search for: pjsip_wizard

Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29] NOTICE[22492]: res_pjsip/pjsip_distributo...

Hello, I've just discovered PJSIP 's support of set_var setting in pjsip.conf. Is this setting also supported in pjsip_wizard.conf ? On a fresh 13.8.2, it doesn't seem but I may have missed somthing. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160425/96b4e791/attachment.html>

On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > I made some progress. The first thing I have realized is that it is my > Twilio configuration in pjsip_wizard.conf that was killing me. I have since > removed that entire file from /etc/asterisk and I am able to make > "from-internal" context calls (i.e., calls that do not leave the VoIP > island). > > Here's what I have right now in pjsip_wizard.conf (again, I have removed &gt...

...Joseph <gjoseph at digium.com> wrote: > > > On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I've just discovered PJSIP 's support of set_var setting in pjsip.conf. >> Is this setting also supported in pjsip_wizard.conf ? >> On a fresh 13.8.2, it doesn't seem but I may have missed somthing. >> > > I thought it did. Let me check. > It's working for me... [1000] type = wizard endpoint/set_var = xxxxx=ttttttttt > pjsip show endpoint 1000 ... trust_id_outbound : true...

Yes, it is enabled on port 5060. I do receive a TCP ACK back from the server, so I know the TCP segment is received at the server hosting the Asterisk build. On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: > > OK. Let me ask this. Is anything else necessary, except choosing TCP as > the

...t;> >>> On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07 at gmail.com> wrote: >>> >>>> Hello, >>>> >>>> I've just discovered PJSIP 's support of set_var setting in pjsip.conf. >>>> Is this setting also supported in pjsip_wizard.conf ? >>>> On a fresh 13.8.2, it doesn't seem but I may have missed somthing. >>>> >>> >>> I thought it did. Let me check. >>> >> >> It's working for me... >> >> [1000] >> type = wizard >> endpoint/set_...

...izard and reloading ("pjsip reload" on CLI): some data (specifically endpoint/pickup_group) is not modified. For example, initially I have empty pickup group: tiare*CLI> pjsip show endpoint xxx ... pickup_group : ... Then, I add endpoint/pickup_group = 0,3 to pjsip_wizard.conf, and reload: pickup_group remains empty. Then, if I change the line in pjsip_wizard.conf to endpoint/pickup_group = 0, 3 ^ note the space here! then reload, and I get what was expected: tiare*CLI> pjsip show endpoint xxx ... pickup_group : 0,...

...sport=tcp to my Origination URI after wireshark revealed everything was received as UDP into Asterisk, so we can rule out that issue (I confirmed that I am getting TCP based SIP INVITEs from Twilio, and confirmed that the Asterisk server sends a 401 Unauthorized for the initiation INVITE). Per the pjsip_wizard.conf samples, long ago, I removed pjsip.conf-based Twilio config and placed it all in pjsip_wizard.conf. Thanks, re: wiki, I will be using it heavily, for sure ;-) On Thu, Feb 18, 2016 at 9:56 PM, George Joseph <george.joseph at fairview5.com> wrote: > > > On Thu, Feb 18, 2016 at...

On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > Wow. Incredible. That worked. The backslash is important there; I kept > trying with no backslash and followed the instructions in > pjsip_wizard.conf.sample (in configs/samples) and it says we have to say > > transport=tcp ; the only example however talks about ipv4. > > Is this documented somewhere and I just missed it?? > > So, let me sum the issues and their solutions: > > (a) Inside/from-internal calling. Only ne...

...39;11.12.13.14:38124' (callid: 3532ca0d142e6ce92f0259fd51cb5e43 at 0.0.0.0) - No matching endpoint found The last time I had this error, I was dealing with another SIP trunk and the issue was that I had mixed up "identify" and with "identity", but I have not such type in my pjsip_wizard.conf which looks like this: type = wizard sends_auth = yes sends_registrations = no remote_hosts = sillyapp.pstn.twilio.com\;transport=tcp outbound_auth/username = gobble outbound_auth/password = degookdegook endpoint/context = from-external endpoint/disallow = all endpoint/allow = ulaw aor/qualif...

Hi, sometimes 'pjsip show registrations' shows registrations to the VOIP provider as Rejected. I have already added max_retries = 0 auth_rejection_permanent = no in pjsip_wizard.conf and still asterisk does not recover. I need asterisk to keep trying to register and to renew the registration without requiring manual intervention. How can I make asterisk do that?

