Re: [asterisk-users] Polycom firmware 4.0.1 and paging
Mike. Yes sip.ld is the firmware. I wanted to jump in because i saw you had the phantom ringing problem as well. I am running 3.3.1 and thought upgrading to 3.3.2 would solve that problem did you still have the problem in 3.3.2? I thought I saw in the release notes for 3.3.2 that was resolved. I dont have them infront of me but i suppose it is time to double check as I plan on upgrading 30 phones in the morning. I did test 3.3.2 but the phantom ring seemed so rand i thought i could just no reprouduce it. Thanks!! Jim - Original message - > It does update the sip.ld file, yes. So does all upgrades, no? > > > > Mike > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny > Nicholas Sent: Friday, February 10, 2012 5:39 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging > > > > Did the 4.0.1b update overwrite sip.ld on these phones? If I recall > correctly you have to tweak that file to make auto-answer work correctly. > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt > Sent: Friday, February 10, 2012 4:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging > > > > > > On Fri, Feb 10, 2012 at 10:30 PM, Mike wrote: > > Hi, > > > > I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer > simply stopped functioning. I can downgrade and make it work, upgrading > kills it again. There obviously is a difference in how the newer > firmware is treating this auto answer sip header. > > > > Can anybody tell me if they have Polycom firmware 4.x.x working with > auto-answer/paging? Just so I know it's worth my time to investigate, as > opposed to knowing it`s a Polycom firmware bug? If so, did you have to > make any changes to the SIP header sent to make Polycom phones auto > answer? > > > > Regards, > > > > Mike > > > > > > > > Hi Mike, > > > > Is there a compelling reason to put version 4.0.1b on these phones? > > > > Brian > > > > > > > > > > > > > -- > _ > -- 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] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings
Are you by chance using templates (!) In your sip.con? Ive had access denied errors befor when running as non root. - Original message - > I've just upgraded from 1.8.8.0 to 10.1.0-rc2. Now I'm getting a flood > of: > > WARNING[5100]: db.c:295 ast_db_put: Couldn't execute statment: SQL logic > error or missing database > > AFAIK, I'm not doing any database puts (or gets). There were no such > warnings in 1.8.8.0. > > What do I need to do to silence these warnings? > > 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 -- _ -- 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] allowguest = yes? no?
What they are talking about is SIP URI dialling. Let say you have extension 1000 the rings a phone on your system. With allowguest=yes I would be allowed to dial SIP:/1...@yourdomain.com and assuming the context defined in your [General] section had access to exten 1000 I would connect to that phone. With alloweguest=no my call would be rejected. That does not mean that strangers can not call an IVR and get to your 1000 extension or even a DID that point right to it. If you are going to allowguest=yes you need to take carfule note of your contexts so as not to allow strangers access to parts of your dial plan that have, lets say long distance routes. Does that help? Thanks!! Jim On 01/24/2012 09:34 AM, Gilles wrote: Hello I don't understand how I should use the "allowguest" item: If set to "yes", callers from the Net should authenticate, but then, how can I allow strangers to call extensions in my system? "allowguest If set to no, this disallows guest SIP connections. The default is to allow guest connections. SIP normally requires authentication, but you can accept calls from users who do not support authentication (i.e., do not have a secret field defined).Certain SIP appliances (such as the Cisco Call Manager v4.1) do not support authentication, so they will not be able to connect if you set allowguest=no: allowguest=no|yes" (from "Asterisk – The future of Telephony") Thank you. -- _ -- 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] Force CDR to be written.
Is there a way to Force the CDR data to be written prior to Hanging up the channel? Thanks!! Jim -- _ -- 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] Sip Registration Hijacking
Rate limiting (google) via iptables FTW! Good luck! - Original message - > > > Alejandro Imass wrote 20.01.2012 18:09: > > > I would like to know how > to block this MF because he makes calls at 1-2 AM > > I use this > construction on my servers > > [users] > > exten => > _XXX,1,GotoIfTime(1:00-2:00,*,*,*?block,1,1) > > [block] > exten => > _X.,1,HangUp(1) > > -- > With Best Regards > Mikhail Lischuk > > -- _ -- 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] Call abandoned from queue not showing in CDR (possible bug)
I think in your cdr.conf you are looking for the unanswered= directive. Thanks!! Jim - Original message - > Hi > > I'm using 1.8.7.0 with the RealTime architecture. > > If a call goes into application Queue and is abandoned by the caller, no > entry is made in the CDR. Entries are made into the queue log. > > This cannot be correct behaviour, all calls should show in the CDR. > > Could anyone else try to reproduce this and if others get the same > thing, I'll raise a bug on it. > > Thanks > > Ish > -- > Ishfaq Malik > Software Developer > PackNet Ltd > > Office: 0161 660 3062 > > > -- > _ > -- 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] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
I think the wiki may have just missed func_curl. I have a couple 1.8.x machines with working func_curl. Have you tried to compile it anyway? Thanks!! - Original message - > Hi, > > I've seen that function CURL is missing from 1.8 but back in with 10 > (see wiki.asterisk.org). > > With asterisk 1.8 and above, for a custom CID Name lookup application, > which is the most efficient way to send an HTTP GET from the dialplan > and parse its response (code and content) ? > > Regards > > -- > _ > -- 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] Best non polycom SIP conference room phone
You got me. At first the polycom world was hard to get into. But with a little effort to understand the configs and the joys of central provisioning the Polycom are my go to endpoint. Couple the endless configurablity with Polycom quilaty and I have many happy clients. As an aside is that what they use on the dCap? I have been meaning to get that when time allows. Thanks!! Jim - Original message - > "Luke Hamburg" writes: > > > Carlos- > > Sorry if this is too much of a digression but this piqued my interest > > as I've been pretty happy with Polycom in my limited experience > > (haven't used the SPAs much, just Yealink & Polycom, and an occasional > > Snom here and there). If the config files were not the issue for > > you, then what _were_ the problems? > > "A button has been pressed. Polycom must reboot for the change to take > effect. Reboot now (Y/N)?". Yes it's a recycled Windows joke, but it > applies much better to Polycom than it did to Windows. It is IMHO a bit > mean to use Polycom's in the Asterisk exam; the difficulty of passing > the exam is quite high if you haven't worked with them before. Pretty > much anything else is quicker to get to basic working state. > > Of course, once you get provisioning working they are excellent phones. > > > /Benny > > > -- > _ > -- 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] Question on system command 1.4.43
Does the user Asterisk is running as have access to this device?
Thanks!!
- Original message -
> I have a USB to serial converter attached to my box. pl2303: Prolific
> PL2303 USB to serial adaptor driver
> if I login to the box and send/receive serial commands over this unit it
> works without error EVERY time.
>
> however, if I run the same command set from with-in the extensions.conf
> with System()
> I get errors in dmesg like "pl2303 ttyUSB0: pl2303_open - failed
> submitting read urb, error -22"
> and then obviusly my command does not work.
>
> however, again - If I go back to the command line and my command it runs
> just fine.
>
> Any ideas on what might be happening here?
>
> Jerry
>
> extensions.conf
> exten => s,1,ChanIsAvail(Console/Dsp)
> exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
> exten => s,n,System(/home/silentm/bin/usbserial -start)
> exten => s,n,Playback(beep)
> exten => s,n,Dial(Console/dsp)
> exten => s,n,Hangup
> exten => h,1,System(/home/silentm/bin/usbserial -stop)
>
>
>
> --
> _
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Re: [asterisk-users] Problem w/ PC port on Polycom 335
Agreed. Check the switch for some kind of port security. Most of the time this would disable the interface if more than one MAC is present but you never know. Are there blinky lights on the pc? Also if provisioning via some sort of server check the MAC-boot log that the pgone uploads. Good Luck!! Thanks!! Jim. - Original message - > > Mike Diehl wrote: > > Usually, it just works... > > > > Any ideas? > > I've seen this before. > > One of our facilities have 'smart or managed' switches that have caused > no ends of problems, including preventing computers plugged into the > phones not having network access. > > You may want to review your switches. > > Doug > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little > Temporary Safety, deserve neither Liberty nor Safety." > > > -- > _ > -- 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] func_odbc not returning whole smalldatetime MS Sql field.
Hey All,
Odd thing. I am just trying to return the whole date time stamp from a
SMALLDATETIME field in a MS SQL server.
func_odbc.conf = readsql=SELECT DateCreated FROM [REDACTED] WHERE Code
= '${ARG1}'
Problem is I only get the first 15 back from the field. Like so...
Connected to Asterisk 1.8.6.0 currently running on [REDACTED]-dev (pid =
2240)
Verbosity is at least 3
[REDACTED]-dev*CLI> odbc read ODBC_[REDACTED]-LOOKUP 104809 exec
DateCreated 2011-12-19 13:2
Returned 1 row. Query executed on handle 0 [asterisk-mssql-connector]
Notice how it only returns "2011-12-19 13:2" and not the rest of the
time... I have run the query on the SQL server and then from isql and it
works everytime leaving the only abstraction point Asterisk.
Any thoughts?
Thanks!!
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Re: [asterisk-users] Recording the calls
Perhaps the Monitor CMD is what you are looking for. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor Good Luck!! Jim bilal ghayyad wrote: > Hi All; > > I need to use the recording for the calls, did anyone try this on Asterisk? > How it works? > > By the way: Asterisk support recording or it is another module that I have to > download it and install it? Stable? > > Regards > Bilal > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed
I work for a large health care company who does all their dealings with Cisco and the like. Would not even think of an open source solution for their telephony needs. The whole company had a need for an emergency notification system that would be triggered by a phone call and start paging both traditional pagers and blackberries. One of the other regions in our company went with some vendor supplied solution for a cost upwards of $175,000. I started thinking about it and created a clandestine asterisk system. With asterisk as the backbone to accept the phone call I created a database and some simple scripts to handle paging pagers and blackberries. All for the cost of a server and a digium card. The system is currently in the process of getting approved to be on the network (as it is not a cisco voip gateway ;-) ) Just thought i'd share. Have a good one all!! Jim John Todd wrote: > I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on > Sunday in Los Angeles, and the topic of my talk is "Open Source in an > Economic Downturn". I've got lots of talking points for this talk, > but it would be interesting to hear some short anecdotes about how you > in the Asterisk community are thriving, or at least surviving, by > virtue of the benefits of Open Source. I find that real-world > examples are worth more than all of the bullet points in the world, > and timely stories from the community would be more interesting than > hearing me prattle on. > > Please ensure that your snippet or list of points are in some way > related to the benefits of open-source, or how other alternatives are > less attractive in the "compressed economic environment." I'd > prefer of course to hear about how Asterisk is the silver bullet for > your particular business, but I'm open to any OSS-based solution being > a tool for you at this point. > > Send your comments publicly or privately - let me know if you want to > remain anonymous, otherwise I'll give you free advertising by using > your name or company name in my talk if I use your story. > > PS: Of course, the talk/slides will be available with Creative Commons > Attribution-Noncommercial-Share Alike 3.0 United States License, and I > expect that I'll probably use all or parts of this talk a few times > this year, given the focus on the economy. > > JT > > --- > John Todd email:jt...@digium.com > Digium, Inc. | Asterisk Open Source Community Director > 445 Jan Davis Drive NW - Huntsville AL 35806 - USA > direct: +1-256-428-6083 http://www.digium.com/ > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security issue
What distribution are you using? Below is a tutorial from the ubuntu
site but it should give you the basics of setting up iptables rules. I
have created custom rules for all my servers and the amount of junk
traffic has been dramatically reduced.
Good Luck!!
https://help.ubuntu.com/community/IptablesHowTo
Jim
Eric Fort wrote:
> use IP tables and start with deny all. Follow this by allowing only
> the protocols/ports you want and only the source/destination ip's you
> wish to allow. these can be combined to say allow ssh from anywhere
> but only allow sip (and it's range of ports) to/from a very limited
> set of ip's belonging to say your ITSP. for users that move about a
> bunch they can use vpn to an allowed subnet.
>
> Eric
>
> On Sat, Feb 7, 2009 at 5:47 PM, oumar ndiaye wrote:
>
>> David,
>> Thanks in advance. Where do I change the user/peers definition? Is it in the
>> firewall of the OS? In that case that won't work because the server host
>> other services such as ssh http that are open to any IP as long as the user
>> has the correct credentials. Doesn't asterisk itself has built in security
>> filters?
>>
>> If the only choice is to do in the OS's firewall, then I will need to
>> include the port numbers of SIP, IAX in my firewall rules. In this case,
>> which ports should I block to keep unwanted SIP/IAX connections from
>> specific IP's.
>> Thanks.
>>
>> On Sat, Feb 7, 2009 at 9:29 AM, David fire wrote:
>>
>>> you have many options but you should use it together.
>>> firewall
>>>
>>> in the user/peers definitions add host=
>>> and/or
>>> deny=0.0.0.0/0.0.0.0
>>> permit=/
>>>
>>> change the ip of your server.
>>>
>>> use something like ossec to avoid force brute.
>>>
>>> David
>>>
>>> 2009/2/6 oumar ndiaye
>>>
Is there a way to restrict connection to my asterisk server to users
based on their IP addresses, and not just password. I have some hackers who
connect to my server to make illegitimate solicitation calls to people. I
had to shutdown the server for now until I find a solution. ANY HELP?
Thanks.
ond
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>>>
>>> --
>>> (\__/)
>>> (='.'=)This is Bunny. Copy and paste bunny into your
>>> (")_(")signature to help him gain world domination.
>>>
>>>
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>>>
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>>>
>>
>> --
>> Oumar Ndiaye
>> CTO
>> ANTG Telecom
>> www.antg.com
>> ondi...@antg.com
>> ondi...@alum.mit.edu
>> ond4...@gmail.com
>> Tel: +1-919-291-8742
>>
>>
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>>
>>
>
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