similar to: Problems with firefly connection via SIP

Is there a key sequence to stop a call as its ringing, before the call is answered? The * key stops a call after it is answered, but I'd like a way to cancel the call during the ringing phase. /Obelix ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

I have been trying to connect via sip and things don't seem to work. What do messages like this mean? Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834 (len 361) to 216.127.66.119 returned -1: Invalid argument Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 000638cf3adb579455c0d20b2051ba1d@127.0.0.1 for seqno 102

Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. >From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using

After many months of development, I'm pleased to announced Firefly - an IAX soft phone and network. The firefly softphone - free, runs under windows, allows connection to multiple networks, skinable interface, connection to firefly network, IAX2 protocol, (SIP in next release), codecs supported - iLBC, G.711 ulaw/alaw, GSM. - contact lists, selectable ringtones. download from here -

hello all... newbie question: I have FireFly setup on my laptop and I would like to test this out using IAX in this scenario: FireFly Softphone > Asterisk > TDM Gateway i do not wish to use this on the firefly network, but simply within my own "3rd party" network as the website and setup of FireFly defines it... does anyone have a sample iax.conf and extensions.conf i

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because you current number it is not being answered, and you don't want to hangup before dialling again. /Obelix

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because the current number it is not being answered, and you don't want to hangup before dialling another number. /Obelix

Which Actions and events to the read/write options in manager.conf give access to, ie the options below. read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Are they documented somewhere? /Obelix

hello all, i have a strange problem....i am running SER in front of asterisk, and am testing softphones. x-lite works fine...i can dial, hang up, DTMF, all good. Firefly looks really cool and i'm very impressed with the IM-like interface and the skinning ability, but something strange is happening...when i call from the firefly and run something on the server and press hangup on the client,

Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password

Hi Everyone, I've got a weird problem with both Firefly & iaxLite (both IAX softphones). They don't seem to stop ringing when an incoming call is make to them. If the call is answered the conversation starts both ways but the ringing sound still keeps going and the softphones keep displaying that a call is coming in (but they do not display that the call is answered). I read

Hi All Has anyone used Cubix / Firefly successfully with Asterisk? When someone calls a Cubix softphone, Cubix never seems to answer the call correctly. The other person just hears ringing even though it has been answered. I am using IAX as the SIP support doesn't seem to 100% either. Idefisk works 100% on the same setup. Kind Regards Garth

I did get it to work, and can place and receive calls through the Firefly network via *. Compared to iaxtel or FWD, there is a significantly higher amount of latency, but it is workable. For some reason, this needed to be the last entry in my iax.conf or it would try to authenticate with a different user ID when receiving calls (and obviously would fail. Relevant section from my iax.conf:

I found version 2.0.0 of Firefly on the Freshtel site, but it only has the network setup options for the Freshtel network, despite the final statement on the page http://www.freshtel.net/firefly/download/ that says: ----------------- Standalone SIP / IAX mode: If you want to use Firefly on our network (with your own voicemail etc.) you will need to register a Firefly number. However, you can

Hi all, I try to connect two asterisk-server together. There is a server (obelix) which receives a call. This call should be transfered to another server. In my dialplan at obelix I have the following: exten => 920622201,1,Dial(SIP/outbound:geheim at asterix.local:${EXTEN}) exten => 920622201,n,Hangup exten => i,1,Congestion exten => t,1,Congestion If I call the number 920622201

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

On Ganeral --> Language correct from "portugese" to "portuguese". Kind regards, Miguel Date: Tue, 5 Oct 2004 09:47:08 +0100 (BST) From: gARetH baBB <hick.asterisk@gink.org> Subject: Re: [Asterisk-Users] Firefly 1.9.5 released To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Just a quick announcement for Firefly users that Firefly 1.9.5 is out. Mainly just a bug fix release as we get ready for Firefly 2.0. One notable feature added is DTMF via SIP INFO. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe is the URL As always, send me any bugs, features or suggestions. -Adam

I have an asterisk server up and running, using Firefly in IAX mode works great, even with Firefly behind a NAT (as expected, since IAX works really well with NAT). Now I'm trying to get X-Lite and/or Firefly to work in SIP mode from behind the NAT, and I can't seem to get there. At this point, the phone will successfully register with Asterisk, and the Asterisk qualify messages get

Hi folks an issue I don't understand. I'm running * stable 1.0.3 on public internet, with following iax.conf / sip.conf entries: iax.conf [100] type=friend username=Foo context=default auth=md5,plaintext,rsa secret=secret host=dynamic callerid="Foo" <100> qualify=no sip.conf [10] type=friend username=Bar context=default callerid=Bar <10>