Re: [asterisk-users] sending sip style messages in response
Benjamin Jacob wrote: Hello ppl, Is it possible to send SIP messages as response to the calling UA on failure, for e.g. if a number is blacklisted I would want to send Forbidden to the caller, not just for user comfort but also for testing purposes? I see only Congestion available which sends Service Unavailable. Hangup(CALL_REJECTED) or Hangup(21) should work, I think. Mikael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRTP
Khaled Chehab wrote / skrev: Is SRTP available in asterisk? Or how to implement it ? am using trixbox Regards http://www.e164.org/wiki/AsteriskSRTP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SRTP
Khaled Chehab wrote / skrev: Is SRTP available in asterisk? In a SVN branch. See http://bugs.digium.com/view.php?id=5413 Mikael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicemail - direct call
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... aHR0cDovL3d3dy52b2lwLWluZm8ub3JnL3Rpa2ktaW5kZXgucGhwP3BhZ2U9QXN0ZXJpc2srY21k K1ZvaWNlTWFpbAoKSXQncyBhbGwgaW4gdGhlcmUKCk9uIDIvMTMvMDYsIFRvbWlzbGF2IFBhcuhp bmEgPHRwYXJjaW5hQGxhbWEuaHI+IHdyb3RlOgo+Cj4gSGkgbGlzdCEKPgo+IEhvdyB0byBzZW5k IGEgY2FsbCBkaXJlY3RseSB0byB2b2ljZW1haWwgcmVjb3JkaW5nPwo+Cj4gV2hlbiBJIHB1dCB0 aGlzCj4gZXh0ZW4gPT4gMzEzLG4sVm9pY2VNYWlsLHUyMjEKPiBPciB0aGlzCj4gZXh0ZW4gPT4g MzEzLG4sVm9pY2VNYWlsLGIyMjEKPiBJbiBteSBkaWFsIHBsYW4sIGNhbGxpbmcgcGVyc29uIGZp cnN0IGhlYXJzIGdyZWV0aW5nIG1lc3NhZ2UgKGJ1c3kgb3IKPiB1bnZpYWJsZSkuIEkgd291bGQg bGlrZSB0byBhdm9pZCBncmVldGluZyBtZXNzYWdlIChJIHdvdWxkIHBsYXkgc29tZXRoaW5nCj4g d2l0aCBQbGF5YmFjayBhcHBsaWNhdGlvbikuIElzIGl0IHBvc3NpYmxlPwo+Cj4KPiAtLQo+IFRv bWlzbGF2IFBhcmNpbmEKPiB0cGFyY2luYSNsYW1hLmhyCj4gX19fX19fX19fX19fX19fX19fX19f X19fX19fX19fX19fX19fX19fX19fX19fX18KPiAtLUJhbmR3aWR0aCBhbmQgQ29sb2NhdGlvbiBw cm92aWRlZCBieSBFYXN5bmV3cy5jb20gLS0KPgo+IEFzdGVyaXNrLVVzZXJzIG1haWxpbmcgbGlz dAo+IFRvIFVOU1VCU0NSSUJFIG9yIHVwZGF0ZSBvcHRpb25zIHZpc2l0Ogo+ICAgIGh0dHA6Ly9s aXN0cy5kaWdpdW0uY29tL21haWxtYW4vbGlzdGluZm8vYXN0ZXJpc2stdXNlcnMKPgo=___ --Bandwidth and Colocation provided by Easynews.com -- Thank you, but this is how I see your mail. How can I see it right? http://lists.digium.com/pipermail/asterisk-users/2006-February/146742.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP over TCP: latest news?
Mimmus wrote: Hi, I know it is a FAQ but I'm interested in latest news (if any...) about SIP over TCP support in Asterisk. I found this: https://savannah.nongnu.org/projects/asterisk-tcp/ but I'm not able to understand if project is active and what is its level of development. Thanks Mimmus The patch has been added to Digium BTS, and it's waiting for the new socket interface according to bug report #4903. http://bugs.digium.com/view.php?id=4903 /Mikael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wich IAX soft client allow to specify a different server port?
Antonio Gallo wrote / skrev: [user[:secret[EMAIL PROTECTED]peer[:portno][/exten[@context]] Well but i don't need to dial out, i need to register to asterisk using IAX and 8080 port and all the client i've tested will not allow that into their account config section: they just have the server name/ip not the port. Try to enter server-name:8080 instead of the server-name only in the account config. It think it works if the program uses libiax2. /Mikael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wich IAX soft client allow to specify a different server port?
On Sat, Jan 07, 2006 at 01:19:34PM +0100, Antonio Gallo wrote: I still having problem with remote SIP client, trying to use IAX client instead but i've to specify TCP port 8080 (because of firewall). The IAX protocol is based on UDP, not TCP. I did this on server in bindport=8080 in iax.conf but i cannot find a soft client that allow to set wich server port to use. Any idea? Iaxcomm should work. You can use a complete dial string with username, secret, peer, port number, extension and context if you like, in the following format. [user[:secret[EMAIL PROTECTED]peer[:portno][/exten[@context]] /Mikael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: connect more the one phone to ONE sip Acoount
Olle E Johansson wrote / skrev:
Andreas Koch wrote:
Hello,
how is it possible to connect (register) more the one Phone to One
Sip-Acoount.
With, for example sipgate.de this is not a special feature, it is common.
We have users, what like to have more then one Phone, - Homeoffice
and Desk
Rigth now if a other phone registers whith the data, the first ist
removed.
You have to consider that Asterisk is a multiprotocol PBX and that the
PBX need to be in control of each device connected to the PBX. With
multiple registrations for one account we would break the Asterisk
architecture unless we did some very clever stuff. This has been
discussed quite a lot of times, so please search the mailing list for
more information.
I understand that allowing multiple registrations would break chan_sip,
but how can it break the Asterisk architecture if the forking is done by
the Dial application? Would it really matter if the dial string contains
multiple SIP AOR:s/users, which is possible today, or multiple bindings
for one SIP AOR?
Example of Dial with multiple SIP AOR:s/users, working today:
Dial(SIP/user1SIP/user2)
Example of Dial with multiple bindings for one SIP AOR, expanded by FOO:
Dial(${FOO(user1)}) = Dial(SIP/user1/10.1.1.1SIP/user1/10.2.2.2)
/Mikael
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Re: [Asterisk-Users] forwarding Sip call to IAX and vice-versa
On Fri, Apr 22, 2005 at 12:54:33AM +0300, Daniel HAIDUC wrote: hello everyone. i am a new user with asterisk and just playing with it, trying to make different configurations. i have managed to make a call from a sip to another sip phone and recently from iax to sip, but not from iax to iax and iax to sip below are my extensions.conf and the error asterisk is reporting. i tryed to keep everything as simple as possible. thank you for your responses daniel ... exten= 2000,1,Dial(IAX/dani,20) exten= 2000,2,Answer exten= 2000,3,Hangup ... Executing Dial(SIP/nicu-3cbb, IAX/dani|20) in new stack Apr 21 20:30:42 WARNING[3497]: channel.c:1901 ast_request: No channel type registered for 'IAX' Apr 21 20:30:42 NOTICE[3497]: app_dial.c:746 dial_exec: Unable to create channel of type 'IAX' Try IAX2 instead of IAX in the dial strings. /Mikael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play Sound File Without Answer Channel ??
Angel Diaz wrote: Hi list, Is there anyway to play a Sound File without answering the channel ? I have tried Playback(myfile, noanwer), but there is no audio there... Could you help me please ? Angel It's working with an ISDN phone and zaphfc and asterisk with bristuff patches for me. What type of channel are you using? According to show application playback: Not all channels support playing messages while still hook. /Mikael Magnusson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Shorewall firewall rules
On Sat, Apr 02, 2005 at 11:10:28AM +0200, Remco Barende wrote: I'm trying to get firewalling working but I am clueless as to which ports I need to open, I keep opening more ports and it's not working :( Basically I want SIP and IAX2 to work. IAX2 works fine, but SIP is giving me a headache. It seems that the stateless firewall is not able to handle SIP. I'm using shorewall as my firewall with these rules: ACCEPT netfwudp 4569 ACCEPT fw net udp 4569,5060,1:2 My rtp.conf says this: rtpstart=1 rtpend=2 Whenever I make a call I get these messages: Apr 2 09:18:25 pbx kernel: Shorewall:fw2net:REJECT:IN= OUT=eth1 SRC=myip DST=80.118.132.66 LEN=200 TOS=0x00 PREC=0x00 TTL=64 ID=116 DF PROTO=UDP SPT=17798 DPT=7356 LEN=180 Apr 2 09:18:26 raveon kernel: Shorewall:net2fw:REJECT:IN=eth1 OUT= SRC=80.118.132.66 DST=myip LEN=200 TOS=0x00 PREC=0x00 TTL=53 ID=859 PROTO=UDP SPT=7356 DPT=17798 LEN=180 So it seems that the %*$*$^ server is still trying to out out via a port lower than the range set in rtp.conf What is port 7356 for and what should I open to get it to work? I looked through the wiki but the low level iptables rules posted there do not make any sense to me. Port 7356 is used by the called site to receive rtp packets. I don't think you can have any influence to which port it chooses to use. You will need to allow outgoing udp packets to all ports between 1024 and 65535. For example: ACCEPT netfwudp 4569,5060,1:2 ACCEPT fw net udp 1025:65536 /Mikael Magnusson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo / delay problem
