search for: csipsimple

I'm trying to set up csipsimple on my Droid X. But no joy. Can't get it to register. My sip.conf: [general] ........ tcpenable=yes ........ [Test] transport=tcp,udp type=friend secret=mytest host=dynamic context=cloud-out qualify=60 dtmfmode=auto insecure=port,invite disallow=all allow=ulaw I've tried both udp and tc...

Hello, Iam lookong for an Softphone for iPhor oder Android smartphone using togehter with an headset. I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP phone. Is there an better softphone? Or are there softphone solutions for PC desktop MAC or Android with an headset? I want to save cost for desktop phones. thanks Thomas

With softphone I mean linphone csipsimple or whatever. How should a dialplan lokks like? On 31.01.19 11:26, Antony Stone wrote: > On Thursday 31 January 2019 at 10:59:01, basti wrote: > >> Hello I use this dial paln: >> >> [o2-in] >> exten => o2,1,Answer >> exten => o2,n,Playback(hello-world) &g...

...as well > Regards, > Amit Patkar > > > On April 29, 2017 9:05:22 PM GMT+05:30, Thomas > <thomasitcom at gmail.com> wrote: > > Hello, > Iam lookong for an Softphone for iPhor oder Android smartphone using > togehter > with an headset. > I tried Zoiper and CSipSimple but quality was bad compared to an > desktop SIP > phone. > > Is there an better softphone? > > Or are there softphone solutions for PC desktop MAC or Android with > an > headset? > I want to save cost for desktop phones. > > thanks Thomas > > > -- &gt...

...t; on your configuration. directmedia is not explicitly enabled; I guess it's the default. Joshua basically says there is no way to control which ports are being used for SRTP because that it is "up the endpoint". Such endpoints, in this case, are mobile phones with software like csipsimple or gs-wave (or perhaps zoiper), and I see no way in these programs to define which ports to use for SRTP. Since I have no way to define which ports endpoints use for SRTP, I would have to open all UDP ports in the firewall, and I don't want to do that. Nat is currently not involved yet....

Hi, I followed the TLS/SRTP tutorial on the wiki [0] using Asterisk 11.8.1 on CentOS 6.5 x86_64 and CSipSimple on a Nexus with Android 4.4.x local wifi. The phone seems to register but directly after that things fall apart (turning SELinux off made no difference): *CLI> -- Registered SIP 'encrypted' at 10.0.0.137:58079 > Saved useragent "CSipSimple_crespo-19/r2330" for...

I'm having trouble setting up tls/srtp secure communications on my Asterisk server- I'm still rather new at working with Asterisk. I have enabled tls and encryption and I have csipsimple with tls build on the phone. I'm currently only testing one phone with this capability so far, and the rest still work in the current state. My logging looks like this with verbose turned up: [Jun 7 11:44:13] NOTICE[88483]: chan_sip.c:19842 handle_response_peerpoke: Peer '<user>...

...r > > > > > > On April 29, 2017 9:05:22 PM GMT+05:30, Thomas <thomasitcom at gmail.com> > > wrote: > > > > Hello, > > Iam lookong for an Softphone for iPhor oder Android smartphone using > > togehter with an headset. > > I tried Zoiper and CSipSimple but quality was bad compared to an > > desktop SIP phone. > > > > Is there an better softphone? > > > > Or are there softphone solutions for PC desktop MAC or Android with an > > headset? > > I want to save cost for desktop phones. > > > > thank...

...WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure (continiously) and client hears no sound. After i restart the client program it works fine again for a while. Then the same warning again. Asterisk 1.8.1.1, RealTime engine, sip peer has encrytion->yes The client program is CSipSimple on Android Here are some log file traces: Peer 0010101 is calling some number that is routed to context a2billing [2010-12-23 11:06:22] DEBUG[5941] sip/sdp_crypto.c: local_key64 3gWGFJAffj4Pn393BUPwe3/wOMx5/ndZyPtfno7L len 40 [2010-12-23 11:06:22] DEBUG[5941] sip/sdp_crypto.c: SRTP policy acti...

...ng the transport to UDP allows me to call between endpoints. In TCP however, I can see PJSIP send an invite, but then receives no responses. I've spent all evening trying to figure it out and am a bit stumped now, since changing to UDP works straight away. I'm testing with a snom 760 and cSipSimple, calls don't work in either direction and regardless of local network or mobile network. Any help would be greatly appreciated. Kind Regards, Chirag -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachment...

Hi experts, I am trying Asterisk SRTP in my environment, and find that when Asterisk is behind a NAT, the audi/video UDP ports opened for SRTP relay by Asterisk are local ports on the Asterisk server, media from the two clients out of the NAT (for example from Internet) can not reach the ports, and thus the two client can not establish the secure call via Asterisk. I have set up a STUN server

...e set up a STUN server and configured in rtp.conf, but >seems Asterisk does not do STUN before it opens ports for SRTP. BTW, >Non-SRTP call can work though. > > Anyone can give advice on how to make SRTP work in such an env? I have no problems with a TLS/SRTP call between a GSM with CSipSimple and Asterisk 11.8.1 behind NAT. Have you configured the NAT options in sip.conf? externip=... localnet=... nat=... You may also need to add/change the options below. Check the sip.conf example file to see what these options do and use what's best for your situation. canreinvite=no directmedi...

...o the u/d mode would not make sense any more; however this creates another interesting problem, pls read on) Some endpoints are grouped in pairs so that calling an extension, rings on both devices. (One 'device' is a real handset, usually dumb: SPA112 or SPA301, the other is a softphone (CSipSimple or WebRTC or both) used to bring the incoming CID to users' eye level and to perform some client-side CRM integration ) On Incoming call, as expected, the softphone shows me the CID [as intended] and I can pick up the handset, then the softphone will stop ringing; This far, it works as inten...

Hello I use this dial paln: [o2-in] exten => o2,1,Answer exten => o2,n,Playback(hello-world) exten => o2,n,Ringing exten => o2,n,Dial(SIP/10&SIP/20&Local/s at no-op,25,rt) exten => o2,n,Playtones(425/1000,0/4000) exten => o2,n,Wait(30) exten => o2,n,Hangup() All is fine. Hello world is Playback and I hear a ring tone. If I remove the Playback hello-world. No ring

What are the recommended settings to successfully implement VoIP over 3G/4G data connection. Assume we are talking about using Polycom phones, and the 3G/4G data connection comes from a Cradlepoint router that is plugged in with AC power and has high gain antennas. The device will be stationary, so we will not have to worry about tower handoff?s breaking the connection. This will be for fixed

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is quite conservative in what they want to use :) meaning handsets are important; As the handsets have

...o the u/d mode would not make sense any more; however this creates another interesting problem, pls read on) Some endpoints are grouped in pairs so that calling an extension, rings on both devices. (One 'device' is a real handset, usually dumb: SPA112 or SPA301, the other is a softphone (CSipSimple or WebRTC or both) used to bring the incoming CID to users' eye level and to perform some client-side CRM integration ) On Incoming call, as expected, the softphone shows me the CID [as intended] and I can pick up the handset, then the softphone will stop ringing; This far, it works as intend...

On 2/23/19 1:15 PM, Joshua C. Colp wrote: > On Sat, Feb 23, 2019, at 8:06 AM, hw wrote: >> On 2/22/19 7:56 PM, Joshua C. Colp wrote: >>> On Fri, Feb 22, 2019, at 2:48 PM, hw wrote: >>>> >>>> Hi, >>>> >>>> when trying to use SRTP, I can see UDP traffic from phones to the >>>> asterisk server being dropped be the firewall

...media is not explicitly enabled; I guess it's the default. >> >> Joshua basically says there is no way to control which ports are being >> used for SRTP because that it is "up the endpoint". Such endpoints, in >> this case, are mobile phones with software like csipsimple or gs-wave >> (or perhaps zoiper), and I see no way in these programs to define which >> ports to use for SRTP. >> >> Since I have no way to define which ports endpoints use for SRTP, I >> would have to open all UDP ports in the firewall, and I don't want to do &gt...