[asterisk-users] Why are the hackers scanning for these?
Hey, I'm going thru logs, and I see some very common and interesting things that the hackers are looking for. In a whole bunch of scans, I've noticed that the first guess or two for sip accounts is usually a 10-digit number. I'm asking myself, why these numbers? Are they looking for a voip trunk? Or is it just like a serial number for the scan? What? Here's some examples: 2648061411 3190339404 2685608247 3358171034 2092652562 2206598858 Just trying to follow the advice: Know thy Enemy murf Steve Murphy ParseTree Corp. 57 Lane 17 Cody, WY 82414 ✉ m...@parsetree.com ☎ 307-899-5535 Signature powered by http://www.wisestamp.com/email-install?utm_source=extensionutm_medium=emailutm_campaign=footer WiseStamphttp://www.wisestamp.com/email-install?utm_source=extensionutm_medium=emailutm_campaign=footer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
Here's some examples: 2648061411 3190339404 I'm getting exactly the same. Odds of getting a working number, are like the odds of winning the lottery. My guess is they are either trying to find a voip trunk, or they are trying to make cold calls to the extensions on my system. Sales or something similar. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
My guess is they are looking for 10 digit phone numbers as extensions. Are they all from 1 IP address or from many? If from many, they are likely many serial scan or from a list of suspected VOIP numbers. If from one, and that random, then from a list of suspected VOIP numbers. Since you listed a phone number as part of your signature… I might guess hackers might soon add that number to a scan list. It is one thing to randomly run 2,XXX-, to 999-999-, with skips for the “dead zones,” (0-XXX-XXX-) etc. but another to hit suspected VOIP numbers. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Murphy Sent: Sunday, November 07, 2010 8:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Why are the hackers scanning for these? Hey, I'm going thru logs, and I see some very common and interesting things that the hackers are looking for. In a whole bunch of scans, I've noticed that the first guess or two for sip accounts is usually a 10-digit number. I'm asking myself, why these numbers? Are they looking for a voip trunk? Or is it just like a serial number for the scan? What? Here's some examples: 2648061411 3190339404 2685608247 3358171034 2092652562 2206598858 Just trying to follow the advice: Know thy Enemy murf Steve Murphy ParseTree Corp. 57 Lane 17 Cody, WY 82414 ✉ m...@parsetree.com ☎ 307-899-5535 http://www.wisestamp.com/email-install?utm_source=extensionutm_medium=emailutm_campaign=footer Signature powered by http://www.wisestamp.com/email-install?utm_source=extensionutm_medium=emailutm_campaign=footer WiseStamp http://s.wisestamp.com/pixel.png?p=mozillav=2.0.3t=1289138760949u=949715e=4286 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Sunday, November 07, 2010 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why are the hackers scanning for these? Here's some examples: 2648061411 3190339404 I'm getting exactly the same. Odds of getting a working number, are like the odds of winning the lottery. My guess is they are either trying to find a voip trunk, or they are trying to make cold calls to the extensions on my system. Sales or something similar. We got pounded last weekend, but installed a list of distant IPs in IPTABLES and see nothing this weekend. We have no need to be contacted by any sites more than 2500 miles away, and not too many from within 2500 miles. ;-) Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] scratchy sound on TE410P
asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were scratchy and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. Is this a problem at the carrier? I'm trying to call them now, but it's Sunday morning in the sticks, and my chances of getting someone with a clue are slim to none. I restarted dahdi but that had no effect. I watched dahdi_tool as calls came in and out but there isn't really a lot of information there. Sangoma has a cool tool wanpipemon that shows error stats and such on the span. Is there such a tool for Digium cards? Any suggestions? Thanks, -- Jeff LaCoursiere SunFone j...@sunfone.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
On Sun, Nov 7, 2010 at 10:00 AM, Cary Fitch ca...@usawide.net wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Sunday, November 07, 2010 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why are the hackers scanning for these? Here's some examples: 2648061411 3190339404 I'm getting exactly the same. Odds of getting a working number, are like the odds of winning the lottery. My guess is they are either trying to find a voip trunk, or they are trying to make cold calls to the extensions on my system. Sales or something similar. We got pounded last weekend, but installed a list of distant IPs in IPTABLES and see nothing this weekend. We have no need to be contacted by any sites more than 2500 miles away, and not too many from within 2500 miles. ;-) Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I've just switched my outbound ip address a week ago. Not static, but dhcp on TimeWarner cable. I've registered only with another of our offices. The outbound calls are all pstn bound through Teliax. But somehow my log is filling up with registration requests over this new ip address from a bunch of addresses. How can these guys find my new ip address? Or are they just scanning all ip addresses in creation? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
On Sun, Nov 07, 2010 at 07:11:43AM -0700, Steve Murphy wrote: Hey, I'm going thru logs, and I see some very common and interesting things that the hackers are looking for. In a whole bunch of scans, I've noticed that the first guess or two for sip accounts is usually a 10-digit number. I'm asking myself, why these numbers? Are they looking for a voip trunk? Or is it just like a serial number for the scan? What? It's SIPVicious. Before it starts its sequential scan, it makes sure that it can tell the difference between a valid peer and an unknown one. It tries two random peers, expecting a 404 response to at least one (most likely both) of them. Then, if it later gets a 401 during the sequential scan, it knows it's found a good peer name that can be targeted for password guessing. On the other hand, if both random guesses elicit 401 responses to REGISTERs, it knows that it can't winnow out the real peers, and (normally) just gives up right there. That's why 'alwaysauthreject' is so effective at stopping the attacks (as opposed to blocking them). But if the attacker uses the '--force' option, which causes the scan to press on regardless, or something other than SIPVicious, only something like fail2ban will help, but that won't save your bandwidth like 'alwaysauthreject' will. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
I've just switched my outbound ip address a week ago. Not static, but dhcp on TimeWarner cable. I've registered only with another of our offices. The outbound calls are all pstn bound through Teliax. But somehow my log is filling up with registration requests over this new ip address from a bunch of addresses. How can these guys find my new ip address? Or are they just scanning all ip addresses in creation? sean -- _ Follow the money Just like for Spam, there is money in Sip-Hacking. Anyone that has SIP traffic to move (selling the service) has money. If they can move it for free, even more money. A few servers running Hacking programs (SIPVicious) or e-mail server hacking programs is no big deal and bandwidth at colo centers is unlimited. Then they convert to BOT controllers and have free computers and bandwidth world wide. They generate a database of public IP addresses (DHCP, whatever) and have a target of poorly protected IPs to troll. Lucky you. ;-) Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound on TE410P
On 11/7/10 9:26 AM, Jeff LaCoursiere wrote: asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were scratchy and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. Is this a problem at the carrier? I'm trying to call them now, but it's Sunday morning in the sticks, and my chances of getting someone with a clue are slim to none. I restarted dahdi but that had no effect. I watched dahdi_tool as calls came in and out but there isn't really a lot of information there. Sangoma has a cool tool wanpipemon that shows error stats and such on the span. Is there such a tool for Digium cards? Any suggestions? dahdi_maint -s spanno will provide the error counters that are available for the span. i.e.: ]# ./dahdi_maint -s 1 Span 1: FEC : 0: CEC : 0: CVC : 0: EBC : 0: BEC : 0: PRBS: 0: GES : 0: You can see verify if the drivers are adjusting the gain on a channel with the dahdi_diag tool. 'make dahdi_diag' in dahdi_tools in order to build it, since it's not built by default. Then: ']# dmesg -c /dev/null ./dahdi_diag 1 dmesg -c' And look for the gainalloc output to see if DAHDI is gaining the channels. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
Adding on more thoughts: Think what Google has done in Mapping the Earth, Mapping the Web, and now working on Google Voice and Google Mail. Every one of those makes money either directly and/or synergistically with other components. Now consider someone with telephone interests or spam interests. In this modern database and filtering and probing age, load in ARIN or RIPE IP Ranges, start building database data and filters, and let it run... And the other IP areas too. Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Check out the AstLinux site There is a version there for the Alix boards, though I am not impressed with Alix. IMO overpriced. John Novack Bruce B wrote: Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
On Sun, Nov 7, 2010 at 11:23 AM, Bruce B bruceb...@gmail.com wrote: Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Only compile the modules you need. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
John, AstLinux seems promising. Have you used this flavor in production environment? Paul, So, don't use the Yum repositoy?! And, are you sure that is the only thing needs to be done. I am thinking there is more tweaking need to be done. I am not looking to just install Asterisk but it should be production ready as well. Meaning solid, reliable machine. Thanks On Sun, Nov 7, 2010 at 12:28 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Nov 7, 2010 at 11:23 AM, Bruce B bruceb...@gmail.com wrote: Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Only compile the modules you need. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
On Sun, Nov 7, 2010 at 1:01 PM, Bruce B bruceb...@gmail.com wrote: So, don't use the Yum repositoy?! Usually not. If you don't want to get your hand dirty managing the OS layer, try Askozia[1]. Most embedded solutions will use a modified Busybox installation, allowing for lightweight binaries. Most desktop distros are just too bloated for an embedded solution. [1] http://www.askozia.com/ -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Big practical systems
I don't want to start the How many calls can Asterisk handle? discussion or How many angels can stand on the point of a pin? discussion either. But can anyone contribute some practical knowledge of systems that take in channel bank T1s or DS3s from far away, and process the calls? I am looking for real world, been there, done that, or check the 'Belchfire Systems GigaFiber 65536' system. Not to start the discussion, but Is there a board that will take a DS3 (672 channels) and a system that will handle the calls, or is that a silly question? Is there an IP box that would take the DS3 and then a system that would handle the calls? My guess would be yes because the actual call load would be far lower than 672 calls. Maybe 100-150 or so simultaneous. Each line/call would have to have absolute caller ID. In other words, PSTN call handling. Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
On Sun, Nov 7, 2010 at 11:03 AM, Cary Fitch ca...@usawide.net wrote: Adding on more thoughts: Think what Google has done in Mapping the Earth, Mapping the Web, and now working on Google Voice and Google Mail. Every one of those makes money either directly and/or synergistically with other components. Now consider someone with telephone interests or spam interests. In this modern database and filtering and probing age, load in ARIN or RIPE IP Ranges, start building database data and filters, and let it run... And the other IP areas too. Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All makes me think of forcing an ip address change each night by spoofing the mac address. Each day they'd have to find me anew! sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big practical systems
On 07/11/2010 19:29, Cary Fitch wrote: I don't want to start the How many calls can Asterisk handle? discussion or How many angels can stand on the point of a pin? discussion either. But can anyone contribute some practical knowledge of systems that take in channel bank T1s or DS3s from far away, and process the calls? I am looking for real world, been there, done that, or check the 'Belchfire Systems GigaFiber 65536' system. Not to start the discussion, but Is there a board that will take a DS3 (672 channels) and a system that will handle the calls, or is that a silly question? Is there an IP box that would take the DS3 and then a system that would handle the calls? My guess would be yes because the actual call load would be far lower than 672 calls. Maybe 100-150 or so simultaneous. Each line/call would have to have absolute caller ID. In other words, PSTN call handling. Cary Hi, Did you saw this before: http://lists.digium.com/pipermail/asterisk-users/2008-April/209146.html ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big practical systems
inline I don't want to start the How many calls can Asterisk handle? discussion or How many angels can stand on the point of a pin? discussion either. You just did But can anyone contribute some practical knowledge of systems that take in channel bank T1s or DS3s from far away, and process the calls? Most of us are far too busy. These things are best learned the hardway. I am looking for real world, been there, done that, or check the 'Belchfire Systems GigaFiber 65536' system. Its called Asterisk. Not to start the discussion, but Is there a board that will take a DS3 (672 channels) and a system that will handle the calls, or is that a silly question? There are many. The primary problem it getting a provider to provide you with a DS3. Is there an IP box that would take the DS3 and then a system that would handle the calls? My guess would be yes because the actual call load would be far lower than 672 calls. Maybe 100-150 or so simultaneous. There are a few solutions here and several expensive chunks of hardware. Do you want to put all your eggs in one basket? Each line/call would have to have absolute caller ID. In other words, PSTN call handling. That is between you and the provider. The technology exists on the wire. Cary Gringo Malvado... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Yes, Astlinux and Askozia are the leading candidates for use on such small platforms. I have used Astlinux on Soekris boards which are similar. I wrote it up here: http://www.mgraves.org/?p=1092 That was some time ago but the basics of it are still sound. Here's some further thoughts on small format hardware f Asterisk: http://www.mgraves.org/2010/07/d-i-y-asterisk-appliances-a-question-of-s cale/ Given the recent trend in nettops they now seem like a better value in some ways. Michael On Sun, 07 Nov 2010 12:03:18 -0500, John Novack wrote: Check out the AstLinux site There is a version there for the Alix boards, though I am not impressed with Alix. IMO overpriced. John Novack Bruce B wrote: Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big practical systems
I believe this looks like a standard channel bank. Asterisk generates all audio. Ring and hook status are sent out of band. Dial tones are in-band. Ringback, busy, congestion are in-band audio. I would think a standard T1 card would be fine. That said, I would verify this with the LEC. On Nov 7, 2010, at 1:22 PM, Cary Fitch ca...@usawide.net wrote: Alternate question: Asterisk/PSTN oriented. If an Asterisk system were interfaced via a T1 to a local telco loop to a customer premises: (This is not a T1 to the customer premises, but a T1 to the telco who then demuxes it to copper to the customer premises. IE. In Telecom terms an EEL.) Will Asterisk handle that scenario with common drivers and cards? Who generates the customer audio comfort sounds, ringing, busy, etc? Cary I know a lot, but not everything. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk spontaneous reboot
On 11/06/2010 09:18 PM, Sherwood McGowan wrote:
On Sat, Nov 6, 2010 at 2:45 PM, Jonas Kellensjonas.kell...@telenet.be
wrote:
On 11/06/2010 07:18 PM, Tilghman Lesher wrote:
On Saturday 06 November 2010 11:22:06 Jonas Kellens wrote:
Hello,
I just experienced a spontaneous reboot of Asterisk. This is my log file
/var/log/messages :
Nov 6 16:37:37 vps kernel: miniserv.pl invoked oom-killer:
First line. Your miniserv.pl allocated more memory than is allocated to
the system, so the dreaded OOM killer came into play and killed a selected
process. Have you considered enabling swap memory?
I have 512 MB real RAM and 1024 of swap.
bash-3.2# cat /proc/meminfo
MemTotal: 524288 kB
MemFree: 23760 kB
Buffers: 28564 kB
Cached: 348668 kB
SwapCached: 6536 kB
Active: 193972 kB
Inactive: 231216 kB
HighTotal: 0 kB
HighFree:0 kB
LowTotal: 524288 kB
LowFree: 23760 kB
SwapTotal: 1048568 kB
SwapFree: 949456 kB
Dirty: 768 kB
Writeback: 0 kB
AnonPages: 46652 kB
Mapped: 16884 kB
Slab:21000 kB
PageTables: 8084 kB
NFS_Unstable:0 kB
Bounce: 0 kB
CommitLimit: 1310712 kB
Committed_AS: 321288 kB
VmallocTotal: 34359738367 kB
VmallocUsed: 784 kB
VmallocChunk: 34359737535 kB
miniserv.pl... I have webmin running yes and it was stopped after the
restart of Asterisk...
So the bad one in this story is WebMin that was eating up all the memory ?
Jonas.
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Yessir, that's the culprit in this case
Strange, today I saw this in the logs :
Nov 7 17:02:18 vps kernel: crond invoked oom-killer: gfp_mask=0x201d2,
order=0, oomkilladj=0
Nov 7 17:02:18 vps kernel:
Nov 7 17:02:18 vps kernel: Call Trace:
Nov 7 17:02:18 vps kernel: [802bf74e] out_of_memory+0x8b/0x203
Nov 7 17:02:18 vps kernel: [8020f947] __alloc_pages+0x27f/0x308
Nov 7 17:02:18 vps kernel: [802138db]
__do_page_cache_readahead+0xc6/0x1ab
Nov 7 17:02:18 vps kernel: [802141c7] filemap_nopage+0x14c/0x360
Nov 7 17:02:18 vps kernel: [80208e8c]
__handle_mm_fault+0x442/0x1445
Nov 7 17:02:18 vps kernel: [8028866d] deactivate_task+0x28/0x5f
Nov 7 17:02:18 vps kernel: [8026769a] do_page_fault+0xf7b/0x12e0
Nov 7 17:02:18 vps kernel: [8025c8ff] hrtimer_cancel+0xc/0x16
Nov 7 17:02:18 vps kernel: [80263b14] do_nanosleep+0x47/0x70
Nov 7 17:02:18 vps kernel: [8025c7ec]
hrtimer_nanosleep+0x58/0x118
Nov 7 17:02:18 vps kernel: [8026082b] error_exit+0x0/0x6e
Nov 7 17:02:18 vps kernel:
Nov 7 17:02:18 vps kernel: Mem-info:
snip
So this time it is crond that invoked oom-killer...
I've had this since I commented out this in /etc/asterisk/logger.conf :
exec_after_rotate=gzip -9 ${filename}.2
Whenever I do a logger rotate on the Asterisk CLI, the CLI hangs...
Kind regards,
Jonas.
