On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp <dan at amtelco.com> wrote: > > Yes, everything is behind the same NAT. > > > > For the application I?m working on, the only endpoint is the endpoint to > Vitelity. > > We use AMI to Originate calls from Asterisk endpoint through Vitelity to > phones. > > After that, we control the call through AMI to perform the

On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp <dan at amtelco.com> wrote: > > Hi George, > > > > Thank you for looking into this. > > This is behind a nat? > > > Just to be clear...both the pbx and local endpoints are behind the same NAT? > [global] > > type = global > > debug = yes > > > > [transport1] > > type = transport

On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com> wrote: > > I am not sure if I entered the correct settings for the transport > information. > > For the local_net, I entered my local ip address, but no mask. I will > check with the network admin so he can verify the settings I entered. > > > You need the network and mask. For example if the ip

> 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

Greetings ! My goal is to get Twilio trunking working, and with TLS/SRTP. I see this concerning message in my log: [Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object of type 'endpoint' with id ?twilio' from configuration file ?pjsip.conf? Thus, ?pjsip show endpoints? does not show the endpoint for the Twilio trunk. Hoping for a sanity check of

Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes sendrpid=yes When I use these settings to originate calls using the sip.conf they sent me, everything works. Action: Originate ActionID: S8 Channel:

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

On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp <dan at amtelco.com> wrote: > > Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0. > > > > Same problem is happening with both of them. > > > > Could this be caused by PJPROJECT 2.3? > > > > Anyone have any suggestions for what I can try? > > > > My boss is giving me until

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 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: