[Asterisk-Users] hybrid clients

[Scott Huang]

Hi, folks,

Is it possible to connect a IAX softphone to a SIP softphone via Asterisk?

IAX client -- Asterisk -- SIP client

I tried that, and I was able to dial and talk to my IAX client from the
SIP client. But not the other way around, I couldn't dial the SIP
client from the IAX client. The SIP client was not ringing. Asterisk
showed some WARNING messages:

Aug 21 20:24:38 WARNING[11044]: chan_sip.c:1047 retrans_pkt: Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Critical
Request)
  == No one is available to answer at this time (1:0/0/0)

What was wrong? Your help is highly appreciated.

-Scott


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Re: [Asterisk-Users] hybrid clients

[Scott Huang]

Thanks, Lance.

The two softphones and the Asterisk server are on the same subnet.
There is no NATing here. I'll turn on the show peers option in sip.conf
and iax.conf to see if that help.

I also have a hardphone connected to FXS on the asterisk server, it has
no problem talking to either the SIP softphone or the IAX softphone.

Thanks

-ScottOn 8/21/05, Lance Grover <[EMAIL PROTECTED]> wrote:
On 8/21/05, Scott Huang <[EMAIL PROTECTED]> wrote:>  Is it possible to connect a IAX softphone to a SIP softphone via Asterisk?>>  IAX client -- Asterisk -- SIP client
>>  I tried that, and I was able to dial and talk to my IAX client from the SIP> client. But not the other way around, I couldn't dial the SIP client from> the IAX client. The SIP client was not ringing. Asterisk showed some WARNING
> messages:It is compleetly possible, it sounds like the problem you have couldbe due to nat traversal on the sip client - sip does not like nattraversal.  Also make sure both phones are registering using "sip show
peers" and "iax show peers" from the asterisk cli.--Thanks,Lance Grover___Asterisk-Users mailing list
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[Asterisk-Users] Voice Encryption

[Scott Huang]

Hi,



Does Asterisk support encryption of voice traffic? I found following
wiki that describes IAX RSA authentication. I was able to implement the
public/private key authentication among three Asterisk servers
connected using IAX protocol. I am not certain if voice traffic can
also be encrypted among the Asterisk servers. Your help is highly
appreciated.



http://www.voip-info.org/tiki-index.php?page=Asterisk+iax+rsa+auth



Thanks in advance



-Scott
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Re: [Asterisk-Users] Voice Encryption

[Scott Huang]

Hi, Michael and Tim, 

Thanks for the help. It looks like Mark Spencer checked in some basic
IAX encryption code in Dec, 2004. When "encryption" is set, IAX
encrypts frames using AES128 which is the only supported algorithm. As
of clients, I would think KIAX and DIAX would work as well. 

Thanks

-Scott


Date: Wed, 28 Sep 2005 14:28:11 -0700
From: Michael Jia <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Voice Encryption
To: Asterisk Users Mailing List - Non-Commercial Discussion
        
Cc: [EMAIL PROTECTED]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"

Hi, Scott

The source code I found is in chan_iax.c, from the name of parameters, it
looks like it is using
AES. Can anyone here confirm that?

static int encrypt_frame(aes_encrypt_ctx *ecx, struct ast_iax2_full_hdr *fh,
unsigned char *poo, int *datalen)
{
int padding;
unsigned char *workspace;
workspace = alloca(*datalen + 32);
if (!workspace)
return -1;
if (ntohs(fh->scallno) & IAX_FLAG_FULL) {
struct ast_iax2_full_enc_hdr *efh = (struct ast_iax2_full_enc_hdr *)fh;
if (option_debug)
ast_log(LOG_DEBUG, "Encoding full frame with length %d\n", *datalen);
padding = 16 - ((*datalen - sizeof(struct ast_iax2_full_enc_hdr)) % 16);
padding = 16 + (padding & 0xf);
memcpy(workspace, poo, padding);
memcpy(workspace + padding, efh->encdata, *datalen - sizeof(struct
ast_iax2_full_enc_hdr));
*datalen += padding;
workspace[15] &= 0xf0;
workspace[15] |= (padding & 0xf);
memcpy_encrypt(efh->encdata, workspace, *datalen, ecx);
if (*datalen >= 32 + sizeof(struct ast_iax2_full_enc_hdr))
memcpy(poo, workspace + *datalen - 32, 32);
} else {
struct ast_iax2_mini_enc_hdr *efh = (struct ast_iax2_mini_enc_hdr *)fh;
if (option_debug)
ast_log(LOG_DEBUG, "Encoding mini frame with length %d\n", *datalen);
padding = 16 - ((*datalen - sizeof(struct ast_iax2_mini_enc_hdr)) % 16);
padding = 16 + (padding & 0xf);
memset(workspace, 0, padding);
memcpy(workspace + padding, efh->encdata, *datalen - sizeof(struct
ast_iax2_mini_enc_hdr));
workspace[15] &= 0xf0;
workspace[15] |= (padding & 0x0f);
*datalen += padding;
memcpy_encrypt(efh->encdata, workspace, *datalen, ecx);
if (*datalen >= 32 + sizeof(struct ast_iax2_mini_enc_hdr))
memcpy(poo, workspace + *datalen - 32, 32);
}
return 0;
}