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[asterisk-users] install
I have installed Asterisk before w/ no issues but while trying today (1.6.2.13 and centors 5.4) I receive the following at the CLI: The configure script must be executed before running 'make'. Please run ./configure. Any tricks on getting through this? I did not select to libpri or zapata. only asterisk as i am building a voip only design on rackspace.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] install
On 7 Nov 2010, at 20:59, Thomas Perron wrote:
I have installed Asterisk before w/ no issues but while trying today
(1.6.2.13 and centors 5.4) I receive the following at the CLI:
The configure script must be executed before running 'make'.
Please run ./configure.
Any tricks on getting through this?
Type ./configure
S
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Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
There is a network of telephone switch collectors, worldwide that uses Asterisk to interface the network with their switches, as well as members who have an interest in these old switches but don't yet have one working. I have personally set up about 20 nodes with AstLinux on HP thin clients, 55xx and 57xx, mainly in the US. At least one has a single port T1card, the TE110, others use SIP phones and SIP ATA's, others use Cisco 3810's with a SIP IOS. Several are running on old 55XX versions with only 128 Meg of Ram They all simply work. One I monitor closely has been up around 180 days now. Configure modules.conf to noload stuff you don't need AstLinux has a nice web interface for ease of configuration. The older HP thin clients USED to be available on eBay at bargain prices, though lately the MagicJack crowd seems to have run up the price a bit As long as you don't want to do any thing fancy, AstLinux will do nicely John Novack Bruce B wrote: John, AstLinux seems promising. Have you used this flavor in production environment? Paul, So, don't use the Yum repositoy?! And, are you sure that is the only thing needs to be done. I am thinking there is more tweaking need to be done. I am not looking to just install Asterisk but it should be production ready as well. Meaning solid, reliable machine. Thanks On Sun, Nov 7, 2010 at 12:28 PM, Paul Belanger paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com wrote: On Sun, Nov 7, 2010 at 11:23 AM, Bruce B bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Only compile the modules you need. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] install
When all else fails, do what the program tells you to do!
The requirement to run ./configure has been around since sometime in 1.4
And CentOS 5.5 is current. You might wan to update it first?
John Novack
Thomas Perron wrote:
I have installed Asterisk before w/ no issues but while trying today
(1.6.2.13 and centors 5.4) I receive the following at the CLI:
The configure script must be executed before running 'make'.
Please run ./configure.
Any tricks on getting through this?
I did not select to libpri or zapata.
only asterisk as i am building a voip only design on rackspace.com
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Re: [asterisk-users] Asterisk spontaneous reboot
On Sun, Nov 7, 2010 at 3:58 PM, Jonas Kellens jonas.kell...@telenet.be wrote: On 11/06/2010 09:18 PM, Sherwood McGowan wrote: On Sat, Nov 6, 2010 at 2:45 PM, Jonas Kellensjonas.kell...@telenet.be wrote: On 11/06/2010 07:18 PM, Tilghman Lesher wrote: On Saturday 06 November 2010 11:22:06 Jonas Kellens wrote: Hello, I just experienced a spontaneous reboot of Asterisk. This is my log file /var/log/messages : Nov 6 16:37:37 vps kernel: miniserv.pl invoked oom-killer: First line. Your miniserv.pl allocated more memory than is allocated to the system, so the dreaded OOM killer came into play and killed a selected process. Have you considered enabling swap memory? I have 512 MB real RAM and 1024 of swap. bash-3.2# cat /proc/meminfo MemTotal: 524288 kB MemFree: 23760 kB Buffers: 28564 kB Cached: 348668 kB SwapCached: 6536 kB Active: 193972 kB Inactive: 231216 kB HighTotal: 0 kB HighFree: 0 kB LowTotal: 524288 kB LowFree: 23760 kB SwapTotal: 1048568 kB SwapFree: 949456 kB Dirty: 768 kB Writeback: 0 kB AnonPages: 46652 kB Mapped: 16884 kB Slab: 21000 kB PageTables: 8084 kB NFS_Unstable: 0 kB Bounce: 0 kB CommitLimit: 1310712 kB Committed_AS: 321288 kB VmallocTotal: 34359738367 kB VmallocUsed: 784 kB VmallocChunk: 34359737535 kB miniserv.pl... I have webmin running yes and it was stopped after the restart of Asterisk... So the bad one in this story is WebMin that was eating up all the memory ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yessir, that's the culprit in this case Strange, today I saw this in the logs : Nov 7 17:02:18 vps kernel: crond invoked oom-killer: gfp_mask=0x201d2, order=0, oomkilladj=0 Nov 7 17:02:18 vps kernel: Nov 7 17:02:18 vps kernel: Call Trace: Nov 7 17:02:18 vps kernel: [802bf74e] out_of_memory+0x8b/0x203 Nov 7 17:02:18 vps kernel: [8020f947] __alloc_pages+0x27f/0x308 Nov 7 17:02:18 vps kernel: [802138db] __do_page_cache_readahead+0xc6/0x1ab Nov 7 17:02:18 vps kernel: [802141c7] filemap_nopage+0x14c/0x360 Nov 7 17:02:18 vps kernel: [80208e8c] __handle_mm_fault+0x442/0x1445 Nov 7 17:02:18 vps kernel: [8028866d] deactivate_task+0x28/0x5f Nov 7 17:02:18 vps kernel: [8026769a] do_page_fault+0xf7b/0x12e0 Nov 7 17:02:18 vps kernel: [8025c8ff] hrtimer_cancel+0xc/0x16 Nov 7 17:02:18 vps kernel: [80263b14] do_nanosleep+0x47/0x70 Nov 7 17:02:18 vps kernel: [8025c7ec] hrtimer_nanosleep+0x58/0x118 Nov 7 17:02:18 vps kernel: [8026082b] error_exit+0x0/0x6e Nov 7 17:02:18 vps kernel: Nov 7 17:02:18 vps kernel: Mem-info: snip So this time it is crond that invoked oom-killer... Please read up on how the oom killer works. crond didn't invoke anything, but was rather the unfortunate task chosen to be sacrificed. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big practical systems
On Sun, Nov 7, 2010 at 1:29 PM, Cary Fitch ca...@usawide.net wrote: But can anyone contribute some practical knowledge of systems that take in channel bank T1s or DS3s from far away, and process the calls? Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a channelized DS3 into PRIs. Not to start the discussion, but Is there a board that will take a DS3 (672 channels) and a system that will handle the calls, or is that a silly question? If by board, you mean PCI board for shoving in something with an intel cpu, not that I've ever heard. Digium sells 4x port PRI boards, and some competitor sells an 8x port PRI board, but I've never tried any boards not made by Digium. The only thing silly is the idea of trusting that many calls to PC hardware. Is there an IP box that would take the DS3 and then a system that would handle the calls? Yes, embedded hardware from a vendor you've heard of will do that. Cisco makes a 3845 which can terminate about 20 PRIs in one appliance. My guess would be yes because the actual call load would be far lower than 672 calls. Maybe 100-150 or so simultaneous. Well, then it's not really a DS3. If it can't do the whole thing without melting down, it shouldn't advertise itself as DS3. The Adtran gear works rock solid when pushed to the limit. If you're just talking 150 calls, you could do that with two 4x port cards in a single PC. I thought you were talking a lot bigger. Each line/call would have to have absolute caller ID. In other words, PSTN call handling. Ummm, there's no such thing as absolute caller ID. You wanna try that question again? callerID is not legally binding, is not used by billing, anybody can spoof it. The closest you can get is to have a LEC provide ANI. You don't need PRI to get that. You can get that via a quality voip provider, or yourself using your own termination gear to convert into voip. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Hello,
I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The
backend is a MySQL database running through the ODBC backend in Asterisk. At
this point everything works in terms of phones registering, placing calls
between them, etc. However, I am having a problem setting the Caller ID number
whenever I am using the Realtime database for the SIP users/peers. If I use a
static sip.conf configuration instead of the database, everything works fine.
Unfortunately a static sip.conf file won't work in my application.
In this example:
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)}) ;;;PS: This shows the
correct number of 2 on the CLI console...
exten = 412,n,Dial(SIP/412)
Whenever another phone calls extension 412, the call is forwarded to SIP/412
and should have TEST as the CallerID name and 2 as the CallerID number.
But, whenever I am using the realtime backend, the caller ID number always
displays on the destination phone as that phone's username. Meaning, if phone
SIP/412 receives the call from the example above, the caller ID name displayed
is TEST but the caller ID number is always 412.
What could be causing this?
Brett Woollum
br...@woollum.com
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Re: [asterisk-users] Big practical systems
Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a channelized DS3 into PRIs. Thanks, will look at that. Ah, a DS3/T1 mux. I was looking for a DS3 PC Card... it would have 672 channels but the system doesn't need to handle but 20% of them at one time. If you're just talking 150 calls, you could do that with two 4x port cards in a single PC. I thought you were talking a lot bigger. ==I mean DS3 with 672 channel paths. There are 672 subscribers out there. I am saying that only a percentage of them are talking at peak times. We need to supervise 672 lines and expect 15% to talk at the same time. Each line/call would have to have absolute caller ID. In other words, PSTN call handling. Ummm, there's no such thing as absolute caller ID. You wanna try that question again? callerID is not legally binding, is not used by billing, anybody can spoof it. ===I mean we have to provide service and know what line is calling, not just provide anonymous service to a lot of people. We can't just mux a bunch of lines in to the Asterisk box with no identification. Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with HUD Lite
Has anyone used HUDlite recently and got it operating with Open Source Asterisk 1.6 or 1.8? I have read the instructions on HUDLite but it appears that it is only suited to Fonality versions like Trixbox. I would like to test HUDLite as a presence panel. If there are other options we are open to this? Rupert Utteridge -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with HUD Lite
Forget about HUDlite you want iSymphony, http://www.getisymphony.com/ On Sun, Nov 7, 2010 at 5:02 PM, Rupert Utteridge rupe...@dtasia.com.auwrote: Has anyone used HUDlite recently and got it operating with Open Source Asterisk 1.6 or 1.8? I have read the instructions on HUDLite but it appears that it is only suited to Fonality versions like Trixbox. I would like to test HUDLite as a presence panel. If there are other options we are open to this? Rupert Utteridge -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trixbox/Asterisk integration With SugarCRM
Hello All, i have one simple Question regarding integration of asterisk into sugar crm whether using trixbox or normal asterisk, can anyone have any link , forum or tutorial where i can find some information and some starting point . any help appreciated regards Dhaval -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VAD in asterisk
hey, I want to ask whether VAD is the asterisk functionality or softphones's functionality. Because I am using speex and zoiper but configuring VAD=true in codecs.conf does not suppress silence .. Thank in advance for help :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI SUBSCRIBE Settings
Hello list members, We're trying to get MWI notifications on our ATA device and we set it to send SUBSCRIBE messages to Asterisk, but it gets UNAUTHORIZED messages, despite the fact that we set the following lines in its settings in sip.conf: subscribemwi=yes mailbox...@from-extensions We need help in understanding how this works and what we are doing wrong. This is the SIP debug we get: --- SIP read from UDP:10.0.0.4:5090 --- SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: sip:2...@10.0.0.4:5090 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 - --- (13 headers 0 lines) --- Creating new subscription Sending to 10.0.0.4 : 5090 (no NAT) list_route: hop: sip:2...@10.0.0.4:5090 Found peer '21' for '21' from 10.0.0.4:5090 --- Transmitting (no NAT) to 10.0.0.4:5090 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=ePBX, nonce=26c43866 Content-Length: 0 Scheduling destruction of SIP dialog '055f7edd4081e1ec0f176e0a4b395...@10.0.0.4' in 6400 ms (Method: SUBSCRIBE) --- SIP read from UDP:10.0.0.4:5090 --- SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: sip:2...@10.0.0.4:5090 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 - --- (13 headers 0 lines) --- Ignoring this SUBSCRIBE request Found peer '21' for '21' from 10.0.0.4:5090 --- Transmitting (no NAT) to 10.0.0.4:5090 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=ePBX, nonce=26c43866 Content-Length: 0 Scheduling destruction of SIP dialog '055f7edd4081e1ec0f176e0a4b395...@10.0.0.4' in 6400 ms (Method: SUBSCRIBE) --- SIP read from UDP:10.0.0.4:5090 --- SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: sip:2...@10.0.0.4:5090 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 - --- (13 headers 0 lines) --- Ignoring this SUBSCRIBE request Found peer '21' for '21' from 10.0.0.4:5090 --- Transmitting (no NAT) to 10.0.0.4:5090 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=ePBX, nonce=26c43866 Content-Length: 0 --- SIP read from UDP:10.0.0.4:5090 --- SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: sip:2...@10.0.0.4:5090 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 - --- (13 headers 0 lines) --- Ignoring this SUBSCRIBE request Found peer '21' for '21' from 10.0.0.4:5090 --- Transmitting (no NAT) to 10.0.0.4:5090 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=ePBX, nonce=26c43866
