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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:

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 supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm > > Yes, the GUI is not that much user friendly as Zoiper is - but at least a very good and stable client for testing purposes ;-)...

...e that is why I don't get any SDP coming back. The rfc says the ACCEPT SHOULD be present so I'm thinking that is a Asterisk bug perhaps. In example 1 My own UAC code generated OPTIONS includes the Accept header yet still I see no SDP coming back from endpoints. I have tried using X-lite and PhonerLite softclients. I'm hoping there is a simple explanation or something I can do. Is Anyone able to query codec capability for any endpoints outside of a normal INVITE? I would like to know how you do so. Below is excerpt from the automatic OPTIONS query I see in the sip logs from setting verify=t...

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 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan&...

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

...> > > > 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 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&time...

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

...> > > > 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 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&time...

...originate) a call and pass both the caller id name and number? 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 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan&...

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

...To: <sip:991 at 192.168.1.4:5060>;tag=003d3418e2fce011b081701a0413e3f3 Call-ID: 010fdb653903a2022b99ed1d40c0b8db at 192.168.1.2:5060 CSeq: 102 OPTIONS Contact: <sip:991 at 192.168.1.4:5060> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Server: SIPPER for PhonerLite Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111025/fef06a6d/attachment.htm>

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 Polycom 331.? I set the externip to the router real IP address. Also, I set nat=yes. What could be the problem? Why I am able to regi...

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 file to the configuration of the the sip client) In the WSS option, I assume browsers negotiates for the the tls certficate and keys. Below are my debug code and the brief logs, http....

...tion: Originate ActionID: 100 Channel: SIP/1000 Exten: 1 Context: createcall Priority: 1 Timeout: 3 CallerID: SIP/1000 Variable: OriginateCallId=100 Async: true Is there a configuration setting I am missing? I've tried calling a Linksys SIP phone and I've also tried it with PhonerLite SIP Client, both are doing the same thing. Have a great day! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101001/e27cdd71/attachment.htm

...te (originate) a call and pass both the caller id name and number? 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 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan&...

...tion 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 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&f...

Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: > On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: >> Hi, >> >> Got problems with incoming SIP calls. >> >> Scenario: >> >> Server1: 3cx or any other server >> >> Server2: Asterisk 16.2.1 . PJPROJECT 2.8 >> >> Server2 registers on Server1 with SIP ID 1121.

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 inline:SL+jOTOj8J1jTFgC+ETx5ORfFEWB5kxk...

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