similar to: From the CLI, how can I hangup a channel name that includes a space character?

Thanks Doug. Turns out if using hangup request does not work with the escaped character CLI> hangup request PJSIP/1003\ a-00000007 Usage: channel request hangup <channel>|<all> Request that a channel be hung up. The hangup takes effect the next time the driver reads or writes from the channel. If 'all' is specified instead of a channel name, all

>>> Is there some control character(s) for the CLI to interpret everything in between as a single argument? I think you can typically use tab completion when working with spaces or you can escape the space with a back slash For example Doug Lytle would be Doug\ Lytle Doug

I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at

Hello! I'm still trying to get a working t.38 configuration w/ pjsip. I'm now able to send t.38 faxes to my own extension: hylafax -> t38modem -> extension -> extension -> t38modem -> hylafax. The fax is sent by t38modem. The receiving part of t38modem accepts the call, sends ReInvite for t.38 and things are working as expected. Now, let's do the nearly same

Hi, Is there a way to add another extension to a live dial, for example Dial(PJSIP/1000,,) and after 20 secondes change it to Dial(PJSIP/1000&PJSIP/1001,,) I am open to suggestions such as using manager or stasis. Thanks in advance. Best regards, Kkh -------------- next part -------------- An HTML attachment was scrubbed... URL:

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:

Hi Dan, as far as PPI and PAI Header, we use the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp <dan at amtelco.com> wrote: > An

I just realized that you are asking about dynamic AORs, not static Contacts in an AOR. That may be the difference. I have never actually tried giving a dynamic AOR a different name. And you wouldn't want more than one dynamic AOR, you'd just use an AOR that allowed more than 1 contact. On Wed, Apr 1, 2015 at 2:59 PM Trey Hilyard <kctrey at gmail.com> wrote: > I don't know

I am running asterisk 13.1.0 In pjsip.conf, the endpoint section has an aors and an auth field. I can name the auth field anything I want. The key is to set the auth=field accordingly. However, when I try this with the aors field, it never works. It seems I have to name the aors=field to match the name of the endpoint section. Is this correct? Would there ever be a need for multiple aors to

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

We have a product that uses Asterisk via AMI. I am relatively certain we used to be able to play prompts by actions like the following to make asterisk play the confbridge-join prompt when a new user joins the confbridge. However, that doesn't seem to work now. Action: SetVar ActionID: C58 Channel: PJSIP/1003-00000003 Variable: CONFBRIDGE(bridge,sound_join) Value: en/confbridge-join Does

On Sun, Jun 11, 2017, at 01:31 PM, Michael Maier wrote: > On 06/11/2017 at 04:39 PM Joshua Colp wrote: > > On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote: > > > > <snip> > > > >>> > >>> PJSIP uses a dispatch model. The request is queued up, acted on, and > >>> then that's it. The act of acting on it removes it from

On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote: > Has anybody any idea why asterisk drops the media stream in the 200 OK? > The channel has been T38_ENABLED before! Or is it necessary to add more > debug code? Who does the negotiating? > Only asterisk or is pjsip doing some parts, too? Asterisk does the T.38 negotiation and produces the answer SDP, PJSIP does the SDP

Hello, I am trying to force a registration and unregistration with my SIP trunks, but I see pjsip send unregister, but no register. I.e., I am looking for pjsip send register. Is there any such command? If so, why do I not see it in my CLI? Should I upgrade to 13.2.0? Any insight appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear friends This is my simple dialplan [demopjsip] exten => _X.,1,Dial(PJSIP/${EXTEN}@10.10.10.2) exten => _X.,n,Hangup() I need to dial out via an IP address, not using an endpoint, as shown above. It fails with Executing [19544447408 at demopjsip:3] Dial("PJSIP/federico-00000002", "PJSIP/195XXX7408 at 10.10.10.2") in new stack [Jun 26 00:39:00] ERROR[10136]:

On Mon, Jun 5, 2017, at 12:00 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote: > > On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > > > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: > > >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > > >>> Just a guess (without knowing about your network), but are

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)

On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote: <snip> > > > > PJSIP uses a dispatch model. The request is queued up, acted on, and > > then that's it. The act of acting on it removes it from the queue. > > That's the *expected* behavior ... . I rechecked again and again. All > existing tcpdumps. The "resent" package isn't part of

On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > Just a guess (without knowing about your network), but are the two ends > points on public networks and visible to one another? If not the reinvite > may be passing an internal (nat'ed) address to the other and the connection > will fail...just a though t38modem -tt -o /var/log/t38modem.log --no-h323 -u 91 --sip-listen

On 06/14/2017 at 10:17 PM, Joshua Colp wrote: > On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > <snip> > >> >> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >> Just one exception - and that's the package in question, which can't be >> seen in tcpdump. >> >> I extended the above patch by adding