...0 3:03:47 PM CET hw wrote: >> Hi, >> >> sometimes 'pjsip show registrations' shows registrations to the VOIP >> provider as Rejected. I have already added >> >> >> max_retries = 0 >> auth_rejection_permanent = no >> >> >> in pjsip_wizard.conf and still asterisk does not recover. >> >> I need asterisk to keep trying to register and to renew the registration >> without requiring manual intervention. How can I make asterisk do that? > No ideas? > > If pjsip is not able to recover after the internet connecti...

...tions > are: > > INVITE > < 100 TRYING > < 403 FORBIDDEN > > Or is this normal? Must Twilio respond with a 407 which will cause Asterisk > to authenticate? > > Twilio has a nice technical document to setup a trunk with PJSIP. It includes an example for a pjsip_wizard.conf https://assets.cdn.prod.twilio.com/documents/TwilioElasticSIPTrunking-AsteriskPBX-Configuration-Guide-Version2-1-FINAL-09012018.pdf Maybe that helps. And make sure for your outgoing calls to set the callerid to a valid caller Id which ist authorized with your twilio account. It will not allo...

I didn't use pjsip_wizard, I'm kind of crafting this by hand as I learn. I actually have a plain asterisk, and a FreePBX, system to help me learn. I sometimes create something in FreePBX to see what it does to the config files. So that's how I modelled my pjsip.X.conf files If I issue the command "pjsip sho...

...ttps://blogs.asterisk.org/2016/04/20/pjsip-dns-support/ which says: "chan_pjsip will now look for SRV records based on what transports are configured on the system". Does this mean there's now a way of doing it? Because https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip_wizard.conf.sample says: ; Hostnames must resolve to A, AAAA or CNAME records. ; SRV records are not currently supported. Hmmmm... I'm confused!

Hi all, My goal is to trunk two Asterisk servers together using res_pjsip. I'm really not familiar with res_pjsip, having only used chan_sip over a year ago now. So, I apologize in advance if this is an overly basic question. I'm using the below configuration guide for an outbound trunk. My question is: what would the trunk configuration look like on the other Asterisk server? Would it

...se let me know. This bug makes > > pjsip > > makes basically unusable :( > > There are various options in the outbound registration that controls > behavior. I'd suggest providing your actual configuration. I have put the options that should make asterisk re-register in pjsip_wizard.conf as much as I could find them like this: [easybell_HW] type = wizard sends_auth = yes sends_registrations = yes max_retries = 0 auth_rejection_permanent = no forbidden_retry_interval = 200 transport = transport-tls endpoint/cos_audio = 5 endpoint/cos_video = 4 remote_hosts = secure.sip.easybe...

...local_net                  = 127.0.0.1/24 local_net                  = 192.168.50.0/24 local_net                  = 192.168.1.0/24 local_net                  = 10.3.2.0/24 local_net                  = 10.20.1.0/24 local_net                  = 10.10.41.0/24 local_net                  = 10.5.1.0/24 pjsip_wizard ------------- [isoft-sr-in-1] type = wizard transport = transport-udp remote_hosts = 192.81.237.20 aor/max_contacts = 1 aor/remove_existing = yes aor/qualify_frequency = 60 aor/qualify_timeout = 2000 endpoint/ice_support = no endpoint/disallow = g723,slin,ilbc,lpc10,g729,speex,g726aal2,g722 endpoi...

On Thursday, February 27, 2020 3:03:47 PM CET hw wrote: > Hi, > > sometimes 'pjsip show registrations' shows registrations to the VOIP > provider as Rejected. I have already added > > > max_retries = 0 > auth_rejection_permanent = no > > > in pjsip_wizard.conf and still asterisk does not recover. > > I need asterisk to keep trying to register and to renew the registration > without requiring manual intervention. How can I make asterisk do that? No ideas? If pjsip is not able to recover after the internet connection has gone away for a...

...> >> sometimes 'pjsip show registrations' shows registrations to the VOIP > >> provider as Rejected. I have already added > >> > >> > >> max_retries = 0 > >> auth_rejection_permanent = no > >> > >> > >> in pjsip_wizard.conf and still asterisk does not recover. > >> > >> I need asterisk to keep trying to register and to renew the registration > >> without requiring manual intervention. How can I make asterisk do that? > > > > No ideas? > > > > If pjsip is not...