On 9/28/05, tim panton <[EMAIL PROTECTED]> wrote:
>
>
> On 28 Sep 2005, at 07:26, Michael Jia wrote:
>
> Hi, Scott
>
> This is Michael Jia. So far, I searched the lists and with the following
> email threads
> http://lists.digium.com/pipermail/asterisk-dev/2004-December/008295.html

> I don't know what is the current working status now. Maybe somone in the
> lists
> knows.
>
>
> Last I heard IAX2 in CVS Head has encryption that works between 2 asterisk
> systems. There is no documentation (yet) except the source code, and no
> clients that support it (yet).
>
> Tim.
>
>
>
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[Asterisk-Users] Re: Voice Encryption

[Scott Huang]

I went over the code. AES128 is the only algorithm that is suppored
today. More importantly there are some concerns on the vulnerability as
discussed in
http://lists.digium.com/pipermail/asterisk-security/2005-August/60.html.
People are using UDP VPNs to satisfy customer requirements.
http://lists.digium.com/pipermail/asterisk-users/2005-August/120293.html

Thanks

-Scott

--

Message: 6
Date: Wed, 28 Sep 2005 19:09:17 -0500
From: Andy Hamilton <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Re: Voice Encryption
To: Asterisk Users Mailing List - Non-Commercial Discussion
        
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1

> Does Asterisk support encryption of voice traffic? I found following wiki
> that describes IAX RSA authentication. I was able to implement the
> public/private key authentication among three Asterisk servers connected
> using IAX protocol. I am not certain if voice traffic can also be encrypted
> among the Asterisk servers. Your help is highly appreciated.

There has been a little discussion of this topic on the
asterisk-security list somewhat recently.
You may want to look at the message archive from August (
http://lists.digium.com/pipermail/asterisk-security/2005-August/thread.html

) as well as the one from September.

Hope this offers a little help; as someone mentioned, not much
documentation is out there.
>From the August thread:

>Basically it's an automatic features (CVS-HEAD only, btw) where anytime
>a pair of IAX peers have encryption set to "yes" (or, as I understand
>it, a mutually-agreed-upon and supported algorithm) then it just happens
>automagically.
(Brian Capouch)

-a
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[asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Unknown)

[Scott Huang]

2012/12/19 Scott Huang 

> Hi
>
>I've saw some similar case in the mail list, but seems no standard
> answers, so I decide ask here again.
>
>Is there anyone see the message below ? I use asterisk(1.8.11-cert 9)
> in my openbts2.8, and when I made a phone call, the Asterisk CLI poppd the
> following messages.
>
> =
> *CLI>   == Using SIP RTP CoS mark 5
> -- Executing [8690@phones:1] Dial("SIP/IMSI466974600011287-",
> "SIP/IMSI466974104638690") in new stack
> [Dec 18 16:00:49] WARNING[2934]: app_dial.c:2218 dial_exec_full: Unable to
> create channel of type 'SIP' (cause 20 - Unknown)
>   == Everyone is busy/congested at this time (1:0/0/1)
> -- Auto fallthrough, channel 'SIP/IMSI466974600011287-' status
> is 'CHANUNAVAIL'
> ==
>
>The attached files are the sip.conf and extension.conf and wireshark
> trace log.
>
>The part of my setting in sip.conf is:
>
> [IMSI466974104638690];
> callerid=8690 <8690> ;
> regexten=8690;
> canreinvite=no
> type=friend
> allow=gsm
> context=phones
> host=dynamic
> registertrying=yes
>
> [IMSI466974102820333];
> callerid=0333 <0333> ;
> regexten=0333;
> canreinvite=no
> type=friend
> allow=gsm
> context=phones
> host=dynamic
> registertrying=yes
>
>
> [IMSI466974600011287];
> callerid=1287 <1287> ;
> regexten=1287;
> canreinvite=no
> type=friend
> allow=gsm
> context=phones
> host=dynamic
> registertrying=yes
>
>The part of my setting in extensions.conf is:
>
> [phones]
> exten => 8690,1,Dial(SIP/IMSI466974104638690)
> exten => 0333,1,Dial(SIP/IMSI466974102820333)
> exten => 1287,1,Dial(SIP/IMSI466974600011287)
>
>   How to exactly configure asterisk for a sip call ? Thanks very much !
>
> BR/Scott
>
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