RE: [Asterisk-Users] Zombie SIP channels
Hi, -Original Message- Does anyone know how to kill a zombie channel? Here is what I see on a show channels: -- show channels Channel (ContextExtensionPri ) State Appl. Data SIP/frontdesk-72c7 (customercontext 1 ) Up Bridged Call SIP/frontdesk-0461ZOMBIE SIP/frontdesk-0461ZOMBIE (customercontext 100 1 ) Ring Dial SIP/frontdesk|20|t 2 active channel(s) -- No one is on a call - how can I get rid of this without restarting asterisk? This was an issue in older versions of asterisk. It would help if you could tell us what setup you are running. If this is infact your problem too, a simple update of your asterisk to 1.0.3 or later will help. Florian ___ 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] Manager API - Call Transfer/Blind Transfer
Good morning folks, I am quite new to Asterisk but have successfully set it up with some BRI lines, Cisco 7940/7960, queues, voicemail, XML stuff and the flash operator panel. When playing with the manager API to get some stuff integrated within our systems, I stumbled across the Redirect command and the way it is working. Generally, there is a difference between Transfers and Blind Transfers of calls, right? With transfers the transfering one calls the destination while the original caller is on hold. Then, as soon as the transfering one hangs up, the call will be redirected. With blind transfers, the call gets immediately redirected. The transfering one gets hung up and the original caller hears the ring. This is what the manager API's redirect command is working like. It simply redirects the channel to another extension, correct? The question is, how to achieve what I call a transfer? Any way to implement the hold - call-destination - hangup - redirect behaviour through the manager API? I thought about doing something within the context in the extensions definitions but I feel like this would not be working. If this has been asked before - sorry :-) Simply point me to the right month in the archives, I will do the rest ;-) Cheers Sascha ___ 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] CallPickup from SIP phone
So I'm having trouble getting call-pickup working. Got a few different SIP phones (cisco 7940's and SPA-841s) all with pickupgroup=0 in sip.conf. I can't seem to get it working. This *is* possible from SIP phones, right? Do I need to add anything to my dial-plan? Yes, it works fine from my 7960. On the 7960, I pick up ringing calls by pressing *8#. If that does not work for you, then ensure you don't have any extensions.conf entries that override *8, that all phone def's in sip.conf that you want to be able to pickup include something like callgroup=2, and the phone def's that you want to use the *8# have the pickupgroup=2. ___ 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] why asterisk is replying 404 Not Found
[3000] type=friend dtmfmode=INFO insecure=yes canreinvite=no auth=plaintext host=dynamic allow=ulaw [2000] type=friend dtmfmode=INFO insecure=yes canreinvite=no auth=plaintext host=dynamic allow=ulaw i have declared these two users 3000 and 2000. they are registering successfully. problem is that when i am clling for 3000 to 2000. it is replying me 404 Not found and printing (found user '3000' looking for 2000 in default). and when i am calling from 2000 to 3000. it is replying me 404 not found and printing (found user '2000' looking for 3000 in default). can any one tell me what is the problem. i dont know what is problem. i have just compilied my asterisk(stable) using LAN. thanks Found user '3000' Looking for 2000 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.109;branch=z9hG4bKc0a8006d0131c9b14209a71a2c53e2990004 From: rootsip:[EMAIL PROTECTED];tag=4122263641472432750 To: sip:[EMAIL PROTECTED];tag=as2f7c300d Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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] Please share the experience on VoIP phones heavyusing.
Hi, cisco's phones are VoIP only polycom build (video-) conferencing devices. One Cisco model (7930 I thnk) is a polycom in disguise. The code is not 'cisco-like' (at least the version I had. Both brands make very good quality equipments. Good sound, good support, ... Regards, Shaoul Jacobson VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] -Original Message- From: Sergey Kuznetsov [mailto:[EMAIL PROTECTED] Sent: jeudi 10 février 2005 4:15 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Please share the experience on VoIP phones heavyusing. Hi there, Does someone can share the experience with Cisco and Polycom Phones? ___ 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] why asterisk is replying 404 Not Found
What does your extensions.conf file look like? http://www.voip-info.org/wiki-Asterisk+config+extensions.conf Dan Kamran Ahmad wrote: [3000] type=friend dtmfmode=INFO insecure=yes canreinvite=no auth=plaintext host=dynamic allow=ulaw [2000] type=friend dtmfmode=INFO insecure=yes canreinvite=no auth=plaintext host=dynamic allow=ulaw i have declared these two users 3000 and 2000. they are registering successfully. problem is that when i am clling for 3000 to 2000. it is replying me 404 Not found and printing (found user '3000' looking for 2000 in default). and when i am calling from 2000 to 3000. it is replying me 404 not found and printing (found user '2000' looking for 3000 in default). can any one tell me what is the problem. i dont know what is problem. i have just compilied my asterisk(stable) using LAN. thanks Found user '3000' Looking for 2000 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.109;branch=z9hG4bKc0a8006d0131c9b14209a71a2c53e2990004 From: rootsip:[EMAIL PROTECTED];tag=4122263641472432750 To: sip:[EMAIL PROTECTED];tag=as2f7c300d Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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] Cisco 7960 Beating a Dead Horse
Doug Lytle wrote: Keep in mind, you need to include both the P003-07-3-00 and P0S3-07-3-00 in the SIPDefault.cnf and OS79XX.txt You need the P003-07-3-00 in OS79XX.TXT as it contains the application loader and P0S3-07-3-00 in the SIP(Default|MAC).cnf as it contains the actual sip firmware. Stefan ___ 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] ISDN in Spain
We have a Junghanns 4 port card in one * and two HFC-S in another in Spain and both are working Ok. ISDN in Spain is called RDSI but good luck in dealing with Telefonica. We have seen some framing errors on the system with the 4 port card but it does not seem to be causing and problems. Stuart -Original Message- From: Patrick[EMAIL PROTECTED] Sent: 10/02/05 07:28:25 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ISDN in Spain On Wed, 2005-02-09 at 18:07 +0100, Remco Barende wrote: Hi list! Sorry for this slightly off-topic message but does anybody know if the standard for ISDN BRI is the same in Spain as it is in the rest of Europe (or the Netherlands). Will a standard HFC-S card work? Afaik Telefonica offers plain standard ISDN-2 and PRI service so I would say yes. Regards, Patrick ___ 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 [Message truncated. Tap Edit-Mark for Download to get remaining portion.] ___ 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] why asterisk is replying 404 Not Found
[3000] type=friend dtmfmode=INFO insecure=yes canreinvite=no auth=plaintext host=dynamic allow=ulaw [2000] type=friend dtmfmode=INFO insecure=yes canreinvite=no auth=plaintext host=dynamic allow=ulaw i have declared these two users 3000 and 2000. they are registering successfully. problem is that when i am clling for 3000 to 2000. it is replying me 404 Not found and printing (found user '3000' looking for 2000 in default). and when i am calling from 2000 to 3000. it is replying me 404 not found and printing (found user '2000' looking for 3000 in default). It may help you a lot to add context=from-sip in each of the two above. Then in your extensions.conf define this context with the appropriate exten = ... entries for your dialplan. It would appear you are getting caught with not understanding the use of contexts, and your existing sip.conf entries are defaulting to the default context and you don't see/know that. So, given the above, your extensions.conf would have something like this in in: [from-sip] exten = 3000,1,Dial(SIP/3000,15,r) exten = 3000,2,Voicemail(u3000) exten = 3000,102,Voicemail(b3000) exten = 3000,103,Hangup exten = 2000,1,Dial(SIP/2000,15,r) exten = 2000,2,Voicemail(u2000) exten = 2000,102,Voicemail(b2000) exten = 2000,103,Hangup Hope that helps... ___ 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] Re: Asterisk Compile Problem on Red Hat 9 solved
Thanks Noah and Marco, this info will keep me busy for a while... ___ 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] Re: why asterisk is replying 404 Not Found
[default]
;
; By default we include the demo. In a production
system, you
; probably don't want to have the demo there.
;
include = demo
exten = 3000,1,Dial(SIP/${EXTEN})
exten = 2000,1,Dial(SIP/${EXTEN})
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RE: [Asterisk-Users] reboot polycom 1.4.1
Do you have the voIpProt.SIP.specialEvent.checkSync.alwaysReboot set to 1 in the Polycom sip.cfg. I have handsets running with 1.4.1 and the above set to 1 and the reboot script works fine. Stuart -Original Message- From: Richard[EMAIL PROTECTED] Sent: 10/02/05 04:31:42 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: [Asterisk-Users] reboot polycom 1.4.1 Hi, I have a polycom reboot script which sends a NOTIFY with check-sync. It worked fine with 1.3.4. After I upgrade to 1.4.1, it stopped working. Anyone has the same problem? Thanks, Richard ___ 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 [Message truncated. Tap Edit-Mark for Download to get remaining portion.] ___ 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] SIP proxies Asterisk ?
Hello, We hve been trying to make Asterisk work with SIP proxies with no success. Is there support for SIP proxies in Asterisk in the latest versions? Best regards, Vlasis. ___ 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] ISDN in Spain
Hi, Other basic settings that must be right: - framing with crc4 or not - clock source (mostly provided by the 'provider') cabling: a straight cable will most probably be ok but I have seen some strange settings over the time LAYER 1 or 2 problems are often overloocked Good luck Shaoul Jacobson VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] -Original Message- From: Patrick[EMAIL PROTECTED] Sent: 10/02/05 07:28:25 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ISDN in Spain On Wed, 2005-02-09 at 18:07 +0100, Remco Barende wrote: Hi list! Sorry for this slightly off-topic message but does anybody know if the standard for ISDN BRI is the same in Spain as it is in the rest of Europe (or the Netherlands). ___ 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] DTMF Payload type
Hei!! 101 type in Snom is correct. Don't change that. Rennes Michael Di Martino wrote: To All I am using a SNOM 190 w/Asterisk server. Here is my sip.conf [7501] type=friend context=external username=7501 callerid=Telx 7501 7501 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 My question is this. With above settings in my sip.conf specially dtmfmode=rfc2833 What should my DTMF Payload Type: be set to on my SNOM 190 phone. Currently it is set to 101. Should it be set to rfc2833? Regards, Michael DiMartino ___ 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] reboot polycom 1.4.1
hello, could you help me ? I try to set up IM an presence, it's ok with msn 4.7 but asterisk reply 407 error. Does asterisk support IM and presence ? Regards harry --- Stuart Hirst [EMAIL PROTECTED] a écrit : Do you have the voIpProt.SIP.specialEvent.checkSync.alwaysReboot set to 1 in the Polycom sip.cfg. I have handsets running with 1.4.1 and the above set to 1 and the reboot script works fine. Stuart -Original Message- From: Richard[EMAIL PROTECTED] Sent: 10/02/05 04:31:42 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: [Asterisk-Users] reboot polycom 1.4.1 Hi, I have a polycom reboot script which sends a NOTIFY with check-sync. It worked fine with 1.3.4. After I upgrade to 1.4.1, it stopped working. Anyone has the same problem? Thanks, Richard ___ 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 [Message truncated. Tap Edit-Mark for Download to get remaining portion.] ___ 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 Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.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] Softphone..easy to use ?
Hello Guys Im currently using Asterisk on a Red Hat box with an ISDN Card on it.. Works perfect. Now I like to forward a call to a softphone. (from my asterisk menu) Im very new to this, so unsure what softphone I should use ? Can anybody provide me a link with a good Softphone ? (for windows) How is the quality on software? Do you head any different between softphone and regular phone if a person calls you ? And is it hard to forward a call from asterisk to a softphone ? Thank you for the help Regards from Switzerland Matt ___ 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] Problem using TDM400P FXS card
On Wed, 2005-02-09 at 15:43 +0100, cereal killer wrote: I decided to take another approach to se if it s not my card that is faulty ; removed the X101 card , only let the TDM , I put the FXS module on another port (maybe the first is bad). Here are the infos : /etc/asterisk/zapata.conf [channels] language=fr context=cartezap ; context with a few extensions signalling=fxo_ks echocancel=yes busydetect=yes channel = 2 What is the value of immediate ? You should try this: [channels] language=fr context=cartezap ; context with a few extensions signalling=fxo_ks echocancel=yes busydetect=yes immediate=no channel = 2 Which, I think, means don't immediately jump into the dialplan (s exten) instead, provide dialtone, and wait for them to press buttons. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] reboot polycom 1.4.1
I don't believe * supports this yet but I think its being worked on. No doubt some else will know more about this. -Original Message- From: harry gaillac[EMAIL PROTECTED] Sent: 10/02/05 09:57:21 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] reboot polycom 1.4.1 hello, could you help me ? I try to set up IM an presence, it's ok with msn 4.7 but asterisk reply 407 error. Does asterisk support IM and presence ? Regards harry --- Stuart Hirst [EMAIL PROTECTED] a écrit : Do you have the voIpProt.SIP.specialEvent.checkSync.alwaysReboot set to 1 in the Polycom sip.cfg. I have handsets running with 1.4.1 and the above set to 1 and the reboot script works fine. Stuart -Original Message- From: Richard[EMAIL PROTECTED] Sent: 10/02/05 04:31:42 To: 'Asterisk Users Mailing List - [Message truncated. Tap Edit-Mark for Download to get remaining portion.] ___ 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] 7940 VM DTMF not detecting
Hi all, I have a 7940 running the latest SIP firmware (V7 - thanks Doug Lytle for the tip on the V7 firmware upgrade!). Its almost working perfectly - I can make calls either though my local PSTN or over VOIP but for some reason if I dial my voicemail (which is mapped fine to the VM button on the telephone) it doesn't detect my DTML keypresses so when I press 1 for new messages it just ignores it. Otherwise DTMF dialing is working perfectly. Does anyone know why this is happening? Thanks very much, Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.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] Asterisk and Fedora Core 3
Hi guys I'm new to this list and I imagine this question has been asked before, so feel free to point me to the correct references. My question is, how do you install asterisk on Fedora Core 3, with all rpm updates, seeing as there is no kernel-source rpm anymore? Thanks for any advice. -- _/_/_/_/ _/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/ _/ Bill Maidment Maidment Enterprises Pty Ltd Unless you are named Alfred E. Newman, you may read only the odd numbered words (every other word beginning with the first) of the message above. If you have violated that, then you hereby owe the sender AU$10 for each even numbered word you have read. Adapted from Stupid Email Disclaimers (see http://www.goldmark.org/jeff/stupid-disclaimers/) ___ 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] Softphone..easy to use ?
Im very new to this, so unsure what softphone I should use ? Can anybody provide me a link with a good Softphone ? (for windows) http://www.marccharbonneau.com/asterisk/mediaxphone.php Supports gsm, ulaw, alaw see also : http://www.voip-info.org/tiki-index.php?page=Asterisk%20IAX%20clients http://www.voip-info.org/wiki-VOIP+Phones hth ___ 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] Detect hangup
Hi all How do one "record" the tone that a pbx gives for hangup, so that one can use that in indications.conf? thanks Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'___ 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] SIP jitter?
See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532 There isn't even any code for SIP yet. However the iax integration works wonders for a link with just a bit of packet loss and jitter. Voice conversations are nice and crisp and without the pops associated with lost packets or growth of the jitter buffer. Is there a reason why this isn't in HEAD? roy ___ 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] SIP proxies Asterisk ?
Vlasis Hatzistavrou wrote: Hello, We hve been trying to make Asterisk work with SIP proxies with no success. Is there support for SIP proxies in Asterisk in the latest versions? A lot of people use Asterisk with SIP proxys. What is your problem, give us a bit more information. /Olle ___ 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] Asterisk and Fedora Core 3
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 Use the wiki luke. -Original Message- From: Bill Maidment [mailto:[EMAIL PROTECTED] Sent: Thursday, February 10, 2005 5:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk and Fedora Core 3 Hi guys I'm new to this list and I imagine this question has been asked before, so feel free to point me to the correct references. My question is, how do you install asterisk on Fedora Core 3, with all rpm updates, seeing as there is no kernel-source rpm anymore? Thanks for any advice. -- _/_/_/_/ _/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/ _/ Bill Maidment Maidment Enterprises Pty Ltd Unless you are named Alfred E. Newman, you may read only the odd numbered words (every other word beginning with the first) of the message above. If you have violated that, then you hereby owe the sender AU$10 for each even numbered word you have read. Adapted from Stupid Email Disclaimers (see http://www.goldmark.org/jeff/stupid-disclaimers/) ___ 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
[Asterisk-Users] Re: why asterisk is replying 404 Not Found
thanks it is register and receiving the invite. some time my user agent (i am sjphone) is sending invalid address in his contact and SDP. then i try to call form another ua it i transmitting invite to invalid address. __ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.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] Cisco7960/SCCP Transfer Help?
I have a Cisco 7960 running 7.2 of their SCCP image; I am running Asterisk 1.0.5 and using the latest Sourceforge version of SCCP2. When I make a call (or receive one) the Transfer softkey does not show up - as a matter of fact only 2 softkeys show up (redial something else), but those even are not active. On a 7960 running SIP the Transfer and other buttons do show up and are active. What am I missing as far as getting the Transfer button to show up on my SCCP phone? Additionally, the # does not work when talking on an outside line to do a transfer that way; it only works when talking to another internal phone I've intercommed. Help would be very much appreciated :-). Thanks, Bruce -- Bruce M. Himebaugh Himebaugh Consulting, Inc. 330/493-9700 http://www.hcd.net Computer consulting, software/web development systems integration CanNet Internet Services, Inc. 330/484-2260 http://www.cannet.com Providers of World-Wide Connectivity Get Connected ... Stay Connected! ___ 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] Cisco7960/SCCP Transfer Help?
If you select more there Trnsfer and BlndXfer will be displayed BlndXfer for Blind transfer Trnsfer for Confirm transfer This is on 7960 On Thu, 2005-02-10 at 15:09, [EMAIL PROTECTED] wrote: I have a Cisco 7960 running 7.2 of their SCCP image; I am running Asterisk 1.0.5 and using the latest Sourceforge version of SCCP2. When I make a call (or receive one) the Transfer softkey does not show up - as a matter of fact only 2 softkeys show up (redial something else), but those even are not active. On a 7960 running SIP the Transfer and other buttons do show up and are active. What am I missing as far as getting the Transfer button to show up on my SCCP phone? Additionally, the # does not work when talking on an outside line to do a transfer that way; it only works when talking to another internal phone I've intercommed. Help would be very much appreciated :-). Thanks, Bruce -- Bruce M. Himebaugh Himebaugh Consulting, Inc. 330/493-9700 http://www.hcd.net Computer consulting, software/web development systems integration CanNet Internet Services, Inc. 330/484-2260 http://www.cannet.com Providers of World-Wide Connectivity Get Connected ... Stay Connected! ___ 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] Zombie SIP channels
Thanks for the feedback! Running CVS-v1-0-11/12/04 (stable) on Fedora Core 1 with Cisco 7960G's. Asterisk server is on public IP and Cisco 7960G is at client location NAT-ed behind a Cisco soho91-k9 with nine other Cisco 7960G's (each phone has registration expiring every 120 seconds). Here is excerpt from sip.conf [general] disallow=all allow=ulaw port=5060 context=incoming maxexpirey=3600 defaultexpirey=300 canreinvite=no tos=reliability srvlookup=yes videosupport=no dtmfmode=inband nat=yes insecure=very [frontdesk] context=customer type=friend username=frontdesk secret=password host=dynamic canreinvite=no [EMAIL PROTECTED] nat=yes qualify=yes callerid=Front Desk 100 accountcode=customer amaflags=billing This is the first time I have seen this so it does not appear to happen too often. Obviously would rather not upgrade if possible has everything seems running fine. But good to know that if it becomes a problem, I can try upgrading to 1.0.3 or later. Thanks! Pedro On Thu, 10 Feb 2005 09:19:45 +0100, Florian Overkamp [EMAIL PROTECTED] wrote: Hi, -Original Message- Does anyone know how to kill a zombie channel? Here is what I see on a show channels: -- show channels Channel (ContextExtensionPri ) State Appl. Data SIP/frontdesk-72c7 (customercontext 1 ) Up Bridged Call SIP/frontdesk-0461ZOMBIE SIP/frontdesk-0461ZOMBIE (customercontext 100 1 ) Ring Dial SIP/frontdesk|20|t 2 active channel(s) -- No one is on a call - how can I get rid of this without restarting asterisk? This was an issue in older versions of asterisk. It would help if you could tell us what setup you are running. If this is infact your problem too, a simple update of your asterisk to 1.0.3 or later will help. Florian ___ 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] SIP proxies Asterisk ?
Hi Olle, When we accept calls from a SIP proxy without regitration from either side, but with only an INVITE message, the calls fail. If we set the remote proxy to send us the calls by proxying both RTP signaling, then there is no problem. So, we concluded that Asterisk doesn't like it when signaling and RTP come from different IP addresses. Is there a setting on Asterisk which could allow this? I can provide packet captures if you want. Best regards, Vlasis. Olle E. Johansson wrote: Vlasis Hatzistavrou wrote: Hello, We hve been trying to make Asterisk work with SIP proxies with no success. Is there support for SIP proxies in Asterisk in the latest versions? A lot of people use Asterisk with SIP proxys. What is your problem, give us a bit more information. /Olle ___ 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
[Asterisk-Users] A: ISDN / E1-PRI - fax problems - Receiving and setting of Service Indicator (SIN) / Bearer Capability (BC) / High Level Compatibility (HLC) / Low Level Compatibility (LLC)
hi, i made a patch which allows the forwarding and the setting of the Bearer Capability ID during the ISDN SETUP phase. this solves several problems (primarily faxing) with SIN (german: Dienstekennung) and asterisk. http://bugs.digium.com/bug_view_page.php?bug_id=0003547 Frank Sautter wrote: i have the problem that i'm not able to set and receive the Service Indication (SIN) from our E1-PRI and from our ericsson BP250. regards frank ___ 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] Zombie SIP channels
Hi, -Original Message- This is the first time I have seen this so it does not appear to happen too often. Obviously would rather not upgrade if possible has everything seems running fine. But good to know that if it becomes a problem, I can try upgrading to 1.0.3 or later. If my memory serves me correctly, this is the issue: http://bugs.digium.com/bug_view_page.php?bug_id=0002938 It's a two line fix, so if you want you can easily verify and apply manually so you don't have to introduce any other new code. Florian ___ 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] A working config for For FX100P Cards in United Kingdom ?
Hi Does anyone have a working config for a FX100P card working on a U.K. phone line ? If so could you please post your fzaptel.conf and zapata.conf files or at least share whether you have configured for Loopstart, Groundstart or Kewlstart signalling. Thanks Graeme Brown ___ 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] Cisco7960/SCCP Transfer Help?
There is no more button - there is when not on a call, but when on the call only the redial and something else is shown, but not active - no more button :-(. The phone is a 7960G. Also, I made a mistake on the software version in the phone - it is 7.1(2.0) (not 7.2). Thanks, Bruce -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Altus Snyman Sent: Thursday, February 10, 2005 8:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco7960/SCCP Transfer Help? If you select more there Trnsfer and BlndXfer will be displayed BlndXfer for Blind transfer Trnsfer for Confirm transfer This is on 7960 On Thu, 2005-02-10 at 15:09, [EMAIL PROTECTED] wrote: I have a Cisco 7960 running 7.2 of their SCCP image; I am running Asterisk 1.0.5 and using the latest Sourceforge version of SCCP2. When I make a call (or receive one) the Transfer softkey does not show up - as a matter of fact only 2 softkeys show up (redial something else), but those even are not active. On a 7960 running SIP the Transfer and other buttons do show up and are active. What am I missing as far as getting the Transfer button to show up on my SCCP phone? Additionally, the # does not work when talking on an outside line to do a transfer that way; it only works when talking to another internal phone I've intercommed. Help would be very much appreciated :-). Thanks, Bruce -- Bruce M. Himebaugh Himebaugh Consulting, Inc. 330/493-9700 http://www.hcd.net Computer consulting, software/web development systems integration CanNet Internet Services, Inc. 330/484-2260 http://www.cannet.com Providers of World-Wide Connectivity Get Connected ... Stay Connected! ___ 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 ___ 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] SIP jitter?
Yes, It's untested and unfinished and touches the core of asterisk. (maybe causing massive amounts of deadlocks). zoa. Roy Sigurd Karlsbakk wrote: See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532 There isn't even any code for SIP yet. However the iax integration works wonders for a link with just a bit of packet loss and jitter. Voice conversations are nice and crisp and without the pops associated with lost packets or growth of the jitter buffer. Is there a reason why this isn't in HEAD? roy ___ 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 signature.asc Description: OpenPGP digital signature ___ 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] Asterisk and Fedora Core 3
Eric Rees wrote: http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 Use the wiki luke. Thanks. I'm on my feet now :-) -- _/_/_/_/ _/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/ _/ Bill Maidment Maidment Enterprises Pty Ltd Unless you are named Alfred E. Newman, you may read only the odd numbered words (every other word beginning with the first) of the message above. If you have violated that, then you hereby owe the sender AU$10 for each even numbered word you have read. Adapted from Stupid Email Disclaimers (see http://www.goldmark.org/jeff/stupid-disclaimers/) ___ 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] Zombie SIP channels
What is odd is no meetme is being used. But may be related - thanks! Pedro On Thu, 10 Feb 2005 14:37:31 +0100, Florian Overkamp [EMAIL PROTECTED] wrote: Hi, -Original Message- This is the first time I have seen this so it does not appear to happen too often. Obviously would rather not upgrade if possible has everything seems running fine. But good to know that if it becomes a problem, I can try upgrading to 1.0.3 or later. If my memory serves me correctly, this is the issue: http://bugs.digium.com/bug_view_page.php?bug_id=0002938 It's a two line fix, so if you want you can easily verify and apply manually so you don't have to introduce any other new code. Florian ___ 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] SIP jitter?
joachim wrote: Yes, It's untested and unfinished and touches the core of asterisk. (maybe causing massive amounts of deadlocks). So? That's what CVS-HEAD is there for. ___ 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] Asterisk and Sipura SPA-841 SIP phones
Nothing to do with your question, but by any chance, when you plugged the phone into the wall did you hear a dialtone or is this something generated by 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] asterisk@home scary log
Hi everybody, I'm testing [EMAIL PROTECTED] 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log and saw this Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) Feb 9 20:30:07 asterisk1 sendmail[10077]: j1A1U7CY010077: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30068, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7Ns010089 Message accepted for delivery) Feb 9 20:30:17 asterisk1 sendmail[10094]: j1A1U7Ns010089: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30348, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? thanks jl ___ 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] sample REGEX's for astcc
Jason Kawakami wrote: So I have a route with [1-9][0-9][0-9][1-9][0-9]* as a base route that should match NXXNX. Right? wrong! N is 2 to 9, not 1 to 9, so these are not the same. Try [2-9][0-9][0-9][2-9][0-9]* Don Pobanz ___ 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] Re: polycom soundpoint ip 300
I'm able to register sip friends with asterisk but i wish to use presence and instant messaging. asterisk sent back Proxy Authentication required 407 when SUBSCRIBE is sent to asterisk. I don't think you'll have much success. I don't think asterisk can do that. As far as I know, asterisk does not do instant messaging. You can check on the feature list and with the developers, but I'm pretty sure that's not possible. See the wiki here: http://www.voip-info.org/wiki-Asterisk+presence+jtodd harry --- Noah Miller [EMAIL PROTECTED] a écrit : Hi Harry - Can you get SUBSCRIBE registration ?? I don't know what SUBSCRIBE registration is, but looking at your sip.conf, there's a couple of things I would change: [100] type=friend username=100 secret=100 fromuser=100 ; Take this out - it's only needed when you want certain types of sip proxies are trying to ; register to this peer - not normally needed for asterisk. host=dynamic context=sip dtmfmode=rfc2833 ; I'd use inband here. I've tried rfc2833 here, too, but it doesn't seem to work as well as inband progressinband=no ; You don't really need this, and I think it doesn't make sense if you're doing rfc2833 dtmfmode On the Polycom side, you should use the following info: Address: 100 Auth User ID: 100 Auth Password: 100 Other than the SIP server address, these are the only important numbers on the Polycom. - Noah --- Noah Miller [EMAIL PROTECTED] a écrit : Hi Harry - I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. What version of the SIP firmware are you using? I've had success with 1.3.0, 1.3.1, 1.3.4, and 1.4.1. My sip.conf entries for my Polycom phones look like this: [12] type=friend username=12 secret=12 callerid=12 host=dynamic dtmfmode=inband context=no-callwaiting [EMAIL PROTECTED] disallow=all allow=ulaw Are you configuring directly on the phone, or using an FTP or TFTP server? Anybody could help me to configure Asterisk in order to set instant message and presence ? To the best of my knowledge the Presence feature of the Polycom phones does not work with Asterisk. I believe it only works with other IM clients. Hope this helps! Noah Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.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] Configuring Asterisk
Hey list, I'm having problems to get running *. I don't have any digium hardware yet. I just want to perfrom some tests using SIP. I compiled asterisk and zaptel with ztdummy enabled on Fedora Core 3. When I try to start ztdummy I get the following message: localhost# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line0: Unable to open master device /dev/zap/ctl 1 error(s) detected FATAL: Error running install command for ztdummy I have look into /dev directory and there is no such directory called zap, but there is a file called zapctl. How can I fix it? Where can I find information about all the configurations I need to get * running using SIP? I have the vm1-draft1.pdf but I think it's not clear enough (where I should do all that configurations?). Thanks... -- -DdC ___ 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] Configuring Asterisk
Try README.udev in the zaptel src directory.. -Original Message- From: Daniel del Castillo [mailto:[EMAIL PROTECTED] Sent: Thursday, February 10, 2005 8:13 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Configuring Asterisk Hey list, I'm having problems to get running *. I don't have any digium hardware yet. I just want to perfrom some tests using SIP. I compiled asterisk and zaptel with ztdummy enabled on Fedora Core 3. When I try to start ztdummy I get the following message: localhost# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line0: Unable to open master device /dev/zap/ctl 1 error(s) detected FATAL: Error running install command for ztdummy I have look into /dev directory and there is no such directory called zap, but there is a file called zapctl. How can I fix it? Where can I find information about all the configurations I need to get * running using SIP? I have the vm1-draft1.pdf but I think it's not clear enough (where I should do all that configurations?). Thanks... -- -DdC ___ 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
[Asterisk-Users] Tormenta 2 Card number rotary switch
Hello, I had two quad-E1 Tormenta2 cards in my system and yesterday i had to replace the CPU. I noticed the rotary switch on each card that sets the card number or ID... Both switches were set up on 0 and everything worked fine. Should i set one card to 0 and the other one to 1, or... if it works, don't fix it! ? I had some nightmares with IDs on Dialogic hardware with Bayonne and i don't want to go back to that age :( . Sorry if this is a stupid question, i'm newbie in working with these cards. Thanks, Calin. ___ 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] Please share the experience on VoIP phones heavy using.
We buy from both Graybar and Vibes. Not sure but think it took a couple weeks to get phones RMAd. I normally stock my own spares so do not track too close. On Feb 9, 2005, at 10:14 PM, Sergey Kuznetsov wrote: Jerry, Thanks a lot for the feedback! By the way, how long did it take to replace the faulty 10% of phones by RMA? What company did you use to buy it from? Jerry wrote: On Feb 9, 2005, at 9:14 PM, Sergey Kuznetsov wrote: Hi there, Does someone can share the experience with Cisco and Polycom Phones? How rock solid are they? And who will win in sound quality contest? I heard that Cisco phones is a Polycom replicas with changed design. Is that true? What else phones is better to implement to the medium sized business? The rock solid stability and superb sound quality is a must. Both have excellant sound. I think the Polycom speakerphone is a bit better. We are using mostly Polycom these days and our customers love them. My only issue is they do seem to have about a 10% failure rate within 90 days. After that they are solid - so far. They are also less expensive than the Cisco's and seem to have a better feature set and better control of their configs and buttons. I do like the layer 2 troubleshooting capabilities of the Ciscos as the Polycom seem to have no capabilities that I can find. I do not think the Cisco is any kind of a Polycom copy. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 ext. 37 Mobile phone: (647) 287-8448 ___ 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 -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ 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
[Asterisk-Users] Re: polycom soundpoint ip 300
I mean IM based sip with RFC 3428 not AIM,Jabber,... and rfc3265 SUBSCRIBE/NOTIFY harry --- Noah Miller [EMAIL PROTECTED] a écrit : I'm able to register sip friends with asterisk but i wish to use presence and instant messaging. asterisk sent back Proxy Authentication required 407 when SUBSCRIBE is sent to asterisk. I don't think you'll have much success. I don't think asterisk can do that. As far as I know, asterisk does not do instant messaging. You can check on the feature list and with the developers, but I'm pretty sure that's not possible. See the wiki here: http://www.voip-info.org/wiki-Asterisk+presence+jtodd harry --- Noah Miller [EMAIL PROTECTED] a écrit : Hi Harry - Can you get SUBSCRIBE registration ?? I don't know what SUBSCRIBE registration is, but looking at your sip.conf, there's a couple of things I would change: [100] type=friend username=100 secret=100 fromuser=100 ; Take this out - it's only needed when you want certain types of sip proxies are trying to ; register to this peer - not normally needed for asterisk. host=dynamic context=sip dtmfmode=rfc2833 ; I'd use inband here. I've tried rfc2833 here, too, but it doesn't seem to work as well as inband progressinband=no ; You don't really need this, and I think it doesn't make sense if you're doing rfc2833 dtmfmode On the Polycom side, you should use the following info: Address: 100 Auth User ID: 100 Auth Password: 100 Other than the SIP server address, these are the only important numbers on the Polycom. - Noah --- Noah Miller [EMAIL PROTECTED] a écrit : Hi Harry - I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. What version of the SIP firmware are you using? I've had success with 1.3.0, 1.3.1, 1.3.4, and 1.4.1. My sip.conf entries for my Polycom phones look like this: [12] type=friend username=12 secret=12 callerid=12 host=dynamic dtmfmode=inband context=no-callwaiting [EMAIL PROTECTED] disallow=all allow=ulaw Are you configuring directly on the phone, or using an FTP or TFTP server? Anybody could help me to configure Asterisk in order to set instant message and presence ? To the best of my knowledge the Presence feature of the Polycom phones does not work with Asterisk. I believe it only works with other IM clients. Hope this helps! Noah Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.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] SIP jitter?
Actually its not... Its for things supposed to be stable. The jitter buffer is not stable at all, putting this into the cvs-head would mean it would be taken out the day after because all carriers using cvs-head would go down. Its not some addon application you can disable, if this part coredumps, your asterisk coredumps. Btw i am trying the jitter buffer, and as soon as its a little more mature i will start stalking kram to get it into -head, but for now its just too soon. Joachim. Eric Wieling wrote: joachim wrote: Yes, It's untested and unfinished and touches the core of asterisk. (maybe causing massive amounts of deadlocks). So? That's what CVS-HEAD is there for. ___ 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 signature.asc Description: OpenPGP digital signature ___ 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] Video Conference
Hi Florian, thanks for your help. Yes I have enable videosuport in the sip.conf, and I think that i have the proper codecs. This is what i have in my sip.conf... [general] context=default videosupport=yes [097] type=friend username=video secret=video host=dynamic callerid=Video 097 canreinvite=no disallow=all ;allow=ulaw ;allow=alaw ;allow=speex allow=gsm allow=h261 allow=h263 nat=yes context=ip ;qualify=yes ;dtmfmode=rfc2833 Thanks for any help Marco González __ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.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] Grandstream Phone question
Does anyone know out there how to revert the firmware on a Grandstream Budgetone100 back to a more stable version. I have tried resetting it back to factory defaults with no luck . I have also changed the TFTP server in the phones internal webserver to a destination where the latest firmware is the Firmware version i want it to beany thoughts would be great.. Thanks in advance Ken Jerry wrote: We buy from both Graybar and Vibes. Not sure but think it took a couple weeks to get phones RMAd. I normally stock my own spares so do not track too close. On Feb 9, 2005, at 10:14 PM, Sergey Kuznetsov wrote: Jerry, Thanks a lot for the feedback! By the way, how long did it take to replace the faulty 10% of phones by RMA? What company did you use to buy it from? Jerry wrote: On Feb 9, 2005, at 9:14 PM, Sergey Kuznetsov wrote: Hi there, Does someone can share the experience with Cisco and Polycom Phones? How rock solid are they? And who will win in sound quality contest? I heard that Cisco phones is a Polycom replicas with changed design. Is that true? What else phones is better to implement to the medium sized business? The rock solid stability and superb sound quality is a must. Both have excellant sound. I think the Polycom speakerphone is a bit better. We are using mostly Polycom these days and our customers love them. My only issue is they do seem to have about a 10% failure rate within 90 days. After that they are solid - so far. They are also less expensive than the Cisco's and seem to have a better feature set and better control of their configs and buttons. I do like the layer 2 troubleshooting capabilities of the Ciscos as the Polycom seem to have no capabilities that I can find. I do not think the Cisco is any kind of a Polycom copy. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 ext. 37 Mobile phone: (647) 287-8448 ___ 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 -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ 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 ___ 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] SIP jitter?
joachim wrote: Actually its not... Its for things supposed to be stable. The jitter buffer is not stable at all, putting this into the cvs-head would mean it would be taken out the day after because all carriers using cvs-head would go down. Its not some addon application you can disable, if this part coredumps, your asterisk coredumps. We obviously disagree on this point. I feel that 1.0.x is what people should use in production. CVS-HEAD is what people should use for developement and testing, not for production. At least some others feel the same way. ___ 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] Need help with a Cisco 7960
Hi all, I have seen similar discussions about this problem earlier, but I need some help here! I've been using this phone for allmost two years without any problems. Just about a week ago I had the phone unplugged for a few days and when I plugged it in again it had lost all settings, including the settings password. It was reset back to the factory default. Well, then I decided to enter my TFTP settings again, but I can't even do that! I unlock the settings but I can not enter anything on that screen. I get the Phone unprovisioned message. Any idea? I know this is a little off topic, but I'm using it with my Asterisk. Any help would be apreciated! Many thanks, Christian ___ 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] SIP jitter?
Well im using cvs stable in production, but i know several of the carriers out there are using cvs head, because its the only one that has realtime... Anyway, cvs head is not testing. If you want to test the jitter buffer, download the patches compile them and see what happens, then report the results on mantis. If everyone says it doesnt seem to break anything, it will make it to cvs head, if its obviously broken it wont make it to cvs head. I reported it to cause massive deadlocks when using it with several simultaneous calls... This is how the * development people seem to work, and unless someone wants to start a 3rd branch, thats how its going to stay i think I didnt say i wouldnt like the jitter buffer to be in some kind of prepatched asterisk tree, but it should be in a 3rd (all really experimental things should be) I dont think we disagree btw :) Joachim. signature.asc Description: OpenPGP digital signature ___ 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] Need help with a Cisco 7960
Christian Moller wrote: Hi all, settings, including the settings password. It was reset back to the factory default. Well, then I decided to enter my TFTP settings again, but I can't even do that! I unlock the settings but I can not enter anything on that screen. I get the Phone unprovisioned message. Any idea? I know this is a little off topic, but I'm using it with my Asterisk. Any help would be apreciated! To manually enter the TFTP, you need to turn on the ALT TFTP option, then it will allow for this. Doug ___ 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] Using asterisk on a single phone line
I have asterisk installed on a linux workstation (1 phone in and 1 phone out jack). I have a single phone line that goes through this system. The box has a Smart/Lucent V.90 56 Kbps Fax/Data modem. # lspci -v 01:08.0 Communication controller: Lucent Microelectronics LT WinModem Subsystem: Risq Modular Systems, Inc.: Unknown device 044e Flags: fast Back2Back, medium devsel, IRQ 9 Memory at f410 (32-bit, non-prefetchable) [disabled] [size=256] I/O ports at 2400 [disabled] [size=8] I/O ports at 2000 [disabled] [size=256] Capabilities: [f8] Power Management version 2 The phone line goes in the data port and out of the phone port to the phone and I currently make/recieve calls this way. I want to configure Asterisk to show me any available data about any calls (i.e. phone numbers, caller-id) as well as screen unwanted calls to voice mail (if possible), play recoreded messages based on specified incoming phone numbers. My first question is, can asterisk do that? If so, can someone point me to documentation to explain how to set these features up? Also, what (if any) specific kernel support do I need? I build my own linux kernels and I usually turn off telephony and isdn support in the kernel. Thanks, Job___ 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] Asterisk and Sipura SPA-841 SIP phones
On Thu, Feb 10, 2005 at 09:07:58AM -0500, Giovanni Powell wrote: Nothing to do with your question, but by any chance, when you plugged the phone into the wall did you hear a dialtone or is this something generated by asterisk On a SIP phone, the dial tone is locally generated. The Sipura will only generate a dial tone if registrered. BTW, you can easily check on the Sipura web interface that the dial tones are parametered there. -- Nicolas Bougues Axialys Interactive ___ 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] Need help with a Cisco 7960
settings, including the settings password. It was reset back to the factory default. Well, then I decided to enter my TFTP settings again, but I can't even do that! I unlock the settings but I can not enter anything on that screen. I get the Phone unprovisioned message. Any idea? I know this is a little off topic, but I'm using it with my Asterisk. Any help would be apreciated! To manually enter the TFTP, you need to turn on the ALT TFTP option, then it will allow for this. Doug Did you switch DHCP off? Cheers Sascha ___ 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] Need help with a Cisco 7960
Sascha E. Pollok wrote: Did you switch DHCP off? I didn't need to. Doug ___ 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] Cisco 7960 and AutoAnswer.
Chris Wade wrote: C F wrote: On a Cisco 7960 Auto Answer is only configurable using the phone (not via TFTP), does anybody know if it is possible using sip notify or any other way but walking over to the phone? I've got a script that logs into the phone and sets this using the 'test key' functionality of the telnet CLI on the phone. This is the only other way I am aware of at this time. I'll post this on the wiki later this week. -Chris My apologies for not getting this posted yet. I've got several pieces of my setup here that I want to clean up to reflect recent changes in HEAD before I post them. I've also got to sanitize everything for posting to the community - a lot of this stuff is really specific to our implementation here. On top of this, I've also got some deadlines I'm trying to meet right now. Things will clear up for me around the 20th, so for anybody needing this stuff, just hold on until then. -Chris ___ 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] asterisk@home scary log
You've likely been hacked. I have recently had a similar incident where a hacker guessed my root password (MY BAD) and set up an ebay password skimming site. I noticed it when I got similar non-deliverable email messages. Obviously, first change your password and then look at the /var/www/html directory and see if there are unwelcome pages there. Also be sure to check who is logged in currently. I caught the (*%#@ SOB logged in and bounced the bastard. For what it's worth, the hacker's IP address was: 81.12.141.150. Karl Putz -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jean-Louis curty Sent: Thursday, February 10, 2005 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi everybody, I'm testing [EMAIL PROTECTED] 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log and saw this Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) Feb 9 20:30:07 asterisk1 sendmail[10077]: j1A1U7CY010077: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30068, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7Ns010089 Message accepted for delivery) Feb 9 20:30:17 asterisk1 sendmail[10094]: j1A1U7Ns010089: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30348, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? thanks jl ___ 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] Need help with a Cisco 7960
- Original Message - From: Doug Lytle [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 10, 2005 4:13 PM Subject: Re: [Asterisk-Users] Need help with a Cisco 7960 Sascha E. Pollok wrote: Did you switch DHCP off? I didn't need to. Doug Hi all, Yes, now it seem to work! But I can't understand how all settings were reset back to 0! Thanks, Christian ___ 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
[Asterisk-Users] Asterisk 1.0.5 won't pick up incoming calls
Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid argument Notice: Configuration file is /etc/zaptel.conf line 135: Unable to register tone zone 'us' And when I looked at the /var/log/messages file, I get: localhost kernel: Invalid 'next' pointer Asterisk would start fine after that, but it won't pick up any calls. Any chance that these errors cause asterisk not to pick up calls? I have included my configuration files, but they are default files that come with the installation (I made no modifications to the original files). Any help is much appreciated! zaptel.conf: - ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] ; ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ;group = trunkgroup,dchannel[,backup1...] ; ;trunkgroup is the numerical trunk group to create ;dchannelis the zap channel which will have the ;d-channel for the trunk. ;backup1 is an optional list of backup d-channels. ; ;trunkgroup = 1,24,48 ; ; Spanmap: Associates a span with a trunk group ;spanmap = zapspan,trunkgroup[,logicalspan] ; ;zapspan is the zap span number to associate ;trunkgroup is the trunkgroup (specified above) for the mapping ;logicalspan is the logical span number within the trunk group to use. ;if unspecified, no logical span number is used. ; ;spanmap = 1,1,1 ;spanmap = 2,1,2 ;spanmap = 3,1,3 ;spanmap = 4,1,4 [channels] ; ; Default language ; ;language=en ; ; Default context ; context=default ; ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1:Old National ISDN 1 ; switchtype=national ; ; Some switches (ATT especially) require network specific facility IE ; supported values are currently 'none', 'sdn', 'megacom', 'accunet' ; ;nsf=none ; ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown:Unknown ; private:Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; ;pridialplan=national ; ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan) ; ; unknown:Unknown ; private:Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; ;prilocaldialplan=national ; ; Overlap dialing mode (sending overlap digits) ; ;overlapdial=yes ; ; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; with all telcos. ; ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones ; ; priindication = outofband ; ; ; Signalling method (default is fxs). Valid values: ; em: E M ; em_w:E M Wink ; featd: Feature Group D (The fake, Adtran style, DTMF) ; featdmf: Feature Group D (The real thing, MF (domestic, US)) ; featb: Feature Group B (MF (domestic, US)) ; fxs_ls: FXS (Loop Start) ; fxs_gs: FXS (Ground Start) ; fxs_ks: FXS (Kewl Start) ; fxo_ls: FXO (Loop Start) ; fxo_gs: FXO (Ground Start) ; fxo_ks: FXO (Kewl Start) ; pri_cpe: PRI signalling, CPE side ; pri_net: PRI signalling, Network side ; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side ; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side ; sf: SF (Inband Tone) Signalling ; sf_w: SF Wink ; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) ; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) ; sf_featb: SF Feature Group B (MF (domestic, US)) ; The following are used for Radio interfaces: ; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the channel bank) ; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the channel bank) ; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the channel bank) ; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at the channel bank) ; em_rx: Receive audio/COR on an EM interface (1-way) ; em_tx: Transmit audio/PTT on an EM interface (1-way) ; em_txrx: Receive audio/COR AND Transmit audio/PTT on an EM interface (2-way) ; em_rxtx: same as
Re: [Asterisk-Users] asterisk@home scary log
Hi, I've also been a little worried about the security. How did they connect to your system? Through telnet or what? Since I've disabled all such services. Best, Christian - Original Message - From: Karl H. Putz [EMAIL PROTECTED] To: Jean-Louis curty [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 10, 2005 4:18 PM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log You've likely been hacked. I have recently had a similar incident where a hacker guessed my root password (MY BAD) and set up an ebay password skimming site. I noticed it when I got similar non-deliverable email messages. Obviously, first change your password and then look at the /var/www/html directory and see if there are unwelcome pages there. Also be sure to check who is logged in currently. I caught the (*%#@ SOB logged in and bounced the bastard. For what it's worth, the hacker's IP address was: 81.12.141.150. Karl Putz -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jean-Louis curty Sent: Thursday, February 10, 2005 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi everybody, I'm testing [EMAIL PROTECTED] 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log and saw this Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) Feb 9 20:30:07 asterisk1 sendmail[10077]: j1A1U7CY010077: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30068, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7Ns010089 Message accepted for delivery) Feb 9 20:30:17 asterisk1 sendmail[10094]: j1A1U7Ns010089: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30348, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? thanks jl ___ 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 ___ 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] SIP jitter?
It's untested and unfinished and touches the core of asterisk. (maybe causing massive amounts of deadlocks). It should go into CVS soon. Wasn't there a feature freeze around the end of february? Does this mean we'll have to wait till 1.4 or something to get decent sound on SIP? roy ___ 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] asterisk@home scary log
Wow that's scary, how did they gain access? I'm sitting behind a firewall (MS SBS 2003) with restricted ports but would like to check this cant happen to me. These are the files I have in the /var/www/html file addressbook amp.pngcisco files index.html maint nwebmail admin_asterisk directory images mainstyle.css meetme panel is this all good? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl H. Putz Sent: Thursday, February 10, 2005 10:19 AM To: Jean-Louis curty; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log You've likely been hacked. I have recently had a similar incident where a hacker guessed my root password (MY BAD) and set up an ebay password skimming site. I noticed it when I got similar non-deliverable email messages. Obviously, first change your password and then look at the /var/www/html directory and see if there are unwelcome pages there. Also be sure to check who is logged in currently. I caught the (*%#@ SOB logged in and bounced the bastard. For what it's worth, the hacker's IP address was: 81.12.141.150. Karl Putz -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jean-Louis curty Sent: Thursday, February 10, 2005 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi everybody, I'm testing [EMAIL PROTECTED] 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log and saw this Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) Feb 9 20:30:07 asterisk1 sendmail[10077]: j1A1U7CY010077: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30068, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7Ns010089 Message accepted for delivery) Feb 9 20:30:17 asterisk1 sendmail[10094]: j1A1U7Ns010089: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30348, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? thanks jl ___ 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 ___ 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] asterisk@home scary log
I had the system setup to allow http and ssh. The hack came in through ssh. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Christian Moller Sent: Thursday, February 10, 2005 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi, I've also been a little worried about the security. How did they connect to your system? Through telnet or what? Since I've disabled all such services. Best, Christian - Original Message - From: Karl H. Putz [EMAIL PROTECTED] To: Jean-Louis curty [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 10, 2005 4:18 PM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log You've likely been hacked. I have recently had a similar incident where a hacker guessed my root password (MY BAD) and set up an ebay password skimming site. I noticed it when I got similar non-deliverable email messages. Obviously, first change your password and then look at the /var/www/html directory and see if there are unwelcome pages there. Also be sure to check who is logged in currently. I caught the (*%#@ SOB logged in and bounced the bastard. For what it's worth, the hacker's IP address was: 81.12.141.150. Karl Putz -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jean-Louis curty Sent: Thursday, February 10, 2005 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi everybody, I'm testing [EMAIL PROTECTED] 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log and saw this Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) Feb 9 20:30:07 asterisk1 sendmail[10077]: j1A1U7CY010077: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30068, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7Ns010089 Message accepted for delivery) Feb 9 20:30:17 asterisk1 sendmail[10094]: j1A1U7Ns010089: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30348, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? thanks jl ___ 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 ___ 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] asterisk@home scary log
Hi, OK, well, I've disabled SSH/HTTP already so lets hope I will have my system working! Best and thanks, Christian - Original Message - From: Karl H. Putz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 10, 2005 4:56 PM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log I had the system setup to allow http and ssh. The hack came in through ssh. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Christian Moller Sent: Thursday, February 10, 2005 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi, I've also been a little worried about the security. How did they connect to your system? Through telnet or what? Since I've disabled all such services. Best, Christian - Original Message - From: Karl H. Putz [EMAIL PROTECTED] To: Jean-Louis curty [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 10, 2005 4:18 PM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log You've likely been hacked. I have recently had a similar incident where a hacker guessed my root password (MY BAD) and set up an ebay password skimming site. I noticed it when I got similar non-deliverable email messages. Obviously, first change your password and then look at the /var/www/html directory and see if there are unwelcome pages there. Also be sure to check who is logged in currently. I caught the (*%#@ SOB logged in and bounced the bastard. For what it's worth, the hacker's IP address was: 81.12.141.150. Karl Putz -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jean-Louis curty Sent: Thursday, February 10, 2005 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi everybody, I'm testing [EMAIL PROTECTED] 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log and saw this Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) Feb 9 20:30:07 asterisk1 sendmail[10077]: j1A1U7CY010077: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30068, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7Ns010089 Message accepted for delivery) Feb 9 20:30:17 asterisk1 sendmail[10094]: j1A1U7Ns010089: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30348, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? thanks jl ___ 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 ___ 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 ___ Asterisk-Users
RE: [Asterisk-Users] asterisk@home scary log
The hack came in through ssh. IMO, your best defence is an extremely strong root password; I am often mortified by looking at my logs and seeing all of the login attempts through SSH. OT: I am not up on Linux script-kiddie type tools, but I assume that there is a script of some sort that automates SSH probes. Can anyone suggest a good counter i.e. honeypot or throttling logon attempts. Yes, I know I can google it, but I'd rather hear the opinion of real Linux experts rather than the experts at About.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] asterisk@home scary log
ok ssh now disabled root password changed... where can I catch the message that are supposely sent by [EMAIL PROTECTED] ? On Thu, 10 Feb 2005 10:56:53 -0500, Karl H. Putz [EMAIL PROTECTED] wrote: I had the system setup to allow http and ssh. The hack came in through ssh. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Christian Moller Sent: Thursday, February 10, 2005 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi, I've also been a little worried about the security. How did they connect to your system? Through telnet or what? Since I've disabled all such services. Best, Christian - Original Message - From: Karl H. Putz [EMAIL PROTECTED] To: Jean-Louis curty [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 10, 2005 4:18 PM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log You've likely been hacked. I have recently had a similar incident where a hacker guessed my root password (MY BAD) and set up an ebay password skimming site. I noticed it when I got similar non-deliverable email messages. Obviously, first change your password and then look at the /var/www/html directory and see if there are unwelcome pages there. Also be sure to check who is logged in currently. I caught the (*%#@ SOB logged in and bounced the bastard. For what it's worth, the hacker's IP address was: 81.12.141.150. Karl Putz -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jean-Louis curty Sent: Thursday, February 10, 2005 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi everybody, I'm testing [EMAIL PROTECTED] 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log and saw this Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) Feb 9 20:30:07 asterisk1 sendmail[10077]: j1A1U7CY010077: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30068, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7Ns010089 Message accepted for delivery) Feb 9 20:30:17 asterisk1 sendmail[10094]: j1A1U7Ns010089: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30348, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? thanks jl ___ 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 ___ 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] asterisk@home scary log
On Thu, 2005-02-10 at 10:56 -0500, Karl H. Putz wrote: I had the system setup to allow http and ssh. The hack came in through ssh. I doubt you where hacked via ssh. Most likely you had your password brute force cracked. -Original Message- [mailto:[EMAIL PROTECTED] Behalf Of Christian Moller Sent: Thursday, February 10, 2005 10:39 AM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log your system? Through telnet or what? What moron still uses telnet these days? - Original Message - From: Karl H. Putz [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log You've likely been hacked. I have recently had a similar incident where a hacker guessed my root password (MY BAD) and set up an ebay password skimming site. This is a good example of why ease of use is not always a good thing. Had you actually had to learn more before you had an install, you would have been through a text or two that mention password strengths. And not to disparage the creator/maintainer of [EMAIL PROTECTED], but you really need to trust that your install was a little hardened before placing it on the network. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Re: asterisk@home scary log
On 10/02/05 15:10 +0100, Jean-Louis curty wrote: so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? There's a chance that you may have been hacked, but the logs you post look more like your mailserver is an open relay. What OS/Distro are you using, what version, and do you have the latest patches applied? What services are you running? Look for strange entries with uid 0 in your passwd file. Also check for root kits with a rootkit checker (chkrootkit.org). If everything pans out security-wise then the only problem is that you MTA is configured to be an open relay. If that's the case, then you need to fix it right away before you get on umpteen million blackhole lists. Consult the docs and/or community for the MTA that you're using to fix that. Jason ___ 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] asterisk@home scary log
On 10-Feb-2005, Colin Anderson wrote: IMO, your best defence is an extremely strong root password; I am often mortified by looking at my logs and seeing all of the login attempts through SSH. IMO, your best defence is leaving ssh's default setting which disallows root logins entirely. There's no reason for a remote user to ever have to log in as root. Root access should be obtained by a logged-in normal user using sudo, or su. Despite the fact that Linux distros seem to not support the wheel-group, you can accomplish a similar effect using sudo. -- David McNett [EMAIL PROTECTED] http://slacker.com/~nugget/ ___ 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] asterisk@home scary log
Why would you even want SSH exposed to the world? In fact, why expose it to anything but your local admin console, or *maybe* a vpn tunnel server if absolutely necessary? -d At 10:08 AM 2/10/2005, you wrote: The hack came in through ssh. IMO, your best defence is an extremely strong root password; I am often mortified by looking at my logs and seeing all of the login attempts through SSH. OT: I am not up on Linux script-kiddie type tools, but I assume that there is a script of some sort that automates SSH probes. Can anyone suggest a good counter i.e. honeypot or throttling logon attempts. Yes, I know I can google it, but I'd rather hear the opinion of real Linux experts rather than the experts at About.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 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] Re: asterisk@home scary log
hummm if that's the case I might not be the only one! I only installed the [EMAIL PROTECTED] iso (based on centos distro )and did not change a little comma of the configuration of sendmail, MTA is configured by default already by [EMAIL PROTECTED] jl On Thu, 10 Feb 2005 11:09:29 -0500, Jason Stewart [EMAIL PROTECTED] wrote: On 10/02/05 15:10 +0100, Jean-Louis curty wrote: so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? There's a chance that you may have been hacked, but the logs you post look more like your mailserver is an open relay. What OS/Distro are you using, what version, and do you have the latest patches applied? What services are you running? Look for strange entries with uid 0 in your passwd file. Also check for root kits with a rootkit checker (chkrootkit.org). If everything pans out security-wise then the only problem is that you MTA is configured to be an open relay. If that's the case, then you need to fix it right away before you get on umpteen million blackhole lists. Consult the docs and/or community for the MTA that you're using to fix that. Jason ___ 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] asterisk@home scary log
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Thursday, February 10, 2005 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log On Thu, 2005-02-10 at 10:56 -0500, Karl H. Putz wrote: I had the system setup to allow http and ssh. The hack came in through ssh. I doubt you where hacked via ssh. Most likely you had your password brute force cracked. That is what I meant to report to the list. SSH was simply the transport mechanism. Karl -Original Message- [mailto:[EMAIL PROTECTED] Behalf Of Christian Moller Sent: Thursday, February 10, 2005 10:39 AM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log your system? Through telnet or what? What moron still uses telnet these days? - Original Message - From: Karl H. Putz [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log You've likely been hacked. I have recently had a similar incident where a hacker guessed my root password (MY BAD) and set up an ebay password skimming site. This is a good example of why ease of use is not always a good thing. Had you actually had to learn more before you had an install, you would have been through a text or two that mention password strengths. And not to disparage the creator/maintainer of [EMAIL PROTECTED], but you really need to trust that your install was a little hardened before placing it on the network. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] asterisk@home scary log
Colin Anderson wrote: The hack came in through ssh. IMO, your best defence is an extremely strong root password; I am often mortified by looking at my logs and seeing all of the login attempts through SSH. OT: I am not up on Linux script-kiddie type tools, but I assume that there is a script of some sort that automates SSH probes. Can anyone suggest a good counter i.e. honeypot or throttling logon attempts. Yes, I know I can google it, but I'd rather hear the opinion of real Linux experts rather than the experts at About.com. Most scripts use port 22 as it would be too big a task to scan for ssh on all ports, so I run my ssh server way above port 1024. This has, touch wood, prevented any unusual activity in the last few months. Chris. ___ 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] asterisk@home scary log
On Thu, 2005-02-10 at 09:08 -0700, Colin Anderson wrote: The hack came in through ssh. IMO, your best defence is an extremely strong root password; I am often mortified by looking at my logs and seeing all of the login attempts through SSH. OT: I am not up on Linux script-kiddie type tools, but I assume that there is a script of some sort that automates SSH probes. Can anyone suggest a good counter i.e. honeypot or throttling logon attempts. Yes, I know I can google it, but I'd rather hear the opinion of real Linux experts rather than the experts at About.com. First, turn off root access from ssh. That is the first problem. Root should never be allowed to login except on console. Second, become familiar with su or sudo. Once you learn to login as your user and use su to become root, you learn that you have about three times as long of a root password. The first portion being a valid username, the second portion being a password for that username, and the third portion is either a root password or a valid local root exploit code. Recently the topic of brute force ssh attacks came up on our linux users group mailing list. The best option we had suggested was to do the above, then move ssh to a non standard port. Most scripts that are going to attack you are not going to consider the possibility that you are on a non standard port. Either you answer where they expect or they move on. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] asterisk@home scary log
Please all keep in mind that there are plenty of additional configs possible to Iptables. I should have restricted the originating IP address for TCP port 22 to come from at least my dhcp served address range. That would have blocked all hackers except those originating from within my specific ISP's dhcp served range. Not perfect but a good sight better that wide open! Karl Putz -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Christian Moller Sent: Thursday, February 10, 2005 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi, OK, well, I've disabled SSH/HTTP already so lets hope I will have my system working! Best and thanks, Christian - Original Message - From: Karl H. Putz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 10, 2005 4:56 PM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log I had the system setup to allow http and ssh. The hack came in through ssh. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Christian Moller Sent: Thursday, February 10, 2005 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi, I've also been a little worried about the security. How did they connect to your system? Through telnet or what? Since I've disabled all such services. Best, Christian - Original Message - From: Karl H. Putz [EMAIL PROTECTED] To: Jean-Louis curty [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 10, 2005 4:18 PM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log You've likely been hacked. I have recently had a similar incident where a hacker guessed my root password (MY BAD) and set up an ebay password skimming site. I noticed it when I got similar non-deliverable email messages. Obviously, first change your password and then look at the /var/www/html directory and see if there are unwelcome pages there. Also be sure to check who is logged in currently. I caught the (*%#@ SOB logged in and bounced the bastard. For what it's worth, the hacker's IP address was: 81.12.141.150. Karl Putz -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jean-Louis curty Sent: Thursday, February 10, 2005 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] scary log Hi everybody, I'm testing [EMAIL PROTECTED] 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log and saw this Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) Feb 9 20:30:07 asterisk1 sendmail[10077]: j1A1U7CY010077: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30068, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7Ns010089 Message accepted for delivery) Feb 9 20:30:17 asterisk1 sendmail[10094]: j1A1U7Ns010089: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30348, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? thanks jl ___ 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
Re: [Asterisk-Users] asterisk@home scary log
On Thu, 10 Feb 2005 10:47:22 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: And not to disparage the creator/maintainer of [EMAIL PROTECTED], but you really need to trust that your install was a little hardened before placing it on the network. Indeed. The default root password for a from-scratch [EMAIL PROTECTED] is very, very easy to guess. And I don't recall having the option to change it during the install, either, so I guess there are several hundreds (thousands?) of [EMAIL PROTECTED] boxes out there with the same weak password. (Yes, I do know how to change a root password post-install.. but the point is that the typical target user of such a hands-free product probably doesn't) Peter ___ 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] CallPickup from SIP phone
On Thu, February 10, 2005 3:39 am, Rich Adamson said: So I'm having trouble getting call-pickup working. Got a few different SIP phones (cisco 7940's and SPA-841s) all with pickupgroup=0 in sip.conf. Yes, it works fine from my 7960. On the 7960, I pick up ringing calls by pressing *8#. If that does not work for you, then ensure you don't have any extensions.conf entries that override *8, that all phone def's in sip.conf that you want to be able to pickup include something like callgroup=2, and the phone def's that you want to use the *8# have the pickupgroup=2. Should I be dialing star-eight-pound or star-eight followed by a callgroup or pickupgroup number? You suggest setting pickupgroup for all the phones. I've done this setting the value to 0 thinking that users would be dialing *80 to pickup ringing calls in group 0. Where should the callgroup settings go? I don't have that anywhere. Seems I'm mixing things up. Paul -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ 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] asterisk@home scary log
On Feb 10, 2005, at 17:12, denon wrote: Why would you even want SSH exposed to the world? In fact, why expose it to anything but your local admin console, or *maybe* a vpn tunnel server if absolutely necessary? SSH is perfectly fine, but the first thing I do is disallow any *password-based* access. Only SSH key access is allowed, ever. jens ___ 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] asterisk@home scary log
On Thu, 2005-02-10 at 10:47 -0600, Steven Critchfield wrote: This is a good example of why ease of use is not always a good thing. Had you actually had to learn more before you had an install, you would have been through a text or two that mention password strengths. Apropos ease of use: on publicly accessible servers I disable OpenSSH password access anyway, and allow login only by key. The key passphrase never travels across the net, and per ssh-add it can be stored by an agent which keeps it in memory until log off from your desktop session. I.e. you have to type it only once. Altogether, this gives much more security together with maximum ease of use. Also, just fyi, ssh account and password guessing resp. cracking seems to be hip right now, since I see attempts in my log on a daily basis. Fortunately, with passwords disabled, and as long as OpenSSH itself isn't vulnerable (buffer overflow etc.), I really don't need to be paranoid about this Regards, Bruno. ___ 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] Video Conference
Hi, -Original Message- Yes I have enable videosuport in the sip.conf, and I think that i have the proper codecs. This is what i have in my sip.conf... [097] type=friend username=video secret=video host=dynamic callerid=Video 097 canreinvite=no disallow=all ;allow=ulaw ;allow=alaw ;allow=speex allow=gsm allow=h261 allow=h263 nat=yes context=ip ;qualify=yes ;dtmfmode=rfc2833 Nat=yes ?? Is your client in NAT ? Having two RTP streams might confuse the firewall in such a case. Other than that, this would probably work. I have setups not very different. Florian ___ 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] asterisk@home scary log
On Thu, 10 Feb 2005 09:08:54 -0700, Colin Anderson OT: I am not up on Linux script-kiddie type tools, but I assume that there is a script of some sort that automates SSH probes. Can anyone suggest a good counter i.e. honeypot or throttling logon attempts. Yes, I know I can google it, but I'd rather hear the opinion of real Linux experts rather than the experts at About.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 Why not use keys and disable passwords all together? Geoff -- I have some G-Mail invites. Let me know if you want one. ___ 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] asterisk@home scary log
IMO, your best defence is leaving ssh's default setting which disallows root logins entirely. There's no reason for a remote user to ever have to log in as root. Root access should be obtained by a logged-in normal user using sudo, or su. I'm not sure what happens when you do a fresh compile and install of OpenSSH, but every distro I've ever worked with (Red Hat, Gentoo, Slackware, Vector, Tao, Yellow Dog, Debian, Knoppix, SuSe, Linspire, FreeBSD, OpenBSD, Darwin, OS X) has allowed root logins via SSH by default. Maybe they're changing that on newer versions of some distros. I dunno. But yes, make a strong password, and only login as a normal user. Do sudo's or su's to root once logged in. I can't imagine totally disabling SSH on an Asterisk machine! ___ 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] Re: asterisk@home scary log
One good step is to 'test' your public IPs against any mistake/hole like this. I've used http://www.ordb.org in the past for this purpose, others for sure are available. I would assume is a valuable feedback to provide to the folks from [EMAIL PROTECTED], to have a more conservative configuration in their default install. Jean-Louis curty wrote: hummm if that's the case I might not be the only one! I only installed the [EMAIL PROTECTED] iso (based on centos distro )and did not change a little comma of the configuration of sendmail, MTA is configured by default already by [EMAIL PROTECTED] On Thu, 10 Feb 2005 11:09:29 -0500, Jason Stewart [EMAIL PROTECTED] wrote: On 10/02/05 15:10 +0100, Jean-Louis curty wrote: so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? There's a chance that you may have been hacked, but the logs you post look more like your mailserver is an open relay. What OS/Distro are you using, what version, and do you have the latest patches applied? What services are you running? Look for strange entries with uid 0 in your passwd file. Also check for root kits with a rootkit checker (chkrootkit.org). If everything pans out security-wise then the only problem is that you MTA is configured to be an open relay. If that's the case, then you need to fix it right away before you get on umpteen million blackhole lists. Consult the docs and/or community for the MTA that you're using to fix that. ___ 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] asterisk@home scary log
You can always set up ssh to use host keys. Here are two howto's on what else? How to set them up. http://www.securityfocus.com/infocus/1806 Part 1 http://www.securityfocus.com/infocus/1810 Part 2 Dan. Steven Critchfield wrote: On Thu, 2005-02-10 at 09:08 -0700, Colin Anderson wrote: The hack came in through ssh. IMO, your best defence is an extremely strong root password; I am often mortified by looking at my logs and seeing all of the login attempts through SSH. OT: I am not up on Linux script-kiddie type tools, but I assume that there is a script of some sort that automates SSH probes. Can anyone suggest a good counter i.e. honeypot or throttling logon attempts. Yes, I know I can google it, but I'd rather hear the opinion of real Linux experts rather than the experts at About.com. First, turn off root access from ssh. That is the first problem. Root should never be allowed to login except on console. Second, become familiar with su or sudo. Once you learn to login as your user and use su to become root, you learn that you have about three times as long of a root password. The first portion being a valid username, the second portion being a password for that username, and the third portion is either a root password or a valid local root exploit code. Recently the topic of brute force ssh attacks came up on our linux users group mailing list. The best option we had suggested was to do the above, then move ssh to a non standard port. Most scripts that are going to attack you are not going to consider the possibility that you are on a non standard port. Either you answer where they expect or they move on. ___ 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] SER Asterisk Voicemail
Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java Rockx using sipsak for sending mwi sip notify messages to the phone but is there a simpler way which I am blindly ignoring?? Thank you in advance, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ 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] asterisk@home scary log
On Thu, 2005-02-10 at 10:12 -0600, denon wrote: Why would you even want SSH exposed to the world? In fact, why expose it to anything but your local admin console, or *maybe* a vpn tunnel server if absolutely necessary? What strange world do you live in where you think ssh can be limited to just the console? Do you speak RSA directly to your console, how about 3DES? SSH must be attached to a network where a client app sits between you and the encrypted link. Of course if you change ssh to be root in your above statement, it makes a tad more sense. But then again, since ssh would be oblivious to whether or not the link traversed a vpn tunnel, that doesn't make sense either. And I doubt anyone with the compute power and interest to decipher a ssh encypted link would bat an eyelash at having to go through the vpn link to get to the ssh. Of course at that point you have larger problems. At 10:08 AM 2/10/2005, you wrote: The hack came in through ssh. IMO, your best defence is an extremely strong root password; I am often mortified by looking at my logs and seeing all of the login attempts through SSH. OT: I am not up on Linux script-kiddie type tools, but I assume that there is a script of some sort that automates SSH probes. Can anyone suggest a good counter i.e. honeypot or throttling logon attempts. Yes, I know I can google it, but I'd rather hear the opinion of real Linux experts rather than the experts at About.com. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] asterisk@home scary log
On Thu, Feb 10, 2005 at 10:18:49AM -0500, Karl H. Putz wrote: You've likely been hacked. Don't make such hasty conclusions. I've seen too many strange messages explained as the machine is rooted. First of all: does your computer listen on port 25 (on all interfaces, not just localhost) netstat -lnt | grep :25 If so, it may simply be someone who sends you messages like: From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Assuming that your server is configured to accept mail for yourdoma.in , it will simply bounce the message back to the MX server for gmail.com. If that is the case you can: 1. don't listen on port 25 unless you really need to 2. don't accept mail for domains you don't have to 3. more aggressive spam filterring, e.g., RBL black-listing . - Unlike content filtering and virus checking, RBL black-lists take very little CPU, so they won't take precious system resources your * needs. - allowing only mail for existing users is also very effective. But exposes you to faster dictionary attacks to get the full list of your users I have recently had a similar incident where a hacker guessed my root password (MY BAD) and set up an ebay password skimming site. If someone had root on your machine and that guy was the least competent you shouldn't assume you managed to clean your machine from all the things he put. If this is a production system you want to trust, you should reinstall it from a clean copy or from a backup you can trust. I noticed it when I got similar non-deliverable email messages. Obviously, first change your password and then look at the /var/www/html directory and see if there are unwelcome pages there. Also be sure to check who is logged in currently. I caught the (*%#@ SOB logged in and bounced the bastard. Those are nice workarounds. But you cannot be really sure that those are the only trapdors he left behind. There are simply too many places to put hooks in. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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: How do I match a D? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
Steven Critchfield wrote: You should end up with invalid just like if you pressed a 1 without a extension or pattern matching the 1. Yeah, you're right, I was thinking of when the context doesn't contain an i extension either. Never mind :-) ___ 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] Re: asterisk@home scary log
On Thu, 2005-02-10 at 11:09 -0500, Jason Stewart wrote: There's a chance that you may have been hacked, but the logs you post look more like your mailserver is an open relay. You sure? I run postfix myself and am not proficient in analyzing sendmail logs, but looking at those lines Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) I find the relay (accepting host) is 127.0.0.1. So, even if ignoring the envelope 'from', there seems to be no doubt which host this mail was sent from. Regards, Bruno. ___ 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] asterisk@home scary log
I had the system setup to allow http and ssh. The hack came in through ssh. For those that aren't heavily involved with security topics, there has been many different approachs from many different IP's attempting to: a) exploit known ssh holes, and, b) ssh password guessing We tend to watch these attempts rather closely through intrusion detection tools like snort. As consultants, we are also under retainers to assist other companies with securing their facilities and watching for exploits. The exploit attempts happen every single day. There are multiple password guessing tools commonly available on the Internet. I eval'ed one of the tools and it took five seconds to guess a password that was five characters in length. It took an hour to guess a password that was eight characters, and around twenty-four hours to guess a password that was eight characters made up of uppercase, lowercase and non-alpha characters (eg, complex). Regardless, the guessing process is simply how much time does one want to devote to doing it (eg, what's the return value for spending the time exploiting a system). It doesn't make much difference whether one exposes telnet or ssh. Both can be exploited. But, the more complex you make the password, the more time-consuming and difficult it is to guess it. So, if you must expose either telnet or ssh, make your passwords very long and complex. If your O/S has the capability to lockout the account after 'xx' failed passwords, then do that. Automatically resetting the process after 'y' minutes disrupts the guessing process without the hacker knowing it, but still allows you access after that auto reset. Using something like seven failed attempts with a five minute reset is more then adequate in most cases. ___ 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] asterisk@home scary log
On Thu, Feb 10, 2005 at 10:12:11AM -0600, denon wrote: At 10:08 AM 2/10/2005, you wrote: The hack came in through ssh. IMO, your best defence is an extremely strong root password; I am often mortified by looking at my logs and seeing all of the login attempts through Assuming that a resonably smart attacker has no way of getting a valid username from, e.g, your email. I'm not sure how well can this be automated for script-kiddies, though Why would you even want SSH exposed to the world? Expose ssh to the world for remote administration. It is a great tool for that. A non-standard port is also often useful. In fact, why expose it to anything but your local admin console, or *maybe* a vpn tunnel server if absolutely necessary? and why is a vpn tunnel better than ssh? both leave you basically a password away from the server. ssh *is* a vpn tunnel. Unlike others it is well-understood and easy to configure so chances are you won't make mistakes configuring it. SSH. OT: I am not up on Linux script-kiddie type tools, but I assume that there is a script of some sort that automates SSH probes. Can anyone suggest a good counter i.e. honeypot or throttling logon attempts. Yes, I know I can google it, but I'd rather hear the opinion of real Linux experts rather than the experts at About.com. If you don't mind locking yourself out, use pam_tally.so in /etc/pam.d/ssh . It is documented in the docs of the pam package (e.g: pam.txt) -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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] Multiple SIP registrations for one account?
Cees de Groot wrote: For various reasons a customer of mine is moving from a SER-based to an Asterisk-based installation, mostly because of problems with SIP devices behind NAT trying to reach each other and because it's easier to do accounting when all calls go through Asterisk (canreinvite=no is the idea). canreinvite=yes does not affect call accounting in any way. ___ 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] CallPickup from SIP phone
SIP phones (cisco 7940's and SPA-841s) all with pickupgroup=0 in sip.conf. Yes, it works fine from my 7960. On the 7960, I pick up ringing calls by pressing *8#. If that does not work for you, then ensure you don't have any extensions.conf entries that override *8, that all phone def's in sip.conf that you want to be able to pickup include something like callgroup=2, and the phone def's that you want to use the *8# have the pickupgroup=2. Should I be dialing star-eight-pound or star-eight followed by a callgroup or pickupgroup number? You suggest setting pickupgroup for all the phones. I've done this setting the value to 0 thinking that users would be dialing *80 to pickup ringing calls in group 0. Where should the callgroup settings go? I don't have that anywhere. Seems I'm mixing things up. Unless something has changed in asterisk that I missed, I don't believe it is possible to use *8group #. As mentioned previously, those phones that have a defined 'pickupgroup' are the only ones that can pick up a phone/line that has been defined with a 'callgroup'. If you have a small installation, then just include something like: callgroup=2 pickupgroup=2 in all sip phone definitions. (I don't know if '0' is a valid number that can be assigned to callgroup or pickupgroup. You might test it after you've made it work with some other number.) ___ 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] asterisk@home scary log
On Thu, Feb 10, 2005 at 08:30:32AM -0800, Geoff Scott wrote: Why not use keys and disable passwords all together? Becasue you cannot memorize and type keys. If you can carry the keys with you, that's fine. OTP (One-Time_Password) device should be nice. a keychain for a usbstick that will hold your secret key may be another solution. Both have some non-trivial assumtions about the hardware that prevented me thus far from implementing them. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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] asterisk@home scary log
On Thu, 2005-02-10 at 10:42 -0600, Daniel Wright wrote: You can always set up ssh to use host keys. Here are two howto's on what else? How to set them up. http://www.securityfocus.com/infocus/1806 Part 1 http://www.securityfocus.com/infocus/1810 Part 2 Great links. One may add that first actually deals with host keys, which identify the server to the client, and the second with identities resp. pubkeys, which identify the client to the server. I guess it's actually the latter item we are currently talking about. Host keys are essential as well of course, to prevent phishing and the likes. Regards, Bruno. ___ 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] asterisk@home scary log
Thanks, everyone, for the excellent suggestions. For posterity and for future reference when this thread comes up again, summarizing the best way(s) to defend against SSH logon attempts: 1. Don't allow root thru SSH or Telnet, force logon as regular user and sudo 2. If you must run SSH or Telnet, run it on a non-obvious port 1024 3. Change all default passwords in the system. For example, I run Cyrus-IMAPD on another server and the default password in the install of Cyrus is CYRUS user and CYRUS password - I get at least 5 password attempts per day with that same user/pass combination. (yes, I changed it!) 4. Restrict originating IP's to SSH to only accept your local subnet or a range of trusted IP's 5. Use key-based auth mechanism rather than password. It's my understanding that the key is never sent, only a hash of the key. The target system compares the hash against it's hash of the key, and if it matches, cool. 6. IPSec, (or some other VPN) which is quite problematic cross-platform. Dave McNett wrote: IMO, your best defence is leaving ssh's default setting which disallows root logins entirely. There's no reason for a remote user to ever have to log in as root. Root access should be obtained by a logged-in normal user using sudo, or su. Weird thing is, I never touched the default SSH setting and I log in as root just fine. FC2. Is this documented?? dean collins wrote: Colin, how do I find these logs on the [EMAIL PROTECTED] install? Dunno about [EMAIL PROTECTED], on Fedora/RH, you want to examine the file /var/log/secure. Also, a telltale sign of trouble is when you log on as you in SSH, the console will say the last sucessful logon. If that's not you, or shomeone you know, then you are in trouble. ___ 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
