[Asterisk-Users] Problem with codec negotiation
Title: Message Hi everyone I am having trouble with codec negotiation. I have Asterisk running at the office and a SIP phone at home. In my sip.conf, I have allow ordered as follows: alaw ulaw g729 and gsm On all my office extensions, I have no allow, or disallow entries. My Cisco gateway is setup to do alaw ulaw g729 and gsm My home phone does g729 alaw and ulaw. In sip.conf, I have disallow all and allow g729. In all my extensions and cisco gateway I have canreinvite set to yes and my dial commands don't have the t option, so all sip endpoints can talk directly to each other (rtp). If I call from home to the office, calls go through fine. SIP show channels, shows that the call is g729 as one would expect. If I get a call from Office to home, or from PSTN (via Cisco) to home, the phone rings, but as soon as I answer it hangs up. Asterisk says: May 29 05:45:49 WARNING[7514]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/390-8a3b(256) to SIP/192.168.44.23-08acccf0( -- SIP/390-8a3b is ringing -- SIP/390-8a3b answered SIP/192.168.44.23-08acccf0 May 29 05:45:55 WARNING[7514]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/192.168.44.23-08acccf0( to SIP/390-8a3b(256) May 29 05:45:55 WARNING[7514]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't make SIP/192.168.44.23-08acccf0 compatible with SIP/390-8a3b == Spawn extension (default, 390, 1) exited non-zero on 'SIP/192.168.44.23-08acccf0'SNIP. One week later. I have now purchased two 9.729 licences as I suspected Asterisk was not allowing direct endpoint negotiation. Now my home phone answers, but it is receiving on 9.729 and sending on g.711a This leads to delays building up on the g711a side. I want to calls coming to my office phones to use alaw as the prefered codec in sip.conf, but I want calls to my home and remote sites to be g.729. Asterisk seems to ignore the codec negotiation phase and insists on running two different codecs in two directions. Most sip servers will always use the same codec in both directions based on the first agreed codec. As my home phone is set in sip.conf to only allow g.729, then it should do g.729 in both directions. I see this as a bug. Anyone know how to make it work properly? Thanks to the gurus who might no the answer to this one. Cheers Mark P.S. Why do the real experts not use the users web forum? Much easier to manage than a mailing list. ___ 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] Getting a Cisco gateway to work with Asterisk
Thanks Steve I realised the other day that I don't want the Cisco to register with credentials. There is in fact a hidden credentials command in 12.3(8)T. What I did was take away all registration commands from my sip-ua block in the Cisco. I am using [EMAIL PROTECTED], so I have created a trunk through AMP. I have changed the settings in outbound trunk to the following and created an empty inbound trunk on the web page with no parameters. The result is that in Asterisk sip_additional.conf I have this block [cisco] context=from-pstn host=192.168.44.23 type=friend Now when I try to call into my gateway from the PSTN, I get the following line immediately after the Cisco does an invite Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.23:5060;branch=z9hG4bK3016D6 From: sip:[EMAIL PROTECTED];tag=391004-1A5E To: sip:[EMAIL PROTECTED] Date: Sun, 22 May 2005 14:29:25 GMT Call-ID: [EMAIL PROTECTED] Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 3143229573-3389264345-2148466707-2141291050 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 15 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1116772165 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 328 v=0 o=CiscoSystemsSIP-GW-UserAgent 23 4058 IN IP4 192.168.44.23 s=SIP Call c=IN IP4 192.168.44.23 t=0 0 m=audio 17780 RTP/AVP 8 18 98 3 0 19 c=IN IP4 192.168.44.23 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:98 GSM-EFR/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:19 CN/8000 20 headers, 14 lines Using latest request as basis request Sending to 192.168.44.23 : 5060 (non-NAT) Found no matching peer or user for '192.168.44.23:57704' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 98 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 19 Peer audio RTP is at port 192.168.44.23:17780 Found description format PCMA Found description format G729 Found description format GSM-EFR Found description format GSM Found description format PCMU Found description format CN Capabilities: us - 0x508 (alaw|g729|ilbc), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x2 (gsm), combined - 0x0 (nothing) Looking for 390 in from-sip-external list_route: hop: sip:[EMAIL PROTECTED]:5060 You can see the line Found no matching peer or user for '192.168.44.23:57704' OK, now if I go into the parameters for my trunk and add the line Port=57704 It works!!! Problem is, the port changes. The question then is, where in my Cisco config can I specify the listening (or return) port to 5060 so it does not pick an arbitrary port from the pool? Regards Mark Date: Sun, 22 May 2005 11:10:31 -0400 From: Steve Blair [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Getting a Cisco gateway to work with Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed When you say identify I presume you are trying to get the Cisco to register as a user. To the best of my knowledge it cannot do this. Instead define a peer in sip.conf which is the gateway and place traffic matching this peer into a context that is defined in your extensions.conf file. The Cisco will need dial-peer statements to match inbound dialed digits and forward all matching calls to your Asterisk box. Mark Dutton wrote: Can anyone please help me with sample IOS commands to get a Cisco gateway working properly with Asterisk. I cannot get my Cisco 2801 with BRI interfaces to call into Asterisk. The Cisco identifies itself as sip:[EMAIL PROTECTED] I cannot figure out how to get it to identify as sip:[EMAIL PROTECTED] The gateway works with other SIP servers that don't require authentication, but Asterisk wants it to authenticate, or at least idenitify itself and I cannot work this bit out. If I put in the host address in my sip.conf, I still get a cannot find host 192.168.44.23:random port number, where random port number is actually some random port number. I am at my wits end. Regards Mark --- - ___ 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] Getting a Cisco gateway to work with Asterisk
Can anyone please help me with sample IOS commands to get a Cisco gateway working properly with Asterisk. I cannot getmy Cisco 2801 with BRI interfaces to call into Asterisk. The Cisco identifies itself as sip:[EMAIL PROTECTED] I cannot figure out how to get it to identify as sip:[EMAIL PROTECTED] The gateway works with other SIP servers that don't require authentication, but Asterisk wants it to authenticate, or at least idenitify itself and I cannot work this bit out. If I put in the host address in my sip.conf, I still get a "cannot find host 192.168.44.23:random port number, where random port number is actually some random port number. I am at my wits end. Regards Mark ___ 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] T.38 fax with SIP devices
Hi Jonathon. The boxes are at work, but I am pretty sure the FXO box (6port) is a Micronet SP5050/S and the FXS box (2 port) is the Micronet SP5002/S. http://www.micronet.com.tw I recommend you move to the Digium users forum. I have taken this question there. Not much feeback so far, but it is much better than this old mailing list. Cheers Mark Date: Thu, 7 Apr 2005 10:52:50 -0400 From: Moody [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] T.38 fax with SIP devices To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Hello Mark, I have been working on a similar plan but am still looking for reasonable/tested hardware - can you tell me what devices you are using? Thanks, Jonathon ___ 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] T.38 fax with SIP devices
Right you are Michael. I have some Multitech MVP200s and they do work indeed. Only problem is mine are too old to do SIP. I know Asterisk does not do T.39 but as it only needs to ALLOW the codec when devices are communicating with each other, it can't be too hard to get working. Perhaps the t39fax codec needs to be added to the Asterisk codec list so it knows about it and then it can be added to the allow list in SIP. Mark Date: Thu, 07 Apr 2005 21:17:03 -0700 From: Michael D Schelin [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] T.38 fax with SIP devices To: Scott Wolfe [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello, The Multitech VOIP line supports T38 and I have tested it. It works great. You will need a public IP to make it work. Very expensive though. T38 Is not compatible with Asterisk. ___ 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] T.38 fax with SIP devices
Hi there I have a SIPATA with a fax machine attached and a SIP FXO gateway to the PSTN. When I try to send faxes in either direction, I get nothing but stony silence. I have changed the gateway and the ATA to peer to peer mode to test them and they happily do the T.38 thing and faxes flow. It seems that they initially negotiate a G.729 codec, which is what I want and then when the receiving end detects the fax machine, it wants to re-negotiate and use the t38fax codec. This is the working the Micronet devices use at least. When I put the units into proxy mode and run them through Asterisk, they fail at the negotiation stage. Now I have learned from my dealings with Asterisk and the newsgroup that Asterisk does not do T.38. However, why should it not let devices do T.38? My debug messages from Asterisk don't show it saying no, but the gateways don't wont' setup the T.38 on Asterisk. I have chanded sip.conf to allow=all and there are no explicit rules in the registrations for the gateways. Does anyone have an idea here? For this venture to be truly usable, I have to be able to get FAX working at this basic level. Regards Mark Dutton ___ 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] T.38 fax with SIP devices
I will give this a try Clay, but I don't think it will make a difference. When the gateway detects fax tone, it switches to T.38 and the voice codec becomes irrelevant. I am a relative novice to SIP transactions, but from what I have seen from the debug output of my gateways, the gateway first negotiates a voice codec, in this case g729. This is to be expected as this stage happens as part of the call setup. As soon as either gateway detects fax tone, the debug output of this gateway shows this detection and then it appears to re-invite with the t39fax protocol. When the gateways are in peer to peer mode, this re-invite is successful, but when all invites must be put through the asterisk server, I think Asterisk denies the T39fax protocol. This leads me to the thought, is it hard to tell Asterisk about the protocol so it will accept it? The Micronet gateways see t39fax as a codec, which makes sense and they call it t39fax. Even though t39fax is not strictly a codec, it makes sense that gateways see it as this as for the purposes of a SIP conversation between the gateways, it looks and acts like a codec. More input on this subject is welcome. I can't be the only person who needs fax through SIP devices and T.39 is as old as the hills. I know I could turn off T.39 (actually I can't on the Micronet gateway) and use spandsp, but this is a kludge compared to the rock solid T.39 protocol. Cheers Mark Date: Thu, 7 Apr 2005 09:07:33 -0400 From: Clay Reiche [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T.38 fax with SIP devices To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Mark, Try forcing ulaw on the devices, make a successful ulaw voice call, then try the fax again. Clay From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Dutton Sent: Thursday, April 07, 2005 7:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T.38 fax with SIP devices Hi there I have a SIP ATA with a fax machine attached and a SIP FXO gateway to the PSTN. When I try to send faxes in either direction, I get nothing but stony silence. I have changed the gateway and the ATA to peer to peer mode to test them and they happily do the T.38 thing and faxes flow. It seems that they initially negotiate a G.729 codec, which is what I want and then when the receiving end detects the fax machine, it wants to re-negotiate and use the t38fax codec. This is the working the Micronet devices use at least. When I put the units into proxy mode and run them through Asterisk, they fail at the negotiation stage. Now I have learned from my dealings with Asterisk and the newsgroup that Asterisk does not do T.38. However, why should it not let devices do T.38? My debug messages from Asterisk don't show it saying no, but the gateways don't wont' setup the T.38 on Asterisk. I have chanded sip.conf to allow=all and there are no explicit rules in the registrations for the gateways. Does anyone have an idea here? For this venture to be truly usable, I have to be able to get FAX working at this basic level. Regards Mark Dutton ___ 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] One way speech from H.323 incoming calls, but outgoing calls are OK.
Hi everyone I have successfully compiled and installed OH323 support (finally) into my Asterisk. I want to connect the Asterisk server to our Alcatel OmniPCX Office (OXO)PABX, which has an internal H.323 gateway. I have created the correct dialplans in Asterisk and same in OXO. The OXO only supports G711a G711u G729 and G723.1 codecs. When I call from a SIP phone to OXO using my Grandstream 100 handset with PCMA as the first priority codec, I get perfect speech. When I call from the Alcatel to the Grandstream, I get one way speeche, i.e. I can hear the person on the Alcatel handset, but they can't hear me. Is there any way to debug the connections so as to see what codecs are used in asterisk? I can see all the call setup info with debug on, but I can't see the codec info. It seems strange that the call will only work in one direction as the Alcatel can obviously find a compatible codec when the call is initiated inbound. Any ideas greatly appreciated. Regards Mark Dutton. ___ 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
Subject: Re: [Asterisk-Users] What combination of pwlib and openh323 are required to get Asterisk-oh323 v0.7.1 to compile
Thanks Vamsi I have not been able to locate pwlib - 1.6.6 or openh323 1_13_5. I found the latest versions through sourceforge and I found some older versions on another site, but not these versions. This has been quite frustrating. Anyway, I think by using the asterisk-oh323 branch under channels in the asterisk source tree I will have more luck. At present it seems to compile successfully, but fails linking due to a lib expat, which I have no idea where that comes from. Regards Mark Date: Tue, 8 Mar 2005 13:18:30 +0530 From: Vamsi Pottangi [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] What combination of pwlib and openh323 are required to get Asterisk-oh323 v0.7.1 to compile To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII This worked for me on Fedora Core 3 pwlib - 1_6_6 openh323 - 1_13_5 asterisk-oh323 - 0.7.1 cp pwlib-v1_6_6-src.tar.gz openh323-v1_13_5-src.tar.gz asterisk-oh323-0.7.1.tar.gz /usr/src/ cd /usr/src tar zxf pwlib-v1_6_6-src.tar.gz tar zxf openh323-v1_13_5-src.tar.gz tar zxf asterisk-oh323-0.7.1.tar.gz - Set Environment variables PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib cd /usr/src/pwlib ./configure make opt cd /usr/src/openh323 ./configure -- Remove the line 433 (:protected) in /usr/src/openh323/include/gkserver.h else you would get the below error during compilation /usr/src/openh323/include/gkserver.h:434: error: `virtual H323Transaction::Response H323GatekeeperRRQ::OnHandlePDU()' is protected -- make opt cd /usr/src/asterisk-oh323-0.7.1 Edit makefile and set the paths/options according to your system. Type make to build the oh323wrap library and the ASTERISK OH323 channel driver. Type make install to install the binaries. This will also install a sample configuration file, if there isn't one. Hope this of help to you Cheers, ~Vamsi On Tue, 8 Mar 2005 11:41:19 +0800, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi there I have Asterisk running beautifully on our test server. Over the past few days I have been tearing my hair out trying to compile various versions of asterisk-oh323 on various versions of pwlib and openh323. pwlib is now up to 1.8.3 and openh323 is now 1.15.2 stable. asterisk-oh323 is currently 0.7.1 I have tried these three with many errors. I have tried 0.7.1 with pwlib 1.5.2 and openh 1.12.2 with no luck. I have tried asterisk-oh323 1.5.10 with pwlib 1.5.2 and openh323 and I still get errors. From the mailing list I have gleaned that this version of asterisk-openh323 won't work with the latest asterisk anyway, yet the readme in asterisk-oh323 says to use this version with the aforementioned versions of pwlib and openh323. I can't find the versions of pwlib and openh323 recommended in the asterisk-oh323-0.7.1 readme. The pwlib and openh323 projects always build without error. Asterisk built without errors and most everythings else. I am running a very basic Fedora Core 2 installation. What I would like to know is what is the recommended known good combination to use of asterisk-oh323, pwlib and oh323. Once I have a combination that should work, I can then ask more intelligent questions on how to get it to build properly if I still have errors. Help greatly appreciated. Regards Mark Dutton ___ 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 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] What combination of pwlib and openh323 are
Thanks Pete I never looked in the channels directory of the source tree. Doh! I have been using a separate source tree for asterisk-oh323. On the strength of that, I modified the Makefile in the asterisk channels/h323 directory and it compiled immediately. The only problem now is with the linker. It is looking for a library called expat. I don't suppose you know where this is from? Regards Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Illmayer Sent: Tuesday, 8 March 2005 1:31 PM To: mat Cc: Mark Dutton Subject: [Asterisk-Users] What combination of pwlib and openh323 are Hi Mark Funny you should ask this question, I just spent yesterday integrating building asterisk with h323 support to connect to a Cisco call agent.I cant say if it will work for you but it compiles and loads nicely ! I will be testing this evening # cd /root # wget http://www.voxgratia.org/releases/pwlib-Pandora_release-src-tar.gz # wget http://www.voxgratia.org/releases/openh323-Pandora_release-src-tar.gz # tar zxvf pwlib-Pandora_release-src-tar.gz # tar zxvf openh323-Pandora_release-src-tar.gz # cd /root/pwlib # ./configure make opt make install # cd /root/openh323 # ./configure make opt make install # echo '/root/pwlib/lib' /etc/ld.so.conf # may not be req'd # echo '/root/openh323/lib' /etc/ld.so.conf# may not be req'd # /sbin/ldconfig That gets pwlib and openh323 installed. Are you going to install ztdummy ? You will need to do this if you dont have a digium card installed.. If you need more help, drop me a message # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login (when prompted for a password, enter 'anoncvs') # cvs -z4 checkout -D '2004-12-22' zaptel libpri asterisk That gets the cvs version of Asterisk that works :-)You want this **specific** version of asterisk # cd /usr/src/asterisk/channels/h323 # make # cd /usr/src/asterisk # make # make install That will build the h323 support for asterisk. If I can help any furthur, drop me a mail.. I spent most of yesterday sorting it and was rewarded at midnight with a loading asterisk :-) Regards..Pete Date: Tue, 8 Mar 2005 11:41:19 +0800 From: [EMAIL PROTECTED] Subject: [Asterisk-Users] What combination of pwlib and openh323 are required to get Asterisk-oh323 v0.7.1 to compile To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi there I have Asterisk running beautifully on our test server. Over the past few days I have been tearing my hair out trying to compile various versions of asterisk-oh323 on various versions of pwlib and openh323. pwlib is now up to 1.8.3 and openh323 is now 1.15.2 stable. asterisk-oh323 is currently 0.7.1 I have tried these three with many errors. I have tried 0.7.1 with pwlib 1.5.2 and openh 1.12.2 with no luck. I have tried asterisk-oh323 1.5.10 with pwlib 1.5.2 and openh323 and I still get errors. From the mailing list I have gleaned that this version of asterisk-openh323 won't work with the latest asterisk anyway, yet the readme in asterisk-oh323 says to use this version with the aforementioned versions of pwlib and openh323. I can't find the versions of pwlib and openh323 recommended in the asterisk-oh323-0.7.1 readme. The pwlib and openh323 projects always build without error. Asterisk built without errors and most everythings else. I am running a very basic Fedora Core 2 installation. What I would like to know is what is the recommended known good combination to use of asterisk-oh323, pwlib and oh323. Once I have a combination that should work, I can then ask more intelligent questions on how to get it to build properly if I still have errors. Help greatly appreciated. Regards Mark Dutton ___ 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 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
