[asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Unknown)
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 > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voice Encryption
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 DiscussionMessage-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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Voice Encryption
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 DiscussionCc: [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. > > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Voice Encryption
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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] hybrid clients
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 Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] hybrid clients
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 ___ 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