similar to: OPTIONS support for SDP

I have been sending OPTIONS requests both programatically (my own code), manually via SIP VERIFY PEER x and automatcially by setting verify=yes in sip.conf. The trouble is I do not see anything except an ACK 200 come back from endpoints and it does not contain any SDP/codec info. . My goal is to determine audio and video codec capability in advance of a call INVITE. I notice the Asterisk generated

Microsip (Windows) is free and small. 2.5Mb download, 10Mb RAM usage, does everything I need and configuring TLS is a doddle. http://www.microsip.org/ On 16 February 2017 at 13:04, Max Grobecker <max.grobecker at ml.grobecker.info> wrote: > Hello, > > I'm a big fan of PhonerLite. > It's more poplar in Germany, but also available in English language. > This client

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

Hi at all, i?m trying to use Asterisk 1.8.4 with tls over softphone blink on Windows 7. I configure all like in this tutorial https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial. But it doesn?t work. I?ve tried it with PhonerLite to unsuccesfully. Best Regards Karsten

Hello; I have asterisk?Asterisk 1.8.23.0-vici and Polycom 331 and I am able to register from local area network and not able to register from outside the office. Also from outside the office, I am able to register via PhonerLite softphone and not able to register via Zoiper softphone. So from outside the office, I am not able to register from Zoiper softphone and not able to register from

An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com> Subject: [asterisk-users] With ARI,

Hi, I'm trying to connect to the asterisk pbx via wss, from sipml5.org demo page (http://sipml5.org/call.htm). I used the guide from https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial , to setup the tls. I could make a secure sip call ( SRTP) using the PhonerLite sip client. ( This confirms my sip - tls settings and tls certficates. ( I'd added the tls client certficate

When calling Originate Action, it rings my phone. If I do not answer, I receive a Channel Event: Hangup, followed by receiving an OriginateResponse event with a Failure Response, Reason 3. My phone continues to ring. If I answer the phone at this point, it attempts to answer, but does not succeed. Looking at the asterisk debug, it appears to destroy the SIP dialog for the call. It also

Hi Jöran, Would it be possible to see an example using curl of how you are passing the PAI Header through ARI create? Dan From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Jöran Vinzens Sent: Friday, August 7, 2020 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] With

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

Hi Dan, i did it wrong, sorry: curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,

Hi, I have set up an asterisk with TLS and SRTP support. The SRTP is working with Phonerlite softphone. I have problem with the SRTP, when I make calls on Audiocodes gateway . I got the folloowing messages on asterisk: [Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80

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

Hello The end-of-line or new-line character is not interpreted when I open a shared file using MS notepad. The file was created on a Sun Solaris system - The contents of the file is " I newline am newline testing newline samba" when I do a hex dump of the file on Unix I can see the 0d 0a at the end of each line and the same on the Windows side but when I open the file with Notepad I get

Hello, I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLS authentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authentication enabled? Any suggestions? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Asterisk receives a 180 Ringing with SDP from the called side, then it sends 180 without SDP to the calling side. We would like asterisk to sends to the calling side the same response that was received from the called side. This is Asterisk cert 13.1, is that a new behavior, is there a setting to change this ? ?

Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through which I connected an external PSTN line. I use it as carrier for VoIP calls. I can make successfully calls, but there's one problem, I receive 200 OK with SDP with delay (sometimes more than 30 seconds). So when I make a call through asterisk I receive intially: - 100 Trying - 183 Session Progress, with SDP when the called

Hi All First time poster, long time reader. I just ran into something really bizarre. I've enabled videosupport and have been testing sip calls with Xten Eyebeam software, it works (mostly) However, when I have Videosupport=yes In the [general] section of my sip.conf, broadvoice calls fail w/ "We're sorry your call cannot be completed at this time" So... I've

Hello list, I have a question here that may be a little bit strange for some of you. I would like to send an INVITE from Asterisk to a given client without any SDP anouncement in it. Indeed, that is pretty useful for Click to call applications for instance, where you have no way to know which codecs are supported by the client you try to reach. Moreover, you let the client decide the

How does Asterisk (1.2) handle a 180 WITH SDP? I am seeing different behavior when a call is initiated from an Asterisk server and from an alternate point. With Asterisk, I am hearing ringing and with the other origination point, I am getting a message played on the far-end indicating to wait while the call is connected. I am wondering if the ring in the 1st (Asterisk) scenario is being played