On Sat, Mar 19, 2005 at 07:19:39AM -0600, Rich Adamson wrote: If you are outside the US, there isn't much you can do since the x100p card was specifically designed to operate with US 600 ohm impedance pstn lines. If you have a x100p clone, it is likely the problem. Replace it with something capable of matching the pstn impedance for whatever country you are located in. Loading wcfxo with option opermode=1 should select CTR21 mode, which according to wcfxo.c (in zaptel) should be used in Austria, Belgium, Denmark, Finland, France, Germany, Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands, Norway, Portugal, Spain, Sweden, Switzerland, and UK. The default is to use FCC mode, which is used in the US. Regards, Mikael Magnusson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo / delay problem
On Sat, Mar 19, 2005 at 09:22:22AM -0600, Rich Adamson wrote: If you are outside the US, there isn't much you can do since the x100p card was specifically designed to operate with US 600 ohm impedance pstn lines. If you have a x100p clone, it is likely the problem. Replace it with something capable of matching the pstn impedance for whatever country you are located in. Loading wcfxo with option opermode=1 should select CTR21 mode, which according to wcfxo.c (in zaptel) should be used in Austria, Belgium, Denmark, Finland, France, Germany, Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands, Norway, Portugal, Spain, Sweden, Switzerland, and UK. The default is to use FCC mode, which is used in the US. The x100p does not have a programable chipset for impedance matching. That parameter was intended for the TDM card. Ok, maybe it isn't programmable, but there seems to be two different DAA's used in the clones[1]. Si3012 for the US market and Si3014 for the global market. I assume this has to do with the line interface. I bought a clone in Sweden that has the Si3014. It should support the line impedance used in Sweden I think, since it's marked with the following label: Clas Ohlson AB Modell/Malli: AMI-IE92 Art.nr/nro:32-2055 CE For use in Sweden, Norway and Finland. To be connected to the public switched telefone network. Hereby, Clas Ohlson, declares that this modem is in complience with the essential requirements and other relevant provisions of Directive 1999/5/EC. [1]http://www.intel.com/design/modems/linecard.htm Regards Mikael Magnusson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to Flash() a modem line
Raoul Bönisch wrote: * Eric Wieling [EMAIL PROTECTED] [2005-03-14 16:56]: Raoul Bönisch wrote: Flash is an analog thing. It does not even apply to ISDN. So how does the R key on my ISDN-telephone work then? Raoul An ISDN-telephone uses Q.931 messages for signaling, for example HOLD to put a call on hold and RETRIEVE to pick up one. /Mikael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I reload extensions included in a switch statement in extensions.conf?
Hi, I have two Asterisk servers and I forward calls from one to the other. How do I reload extensions included in a switch statement in extensions.con? I have tried extensions reload, reload and restart now, and it's only restart now that works. Is this how it is supposed to work or can it be a misconfiguration? Regards, Mikael Magnusson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can X100P/X101P detect reversal of line polarity?
On Thu, Jul 08, 2004 at 08:30:59PM +1200, Richard Scobie wrote: One thing to watch for here is RX gain if busydetect does not seem to be working after trying all the combinations. I had a 2 x X101P setup which busydetected perfectly - TX and RX gains were at the default levels. The X101Ps were replaced by a TDM card with 4 x FXO modules and with no config changes, busydetect stopped working. After incrementing in 1dB steps, an RX gain of 3.0 brought back reliable busydetect. I look forward to Rich Adamsons forthcoming writeup on setting up the gain distribution in an Asterisk system, to get everything working optimally. In Sweden we don't get a busy signal when the remote part hangs up. Instead remote hang up is signaled by reversing the polarity of the line. Can X100P/X101P detect polarity reversal when off hook, on hook, or both? Regards, Mikael Magnusson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
