similar to: SIP 482 Loop detected

Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked 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

Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2) Can someone see why SIP-registration fails ?? register => 092779077:XXXX at 85.119.188.3 [3starsnet] type=peer host=85.119.188.3 username=092779077 secret=XXXX fromuser=092779077 fromdomain=sip.3starsnet.com dtmfmode=rfc2833 canreinvite=no insecure=port,invite qualify=yes

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: libresolv.so.2(GLIBC_PRIVATE) is needed by package

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...

Hello all, We're trying to build a small IVR application to allow callers to use the Asterisk for outgoing calls in a 2 steps dialing mode. The context for outgoing calls is called "outgoing" (we have there an LCR and routing mechanism we want to use, depending on the destination). This is what we did, but it doesn't work: exten => _X., 13,

-----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

Hello list ! SETUP : Grandstream --sip--> Local Asterisk (NSLU) --iax--> Hosted Asterisk (VirtualDedicatedServer) --sip--> SIPprovider --> my CellPhone PROBLEM : I've noticed that when I put down the horn of my Grandstream it still takes a while for my GSM/CellPhone to stop ringing. INFORMATION : This is the output on the CLI of the local Asterisk-server : [Oct 3 17:40:33]

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

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 PDF

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 the trampoline of life's experiences, Striving towards a saintly life in

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

Hey, all. I'm looking -- if possible -- for a decent, multi-platform soft-phone. Specifically, Linux and Windows; that way, I'll go through the same issues my end 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

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > What kind of phone are we talking about, both yours that works and your > wife's that does not? Right! > 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

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

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? 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,

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

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 externip,

Hello, all. My apologies up front but I must be brain cramping on something very simple. I've tried to pare down my configuration to the absolute minimum for SIP traffic just to understand how it works. My incoming calls are not finding the "s" extension in my dial-plan. I am assuming SIP calls can do this. I am using Asterisk 1.6.1.1 sip.conf has nothing but: [general]

I am presently configuring a test Asterisk 11 server based on CentOS-6 and I need to employ a softphone for testing. The base repo has ekiga. The EPEL repo has twinkle. I lack the knowledge of whether other packages exist or might be better suited. Which of the two do you recommend? Or, alternatively, what other package might be a better choice? -- *** E-Mail is NOT a SECURE

Hi to all, I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper. I've tested h323 using ohphone and I can talk between them, then I've tested SIP with Twinkle softphones and function very well. Now I have to perform call from h323 to sip and viceversa. How can I do it ???? I receive h323 call from a Cisco Voice GW to my Asterisk and this call have to go to a SIP phone: