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Hello, I have an error while try to install twinkle: # yum install twinkle [...] Resolving Dependencies --> Running transaction check ---> Package twinkle.i386 0:1.2-1.el5.rf set to be updated --> Processing Dependency: libresolv.so.2(GLIBC_PRIVATE) for package: twinkle --> Finished Dependency Resolution Error: Missing Dependency: libr...

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.11...

I am experimenting with Asterisk having downloaded and installed the FreePBX i386 CentOS-6.3 based distro and updated it. The current package level on this system is: asterisk11-11.3.0-49_centos6 freepbx-2.11.0beta2-112 I am using twinkle-1.4.2-7.el6 as a softphone testing tool. There is no firewall on the asterisk host and SELinux is disabled on it. Fail2Ban is installed but I have made no alterations to the default configuration, whatever it is. The asterisk host is configured as 192.168.6.122. The softphone is configured on...

The Gnu Telephony site: http://wiki.gnutelephony.org Has a Centos repo: http://dist.gnutelephony.org/RPMS/ But I caught some text stating that this is for Centos 4.2. Is it really? Is there a difference; i.e. would it be safe to install these on Centos 4.4? Really I am after Twinkle, and it seems there is a lot you need to actually get Twinkle installed...

Anyone have an rpm for twinkle? http://www.twinklephone.com/ A new sip softphone.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to test with a friend who has an Asterisk in his office with the Asterisk which I have in my house. Then I have an extension that he is trying to register remotely. Trying with the Twinkle client, I see that it is registered: - --------------------------------------------------------------------------- 400/400 190.0.163.57 D N 5060 OK (35 ms) - --------------------------------------------------------------------------- but to the few seconds I obt...

...proxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/00493513333333 Now I see this: if I call my phone (00493511111111) from Twinkle it works. If I call it from the phone of my wife, logged in on the same AsteriskNOW of Twinkle and able to speak with Twinkle, it does NOT work and I see that in the Log of my Asterisk: == Using SIP RTP CoS mark 5 [May 28 23:05:59] WARNING[4135]: chan_sip.c:12800 check_auth: username mismatch,...

...If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least, not connected to my Asterisk... If I connect it to the other VM with AsteriskNOW I can call my Twinkle, but NOT my phone connected on my Asterisk, using the "proxy". I can see that in the log: [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username mismatch, have <1234>, digest has <luca> [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: F...

Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in "allow=gsm" line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the...

Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On...

hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/...

...p:201 at 10.1.0.10>;tag=uucwz Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org CSeq: 183 INVITE Contact: <sip:201 at 10.1.0.65> Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 247 v=0 o=twinkle 2122879389 441437466 IN IP4 10.1.0.65 s=- c=IN IP4 10.1.0.65 t=0 0 m=audio 8000 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> - --- (13 heade...

Hi, I've successfully setup Asterisk on my local PC and can make call using Twinkle to the server. But, I cannot call to my Asterisk server at Rackspace. I have been trying several things to figure it out, no luck. My PC is behind NAT, so I've set that up in sip.conf (nat=yes). I can ping my Rackspace server so it seems to be Public-static IP. Anyway, I tried with setting ext...

...nd users do. I've noticed a couple (e.g., minisip, which seems abandoned, and sip-communicator, which, honestly, is probably a great IM client, but has a confusing interface for actual phone calls). So I'm wondering if anyone has any favorites. Failing multi-platform, I'll stick with Twinkle on Linux, and gladly take suggestions for Windows -- OSS if possible, but payware is acceptable. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

...ht! > Can you ping the unreachable phone and does it respond to a ping? I can ping both phones from the VM > Many phones will have a network test function built in to them to help you > determine if the phone is properly connected to the network. Unfortunately not that... I tried with Twinkle from my PC, using the same account of my wife (configured IDENTICALLY to my account, just another username). It don't work... I presume, I configured something wrong in Asterisk... > Do you see anything in the asterisk logs or the logs of the phone itself > (providing the phone puts lo...

...pe=friend > nat=force_rport,comedia > qualify=yes > qualifyfreq=60 > transport=Auto > avpf=no > force_avp=no > icesupport=no > encryption=no > callgroup= > pickupgroup= > dial=SIP/00493513333333 > > > Now I see this: if I call my phone (00493511111111) from Twinkle it works. > If I call it from the phone of my wife, logged in on the same AsteriskNOW of > Twinkle and able to speak with Twinkle, it does NOT work and I see that in the > Log of my Asterisk: > > == Using SIP RTP CoS mark 5 > [May 28 23:05:59] WARNING[4135]: chan_sip.c:12800 c...

Hi, I have already installed 4.2 on my laptop, I only chose GNOME at install time. I now need to use up2date to install kde (I wish to install twinkle softphone). What is the correct way to do that. I did: up2date kde* up2date -u kde* up2date -i kde* It printed a listing of the packages but did not install them. Thanks, Jerry

I tried to install justvoip several times and I cannot install it. Can somebody tell me how to install it on ubuntu? Meybe next version of WINE will support it?

...ked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register => 092779077:XXXX at 85.119.188.3 This the output of SIP show peers : asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status twinkle-candy/twinkle-can (Unspecified) D 0 UNKNOWN twinkle-jonas/twinkle-jon (Unspecified) D 0 UNKNOWN grandstream/grandstream 192.168.1.13 D 5060 OK (35 ms) 3starsnet/092779077 85.119.188.3 N 5060 UN...

I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip