Re: [sipx-users] Strange behavior in dial plan regarding permissions

[Tony Graziano]

Since the Asterisk box is setup as a user also "I think" it creates a
circular logic which might actually defeat dial plan permissions. You
might consider a different way to connect to the Asterisk box. (hint:
the askozia article shows a method that might be compatible using an
SBC, which is why I suggested it.)

I would consider posting to the sipx-dev list. I am not sure it would
be "fixed" if considered a bug in 4.4 (instead maybe just for 4.6
though).

On Thu, Jan 10, 2013 at 9:20 AM, Henry Dogger  wrote:
> I have seen this interesting setup, but since we are also developing on 
> asterisk as we do on sipXecs, we would lose this option.
> But it really shouldn't behave like this in my opion, since I can also 
> imagine using such a dial rule when using just sipXecs
> Any thoughts on the bug/problem?
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org 
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
> Sent: donderdag 10 januari 2013 14:06
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Strange behavior in dial plan regarding permissions
>
> I think this would behave differently using asterisk as a gateway.
>
> Have you considered this?
> http://wiki.sipfoundry.org/display/sipXecs/ACD+solution+based+on+Askozia
>
> Since it passes through a SBC it should not be required to make the dialplan 
> adjustments you are using.
>
> On Thu, Jan 10, 2013 at 7:59 AM, Henry Dogger  wrote:
>> Hi all,
>>
>>
>>
>> We stumbled some time ago on a strange behavior in the dial plan
>> regarding the dial permissions.
>>
>> The situation is as follows:
>>
>>
>>
>> We have a few dial plan rules e.g.
>>
>> -  Mobile phones (required is the mobile call permission)
>>
>> -  Local numbers (required is the local call permission)
>>
>> -  International (required is the international call permission)
>>
>>
>>
>> This all works as aspected, a user without the mobile call permission
>> is not allowed to call mobile phones.
>>
>> But part of our normal setup is a SIP connection between a sipXecs and
>> a Asterisk, calls are being routed from asterisk to sipXecs and the
>> other way around. (the reason why we use an Asterisk is because of the
>> queue functionality, ACD in sipXecs is not satisfying and also openACD
>> is still not good enough for us.)
>>
>> Since registering the asterisk as a user on sipXecs is a problem we
>> decided to create a dial rule in the dial plan with a (to all users on
>> the system) unknown prefix (e.g. 666).
>>
>> So the custom dial rule we created is 666 and 10 digits will result in
>> a dial of the last 10 digits on the gateways configured for outbound calls.
>>
>> The problems we get with this dial rule are:
>>
>> -  The rule has to be on top of the other outbound dial rules
>> (Mobile, Local and International in this example) to work, otherwise
>> sipXecs responds with a unauthorized to Asterisk.
>>
>> -  When this rule is active, all other outbound dial rules (Mobile,
>> Local and International in this example) can be called by all users,
>> even the users without the desired call permissions, so somehow this
>> rule breaks the entire permissions system
>>
>>
>>
>> I am curious if this is normal behavior, or did we stumble upon a bug?
>>
>> We are currently running on 4.4 updated till patch 16.
>>
>>
>>
>> Kind regards,
>>
>>
>>
>> Henry Dogger
>>
>> Telecats BV
>>
>>
>>
>>
>> ___
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>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
>
>
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Re: [sipx-users] One way audio with attended transfers to outside lines

[Tony Graziano]

Can you blind transfer the call from a different User Agent (i.e. Polycom
not bria)? Are you sure the ITSp is sending the calls in via port 5080? Can
you call the auto attendant from the outside and transfer a call? If you
cant call the AA and transfer to an extension they are probably sending the
call to port 5060 and this will always be an issue.

Since you can successfully transfer the call but just NOT with audio, it is
either a re-invite or firewall issue. Does the ITSP support reinvite with
SDP, if not this will also be an issue.

Firewalls must support symmetrical port NAT with no SIP ALG helper turned
on. Some firewalls do not have a symmetrical port NAT option but do support
1:1 NAt which gets you the same effectve solution (outbound traffic is
allowed but inbound traffic needs to be specified.

You also failed to mention if the BRIA user is local or remote. You never
said what type of firewall you are using.

To test you should consider an ITSp account with voip.ms if your ITSp is
not forthcoming with answers.
On Jan 9, 2013 11:26 AM, "Geoff Musgrave" 
wrote:

>  Example: Outside caller calls in to sipx. Call is answered by Bria 3.4.4
> user. Bria user performs a “Call first” (attended) transfer to sales
> person’s cell phone. This places the outside caller on hold and they hear
> hold music. The Bria user speaks to the sales person and then chooses
> “Transfer now” within the Bria application. Bria user see’s “Transfer
> Successful” and moves on. The sales person can hear the outside caller but
> the outside caller cannot hear the sales person.
>
> ** **
>
> Any of that make sense??
>
> ** **
>
> So I suspect my firewall or my ITSP but I can’t find anything in the
> firewall that would cause this as it was working last week. I’m testing
> with my ITSP in about a half hour and if we get anywhere I’ll pass along
> those results. But all the captures we’ve done so far don’t show any
> problems. 
>
> ** **
>
> I’m posting here to get any insight into the matter that I may be
> overlooking.
>
> ** **
>
> Sipx 4.6 is behind a firewall and NAT and all internal users are behind
> NAT as well. Yum update says there are no packages to update so I assume
> I’m on the current/stable version of 4.6.
>
> ** **
>
> Our ITSP did recently require us to have a Gateway setup for inbound and a
> separate one for outbound however that was a few weeks ago.
>
> ** **
>
> Thanks in advance and please let me know of any suggestions or if you
> would like any additional information.
>
> ** **
>
> --
>
> Geoff 
>
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>

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Re: [sipx-users] Strange behavior in dial plan regarding permissions

[Tony Graziano]

I think this would behave differently using asterisk as a gateway.

Have you considered this?
http://wiki.sipfoundry.org/display/sipXecs/ACD+solution+based+on+Askozia

Since it passes through a SBC it should not be required to make the
dialplan adjustments you are using.

On Thu, Jan 10, 2013 at 7:59 AM, Henry Dogger  wrote:
> Hi all,
>
>
>
> We stumbled some time ago on a strange behavior in the dial plan regarding
> the dial permissions.
>
> The situation is as follows:
>
>
>
> We have a few dial plan rules e.g.
>
> -  Mobile phones (required is the mobile call permission)
>
> -  Local numbers (required is the local call permission)
>
> -  International (required is the international call permission)
>
>
>
> This all works as aspected, a user without the mobile call permission is not
> allowed to call mobile phones.
>
> But part of our normal setup is a SIP connection between a sipXecs and a
> Asterisk, calls are being routed from asterisk to sipXecs and the other way
> around. (the reason why we use an Asterisk is because of the queue
> functionality, ACD in sipXecs is not satisfying and also openACD is still
> not good enough for us.)
>
> Since registering the asterisk as a user on sipXecs is a problem we decided
> to create a dial rule in the dial plan with a (to all users on the system)
> unknown prefix (e.g. 666).
>
> So the custom dial rule we created is 666 and 10 digits will result in a
> dial of the last 10 digits on the gateways configured for outbound calls.
>
> The problems we get with this dial rule are:
>
> -  The rule has to be on top of the other outbound dial rules
> (Mobile, Local and International in this example) to work, otherwise sipXecs
> responds with a unauthorized to Asterisk.
>
> -  When this rule is active, all other outbound dial rules (Mobile,
> Local and International in this example) can be called by all users, even
> the users without the desired call permissions, so somehow this rule breaks
> the entire permissions system….
>
>
>
> I am curious if this is normal behavior, or did we stumble upon a bug?
>
> We are currently running on 4.4 updated till patch 16.
>
>
>
> Kind regards,
>
>
>
> Henry Dogger
>
> Telecats BV
>
>
>
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/



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Re: [sipx-users] SIPXCDR errors

[Tony Graziano]

Do you show any "ongoing calls" in CDR history (active calls)? Have
you restarted ONLY CDR service yet? There were issues with this in 4.4
and these were fixed but this was before patch #16. I think you might
be in store for an update. It would be helpful to know if these errors
show on successful or failed calls too, since this was an issue once
before with failed calls only.



On Wed, Jan 9, 2013 at 6:24 AM, Elwin Formsma  wrote:
> Hi Douglas,
>
> On this system 4.4 patch #16
> Stunnel from SipXecs.
>
>
> Kind regards,
> Met vriendelijke groet,
>
>
> Elwin Formsma
> Telecats BV
> -
> Elwin Formsma | Telecats bv | KvK Enschede 06069106 | Tel:   053 488 99 44 | 
> Fax: 053 488 99 10 | E-mail: e.form...@telecats.nl |
>
> Op 9 jan. 2013, om 12:14 heeft Douglas Hubler  het 
> volgende geschreven:
>
>> On Wed, Jan 9, 2013 at 6:09 AM, Michael Picher  wrote:
>>> Elwin,
>>>
>>> Did you try to re-index the database?
>>>
>>> There's a trick to do this... in the search box in the upper right, just
>>> enter some junk.  On submit of the search a button will appear to re-index
>>> the database.
>>
>> Mike, i don't think that will help, that's a lucene index and it's on
>> an entirely different database SIPXCONFIG not SIPXCDR
>>
>> Elwin, this 4.4 system has has yum update, right?  Also be sure you've
>> installed stunnel that comes w/sipxecs and not from centos5.
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Re: [sipx-users] Certain calls failing (not sipx related exactly)

[Tony Graziano]

This is toll capacity related.

Certain localities (LEC's) can charge a premium to land calls in their
market space. As a result carriers reduce their call capacity to these
areas. This results in calls that just wont compete based on the cost
routing from your ITSP. For example, I have always seen issues making
calls to certain exchanges in Minnesota using carrier "xyz", but it
always works from my cell phone.

Not at all sipx related. Get thee to a carrier who has better peering
for toll calls or suffer silently.

On Tue, Jan 8, 2013 at 12:50 PM,   wrote:
> In fact, it’s likely not sipx related at all but I figured I’d ask here.
>
> We get complaints now and then that certain calls never work. One good
> example would be when people post stuff on Craigslist and that CL complains
> that certain voip numbers don’t work. In fact, it is looking like most don’t
> with voip.ms and flowroute.
>
> Wondering if anyone has any knowledge about this and if there is any way of
> fixing the problem, perhaps via the provider or some other means.
> The calls don’t seem to even reach sipx so doesn’t seem to be sipx related?
>
> Thanks.
>
>
>
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Re: [sipx-users] SIPXCDR errors

[Tony Graziano]

If it matters, this is a postgres complaint. I think you would do well
to try to restart CDR and if that doesnt clear it up dig into the
postgres forums. I would suggest if a lot fo your calling behavior
consists of forwards then it is likely a duplicate call id might try
to store and postgres would complain. Looking more directly at
postgres will probably pinpoint this.

On Tue, Jan 8, 2013 at 11:27 AM, Douglas Hubler  wrote:
> y, sounds very familiar, sug. you google it
>
> On Tue, Jan 8, 2013 at 6:10 AM, Elwin Formsma  wrote:
>> Hi,
>>
>> The following errors occur in the sipxcallresolver LOG. Sipxecs 4.4. Is this 
>> a known issue?
>>
>>
>> "2013-01-08T11:05:01.033231Z":20:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>> "2013-01-08T11:05:01.040001Z":21:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>> "2013-01-08T11:05:01.139580Z":22:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>> "2013-01-08T11:05:21.199918Z":23:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>> "2013-01-08T11:05:31.100696Z":24:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>> "2013-01-08T11:05:51.331098Z":25:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>> "2013-01-08T11:06:31.349482Z":26:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>> "2013-01-08T11:06:31.367072Z":27:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>> "2013-01-08T11:06:41.400521Z":28:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>> "2013-01-08T11:06:51.405508Z":29:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>> "2013-01-08T11:06:51.419459Z":30:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>> "2013-01-08T11:06:51.425822Z":31:CDR:ERR:xxx.nl:main::cdr:", ERROR:  
>> null value in column \"from_tag\" violates not-null constraint"
>>
>>
>> Kind regards,
>> Met vriendelijke groet,
>>
>>
>> Elwin Formsma
>> Telecats BV
>> -
>> Elwin Formsma | Telecats bv | KvK Enschede 06069106 | Tel:   053 488 99 44 | 
>> Fax: 053 488 99 10 | E-mail: e.form...@telecats.nl |
>>
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Re: [sipx-users] VIP extension

[Tony Graziano]

By the same token you might be able to craft a rule (dial plan) to redirect
or null process that destination. It would be simpler to turn the ringer
off and simply ignore the calls and add the line to an admin assistant who
would field answering those calls.

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Re: [sipx-users] Incoming calls to SNOM disconnect after 30 seconds or so.

[Tony Graziano]

You should disable gruu on the snom.
On Jan 7, 2013 10:24 AM, "Sven Evensen"  wrote:

> Thanks for a few good tips Tony. The outbound calls work fine, it is only
> the inbound call which fails. It is because sipxProxy is relaying ACK to
> the internal IP of the phone, wrong of course. I have looked at the
> REGISTER and the only field I can see which differs from the soft phone
> (which works fine)  is "X-Real-IP" which does contain the internal IP
>
> REGISTER sip:ec2-50-18-193-48.us-west-1.compute.amazonaws.com SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.110:2048;branch=z9hG4bK-ccn6pkaxxrrw;rport
> From: "Dwayne Kee" <
> sip:11...@ec2-50-18-193-48.us-west-1.compute.amazonaws.com>;tag=rgiklqr928
> To: "Dwayne Kee" <
> sip:11...@ec2-50-18-193-48.us-west-1.compute.amazonaws.com>
> Call-ID: 386d43a0edbb-qfxd1lgiuetf
> CSeq: 2 REGISTER
> Max-Forwards: 70
> Contact:  ;line=v2kaaktt>;q=1.0;reg-id=1;+sip.instance=">";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
> User-Agent: snom300/8.7.3.15
> Allow-Events: dialog
> X-Real-IP: 192.168.1.110
> Supported: path, gruu
> Authorization: Digest username="11404",realm="
> ec2-50-18-193-48.us-west-1.compute.amazonaws.com
> ",nonce="448038dd4fbe27452d637428dc1a565950e76936",uri="sip:
> ec2-50-18-193-48.us-west-1.compute.amazonaws.com
> ",qop=auth,nc=0001,cnonce="35993a6f",response="ad299e3bc739b3d5882f39ea7a64e04f",algorithm=MD5
> Expires: 3600
> Content-Length: 0
>
>
> On Mon, Jan 7, 2013 at 2:39 PM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> it might be more pertinent to post a sanitized config of the phone. As
>> I recall the snom's dont configure "the same" as a polycom (SRV
>> records, etc.). I also know there is a patch just committed for Snom
>> on 4.6, so you might ask the devs (Domenico authored the patch) about
>> what was changed, and why.
>>
>> I "think" you have to disable GRUU on snom. I am confused as to why
>> the snom says it is not behind NAT when the server is in EC3 or
>> wherever. The phone registration should show more helpful information.
>> It would appear (to me) the phone is not configured correctly and
>> sending its private address on outbound calls.
>>
>> On Mon, Jan 7, 2013 at 9:25 AM, Sven Evensen 
>> wrote:
>> > Our sipx 4.4 is running in Amazon, we have mostly mobile externsion (our
>> > product) and soft phones and a few IP phones. The Polycoms we have no
>> issue
>> > with. But we have now installed 3 SNOMs and all three have then problem
>> that
>> > incoming calls all disconnect after half a minute or so.Outbound calls
>> are
>> > fine
>> >
>> > From the trace I can see the 200 OK from the phone is relayed back to
>> the
>> > SIP trunk, then the ACK from the SIP trunk is not relayed back to the
>> phone,
>> > but to SipxRegistrar where it stops up.
>> >
>> > I have attached the sip trace for this. Also a sip trace for a soft
>> phone on
>> > the same user which does not have this issue.
>> >
>> > Also worth mentioning that one SNOMis configured from the template in
>> sipX,
>> > another (which is attached) is configured from the phone, in the same
>> was as
>> > a soft phone. Same issue on both phones.
>> >
>> > --
>> >
>> > Sven Evensen, Operations Consultant
>> >
>> > OnRelay
>> >
>> > +44 (0) 207 902 8123 │ mailto:sven.even...@onrelay.com │
>> www.onrelay.com
>> >
>> >
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>> > solely for the recipient and use by any other party is not authorised.
>> If
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>> > delete this message, along with any attachments, from your computer.
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>> >

Re: [sipx-users] Incoming calls to SNOM disconnect after 30 seconds or so.

[Tony Graziano]

it might be more pertinent to post a sanitized config of the phone. As
I recall the snom's dont configure "the same" as a polycom (SRV
records, etc.). I also know there is a patch just committed for Snom
on 4.6, so you might ask the devs (Domenico authored the patch) about
what was changed, and why.

I "think" you have to disable GRUU on snom. I am confused as to why
the snom says it is not behind NAT when the server is in EC3 or
wherever. The phone registration should show more helpful information.
It would appear (to me) the phone is not configured correctly and
sending its private address on outbound calls.

On Mon, Jan 7, 2013 at 9:25 AM, Sven Evensen  wrote:
> Our sipx 4.4 is running in Amazon, we have mostly mobile externsion (our
> product) and soft phones and a few IP phones. The Polycoms we have no issue
> with. But we have now installed 3 SNOMs and all three have then problem that
> incoming calls all disconnect after half a minute or so.Outbound calls are
> fine
>
> From the trace I can see the 200 OK from the phone is relayed back to the
> SIP trunk, then the ACK from the SIP trunk is not relayed back to the phone,
> but to SipxRegistrar where it stops up.
>
> I have attached the sip trace for this. Also a sip trace for a soft phone on
> the same user which does not have this issue.
>
> Also worth mentioning that one SNOMis configured from the template in sipX,
> another (which is attached) is configured from the phone, in the same was as
> a soft phone. Same issue on both phones.
>
> --
>
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>
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Re: [sipx-users] New Install Problem

[Tony Graziano]

I tend to think its a basic Linux install issue with your server hardware.
If you choose to do the minimal install consider doing it from the network
if you have sufficient bandwidth.

Did you query the Google oracle for its wisdom in these matters as it
relates to you hardware platform?
On Dec 27, 2012 1:25 PM, "Tommy Laino"  wrote:



Its only a single NIC tower. I will try the RPM install
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[sipx-users] Wiki Down!

[Tony Graziano]

Proxy Error

The proxy server received an invalid response from an upstream server.
The proxy server could not handle the request *GET
/<http://wiki.sipfoundry.org/>
*.

Reason: *Error reading from remote server*

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[sipx-dev] Wiki Down!

[Tony Graziano]

Proxy Error

The proxy server received an invalid response from an upstream server.
The proxy server could not handle the request *GET
/<http://wiki.sipfoundry.org/>
*.

Reason: *Error reading from remote server*

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Re: [sipx-dev] Query regarding deletion of default MOH file

[Tony Graziano]

IMO - It's OK to delete this file like any custom uploaded files if the
admin wants to do that.

On Fri, Dec 21, 2012 at 5:48 AM, rekha.h  wrote:

> Hi,
>
> Query regarding deletion of Default MOH file
>
> Steps to reproduce:
>
> 1. Login to superadmin
> 2. Navigate to Features-->Music on Hold
> 3. Default.wav file is present as the default MOH
> 4. Select the Default.wav file and click on delete.
> 5. The default.wav file will get deleted without giving any warning
> message as "You are about to delete the default wave file"
>
> Query is whether default MOH file should get deleted or not while trying
> to delete the file.
>
> Regards,
> Rekha
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Re: [sipx-dev] test message

[Tony Graziano]

This is a test message of the emergency broadcast system...

On Wed, Dec 19, 2012 at 1:22 PM, David Grazio  wrote:

> This is a test message
>
> ** **
>
> Best Regards,
>
> David Grazio
>
> VP of Product Marketing 
>
> eZuce Inc.
>
> 300 Brickstone Square
>
> Suite 201
>
> Andover, MA. 01810
>
> (978) 296-1005 x2016
>
> dgra...@ezuce.com
>
> Skype: David.Grazio1
>
> ** **
>
> ** **
>
> ** **
>
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Re: [sipx-users] ERROR: SipXbridge XML-RPC Exception

[Tony Graziano]

You need at least 4GB to make it behave properly when making configuration
changes.

On Tue, Dec 18, 2012 at 10:58 PM, Matt Nelson <
matthewe...@audiopivotpatc.com> wrote:

>  Is it suggested to do a yum update on this..cause that's what I did.
>
>
> On 12/18/2012 7:39 PM, Todd Hodgen wrote:
>
>  After you reset the Bridge, and until it is fully reset, you typically
> see this error.   Is this machine real slow possibly?
>
> ** **
>
> *From:* sipx-users-boun...@list.sipfoundry.org [
> mailto:sipx-users-boun...@list.sipfoundry.org]
> *On Behalf Of *Matt Nelson
> *Sent:* Tuesday, December 18, 2012 7:30 PM
> *To:* sipx-users
> *Subject:* [sipx-users] ERROR: SipXbridge XML-RPC Exception
>
> ** **
>
> Ok, got the web interface to come up. Now having issues in the sbc report
> area. Getting *
>
> *
>
> -- ERROR: SipXbridge XML-RPC Exception
>
> ** **
>
> any ideas?
>
> ** **
>
> Matt Nelson
>
> Owner/Technical Director
>
> Vitelity/Megapath/Yealink Partner
>
> audiopivotPATC
>
> LIC#10-00016519
>
> Malibu, California
>
> 424.781.1666 itsp/fax
>
> Yuma, Arizona
>
> 928.597.4777 itsp/fax
>
> matthewe...@audiopivotpatc.com
>
> ** **
>
> ** **
>
> "pure technical artistry"
>
>
>
> ___
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> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
>
>
> --
>
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> Owner/Technical Director
> Vitelity/Megapath/Yealink Partner
> audiopivotPATC
> LIC#10-00016519
> Malibu, California
> 424.781.1666 itsp/fax
> Yuma, Arizona
> 928.597.4777 itsp/faxmatthewe...@audiopivotpatc.com
>
>
> "pure technical artistry"
>
>
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>



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Re: [sipx-users] Sorry more info

[Tony Graziano]

cannot allocate memory... describe your VM configuration? I would allocate
at least 4GB before booting up.

On Tue, Dec 18, 2012 at 10:32 PM, Matt Nelson <
matthewe...@audiopivotpatc.com> wrote:

> sorry needed to add i got this in the log
>
> IO error. Could not complete agent command Cannot run program
> "/usr/bin/sipxagent": java.io.IOException: error=12, Cannot allocate
> memory null
>
> --
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> Owner/Technical Director
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> LIC#10-00016519
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> 424.781.1666 itsp/fax
> Yuma, Arizona
> 928.597.4777 itsp/fax
> matthewe...@audiopivotpatc.com
>
>
> "pure technical artistry"
>
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Re: [sipx-users] Registrations Expiring

[Tony Graziano]

If the port is open then it is vulbnerable. It doesn't matter whose
backbone it is on.

There are lots of ways to protect it though.

We are big fans of using mesh VPN's and removing that as an open port
altogether. We have found our chosen method scales pretty well.

In the meantime, your firewall might be able to limit connections per
second from an IP address to that port. With some firewall products, we do
that AND we block countries altogether (depending on the customer and their
geographic footprint). There are lots of things you can do but you HAVE to
do them or suffer the consequences.

On Sun, Dec 16, 2012 at 2:02 PM, Tommy Laino  wrote:

>
>
> Yes I know that 5060 is used for remote users. I had it
> opened because they are going to be deploying a remote sales
> team in a few months. They are using Comcast which has a
> shared pipe. I am wondering if that has anything to do with
> it. When I do remote deployments they almost exclusively use
> FiOS which is a dedicated pipe.
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Re: [sipx-users] Registrations Expiring

[Tony Graziano]

If you have it open, it means that you support remote users. If you do not
support remote users, it should not be open.

There are also numerous things you could do both onboard sipx and at your
firewall to limit the attempts at scripts and other malicious activity
aimed at your voip services in general.

On Sun, Dec 16, 2012 at 11:57 AM, Tommy Laino  wrote:

>
>
> The sipproxy.log file is so large that my text editors will
> not open it. I am assuming that it is an attack. I am going
> to have the IT department close that port on the firewall
> and see if we have any luck.
> --
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Re: [sipx-users] Registrations Expiring

[Tony Graziano]

Is port 5060 exposed in the firewall?  If so it is potentially a script
from outside trying to abuse your system you would need to inspect your
logs to verify.
On Dec 16, 2012 10:52 AM, "Tommy Laino"  wrote:

>
>
> For the last 2 weekends I have had a SipXecs 4.4 with 30
> Polycom 335's that has all the registrations expire. I have
> to resend the server profile and all the phones re-register.
> Weird thing is that this only happens over the weekend or
> when all the phones are inactive for a long period of time.
> Any ideas? I have the logs but not sure what I should be
> looking for to try and troubleshoot this
> --
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Re: [sipx-users] Subscribe forwarding 4.4.0-update #22

[Tony Graziano]

I'm a little lost. I thought this was going to be reverted altogether. I
find call park (cannot pickup) is broken in the last stage update.

So is it being worked on? Or reverted. I'd suggest to have it reverted
until a patch is tested and known to work... anyone using stage to test
against is stuck with a broken call pickup until then.

On Fri, Dec 14, 2012 at 9:01 AM, Elwin Formsma wrote:

>  Hi Joegen,
>
>  We have created an update for this patch to make BLF and Pick-up on
> Huntgroups possible. Please review attached patchfile and let us know what
> you think.
> This has been tested by us with succes. We might have overlooked some
> stuff though, so feedback is welcome.
>
>
>  Kind regards,
>  Met vriendelijke groet,
>
>
> Elwin Formsma
> Telecats BV
> -
> Elwin Formsma | Telecats bv | KvK Enschede 06069106 | Tel:   053 488 99 44
> | Fax: 053 488 99 10 | E-mail: e.form...@telecats.nl |
>
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Re: [sipx-users] Hunt group, and pickup issues

[Tony Graziano]

Did you follow the guidelines in the wiki on hunt groups?
On Dec 13, 2012 3:46 PM, "glomos-info"  wrote:

>  Hi,
>
> ** **
>
> 1st problem 
>
> If i create a hunt group containing some extensions ringing at the same
> time, then when i pick up a phone the connection is established, but sound
> is dead/silence. Then if i put the incoming/active line on hold and again
> off-hold audio comes up ok.
>
> ** **
>
> The strange part is that the same hunt group configured sequentially (not
> ringing at the same time) does not have this problem.
>
> ** **
>
> 2nd problem
>
> When i pick-up a line ringing on another phone with *87XXX the connection
> is established correctly but also no audio
>
> ** **
>
> Using 4.6 with all updates.
>
> ** **
>
> Any ideas?
>
> ** **
>
> GJ
>
> ** **
>
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Re: [sipx-users] Message Waiting Issue

[Tony Graziano]

Its not a big deal. Very glad it is working. This us one if those instances
where a change log would come in handy in deed.
On Dec 12, 2012 4:33 PM, "Tommy Laino"  wrote:

>
>
> Tony I am embarrassed lol I know better and I take full
> responsibility for my stupidity.
>
> I changed 2 phones and sent the profiles, dropped a VM in
> the box and sure enough MWI worked. I will be sending the
> remaining profiles tonite. Thanks guys
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Re: [sipx-dev] SDP NAT rewrite issue

[Tony Graziano]

Right. I was wondering overall how MOH and things work in conjunction with
the SBC itself. It would be nice to see "what" works with an SBC and what
does not (setting expectations).


On Wed, Dec 12, 2012 at 3:22 PM, Josh Patten  wrote:

> Tony,
>
> As the Sangoma SBC uses FreeSWITCH under the hood, this is the last
> remaining interop item that we have with them.
> http://wiki.sipfoundry.org/display/sipXecs/FreeSWITCH+SIP+Trunking+Gatewayis 
> the howto. I haven't posted interop testing results yet but the final
> issue is this particular forking issue. it's not a SIP signalling issue,
> it's the FreeSWITCH RTP engine that has an issue. Sangoma is working to
> wrap their head around where the issue lies.
>
>
> On Wed, Dec 12, 2012 at 11:33 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> I think it would be agreed it has a way to go. Other SBC's can be
>> problematic, so can some ITSP's.
>>
>> It would be good to see a use case and sample for the wiki once the
>> config stabilizes and things are tested.
>>
>> I don't have hairpinned issues with some SBC's or some ITSP's.it would
>> be good to know what itsp the sangoma has been tested against and how
>> things like MOH work or whether it has its onw (or any at all),
>>
>>
>> On Wed, Dec 12, 2012 at 12:09 PM, Josh Patten  wrote:
>>
>>> Sangoma uses FreeSWITCH under the hood.
>>>
>>> FreeSWITCH still has an issue with hairpinned forking that needs to be
>>> addressed. What happens is that when a 183 with SDP comes in from the
>>> outside system (such as when calling a cell phone) the media stream is
>>> established. When the 200 OK with SDP comes from a device other than what
>>> generated the 183, the media stream will still attempt to go to the device
>>> that issued the 183 until the stream is referred somewhere else, such as by
>>> placing the call on hold and retrieving it again.
>>>
>>>
>>> On Wed, Dec 12, 2012 at 10:36 AM, Tony Graziano <
>>> tgrazi...@myitdepartment.net> wrote:
>>>
>>>> Also, realize ANY SBC would be able to have its own dialplan and rules
>>>> to do this.
>>>>
>>>> So where is sangoma on this issue?
>>>>
>>>>
>>>>
>>>> On Wed, Dec 12, 2012 at 11:09 AM, Matt White 
>>>> wrote:
>>>>
>>>>> True, but when the call goes from the ITSP to the SANGOMA, the SBC
>>>>> (sangoma) should be modifying the INVITE with the private info of the
>>>>> SANGOMAie 192.168.10.19
>>>>>
>>>>> But its not, its sending the public ITSP packet into sipx rather than
>>>>> modifying it, which makes sipxproxy think its not a local SIP invite and
>>>>> sipxrelay jumps into action to take over.
>>>>>
>>>>> The point of the SBC is to hide the public info when sending it to the
>>>>> proxy and preform the header modification between the proxy and the ITSP.
>>>>>
>>>>> -M
>>>>>
>>>>>
>>>>>
>>>>> >>> Chris Rawlings  12/12/12 9:11 AM >>>
>>>>>
>>>>> this is the initial invite. This is where the call is comming from the
>>>>> ITSP -> WAN IP 24.229.51.68 -> NAT FIREWALL -> SBC IP 192.168.10.19
>>>>> (FreeSWITCH / SANGOMA)
>>>>>
>>>>> The Sangoma SBC then hands this off to the PBX later in the chain
>>>>> where it creates both an ITSP and PBX side of the call and bridges them.
>>>>>
>>>>> This invite needs to be NAT aware as the SBC is behind a NAT
>>>>> firewall.. all communications on Port 5060 inbound to 192.168.10.19 are
>>>>> ITSP communications. All communications on Port 5080 inbound to
>>>>> 192.168.10.19 are PBX communications.
>>>>>
>>>>> And then finally i was under the impression that at NO TIME will an
>>>>> unmanaged Gateway have a WAN IP address put into any portion of the SIP
>>>>> messaging
>>>>>
>>>>>
>>>>> On Tue, Dec 11, 2012 at 9:09 PM, Joegen Baclor wrote:
>>>>>
>>>>>>  The root of all evil is in the INVITE.   It advertised itself as
>>>>>> NATed by providing a public contact and a public via.  Why is the gateway
>>>>>> doing this?
>>>>>>
>>>>>>
>>>>&

Re: [sipx-users] Call Park Presence

[Tony Graziano]

 own.  I have seen this with Polycom firmwares
>>>> 3.2.4, 3.2.7 and yes 3.3.0 and 4.0.3(this trace).  I have a debug off the
>>>> adtran to send to them, but they asked me were it is failing and I just
>>>> don't know for sure myself.
>>>>
>>>> -Bryan Anderson
>>>>
>>>>
>>>>
>>>> On Fri, Dec 7, 2012 at 11:16 AM, Bryan Anderson wrote:
>>>>
>>>> Thanks for the reply and I will defiantly test it.  We use a T1 for
>>>> service into the Adtran and the Adtran is in SipXecs as an unmanaged
>>>> gateway.
>>>>
>>>> -Bryan Anderson
>>>>
>>>>
>>>>
>>>> On Fri, Dec 7, 2012 at 11:07 AM, Ali Ardestani >>> > wrote:
>>>>
>>>> This is how we implemented call park with polycom and it works (it is a
>>>> workaround though)
>>>>
>>>> 1. Extension 700 forwards to 701, 702, 703 and 703
>>>>
>>>> 2. added the below to the custom config of the phones (this is done so
>>>> that the key does not timeout after 1 minute and call the park orbit
>>>> directly
>>>> >>> call.offeringTimeOut="3600"
>>>> call.directedCallPickupMethod="legacy"
>>>> call.parkedCallRetrieveMethod="legacy"
>>>> >
>>>>
>>>> 3. subscribe to the presence of 700 on user speed dials
>>>>
>>>> 4. Make sure you use the bridge, we had problems with the call
>>>> unparking when we did not use the bridge for incoming calls from trunk
>>>> provider
>>>>
>>>> *5. Our firewal does ALG, so we had to uncheck "Use public address for
>>>> call setup" under Devices=>Gateways(choose the gw)=>ITSP Account, This
>>>> fixed our problem with the calls unparking, maybe your firewall is also
>>>> doing some form of ALG*
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Fri, Dec 7, 2012 at 10:06 AM, Bryan Anderson wrote:
>>>>
>>>> So I noticed some talk in a previous email "Call forward fails to
>>>> external number" about the Adtran 900 series.  I have a couple of comments
>>>> and questions.
>>>>
>>>> We have a TA908e 2nd gen running AOS A5.02.00.E.  We currently have
>>>> not noticed any issue with having an external caller forwarded to and
>>>> external number.  cell => user for 30sec => external number.
>>>>
>>>> What we have had issues with is presence monitoring of call parking.   We
>>>> have a Polycom Soundpoint IP 650 with a sing side car that monitors
>>>> park lines 6000 - 6003.  We can park calls no problem, and so far have
>>>> not had trouble retrieving calls.  Our problem is that once the call
>>>> gets retrieved from the call park the BLF never stops blinking.  I
>>>> have to restart the Park/Presence servers.
>>>>
>>>> This is with SipXecs 4.4.0.
>>>>
>>>> Thoughts and comments would be appreciated.
>>>>
>>>>
>>>> -Bryan Anderson
>>>>
>>>>
>>>> ___
>>>> sipx-users mailing list
>>>> sipx-users@list.sipfoundry.org
>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> --
>>>>  Ali S Ardestani
>>>> Telephony Systems Engineer
>>>> Private National Mortgage Acceptance Company (PennyMac)
>>>> 6101 Condor Drive
>>>> Moorpark, CA 93021
>>>>
>>>>
>>>
>>> --
>>> --
>>> Ali S Ardestani
>>> Telephony Systems Engineer
>>> Private National Mortgage Acceptance Company (PennyMac)
>>> 6101 Condor Drive
>>> Moorpark, CA 93021
>>>
>>> (805) 330-6004 Office
>>> (818) 224-7442 x2654 Office
>>> (626) 817-3512 Mobile
>>> (818) 224-7397 Fax
>>>
>>> ali.ardest...@pnmac.com
>>>
>>>
>>> ___
>>> sipx-users mailing list
>>> sipx-users@list.sipfoundry.org
>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>>
>>
>>
>> ___
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>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>
>
>
> --
> Josh Patten
> eZuce
> Solutions Architect
> O.978-296-1005 X2050
> M.979-574-5699
> http://www.ezuce.com
>
>
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> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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Re: [sipx-dev] SDP NAT rewrite issue

[Tony Graziano]

I think it would be agreed it has a way to go. Other SBC's can be
problematic, so can some ITSP's.

It would be good to see a use case and sample for the wiki once the config
stabilizes and things are tested.

I don't have hairpinned issues with some SBC's or some ITSP's.it would be
good to know what itsp the sangoma has been tested against and how things
like MOH work or whether it has its onw (or any at all),

On Wed, Dec 12, 2012 at 12:09 PM, Josh Patten  wrote:

> Sangoma uses FreeSWITCH under the hood.
>
> FreeSWITCH still has an issue with hairpinned forking that needs to be
> addressed. What happens is that when a 183 with SDP comes in from the
> outside system (such as when calling a cell phone) the media stream is
> established. When the 200 OK with SDP comes from a device other than what
> generated the 183, the media stream will still attempt to go to the device
> that issued the 183 until the stream is referred somewhere else, such as by
> placing the call on hold and retrieving it again.
>
>
> On Wed, Dec 12, 2012 at 10:36 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> Also, realize ANY SBC would be able to have its own dialplan and rules to
>> do this.
>>
>> So where is sangoma on this issue?
>>
>>
>>
>> On Wed, Dec 12, 2012 at 11:09 AM, Matt White wrote:
>>
>>> True, but when the call goes from the ITSP to the SANGOMA, the SBC
>>> (sangoma) should be modifying the INVITE with the private info of the
>>> SANGOMAie 192.168.10.19
>>>
>>> But its not, its sending the public ITSP packet into sipx rather than
>>> modifying it, which makes sipxproxy think its not a local SIP invite and
>>> sipxrelay jumps into action to take over.
>>>
>>> The point of the SBC is to hide the public info when sending it to the
>>> proxy and preform the header modification between the proxy and the ITSP.
>>>
>>> -M
>>>
>>>
>>>
>>> >>> Chris Rawlings  12/12/12 9:11 AM >>>
>>>
>>> this is the initial invite. This is where the call is comming from the
>>> ITSP -> WAN IP 24.229.51.68 -> NAT FIREWALL -> SBC IP 192.168.10.19
>>> (FreeSWITCH / SANGOMA)
>>>
>>> The Sangoma SBC then hands this off to the PBX later in the chain where
>>> it creates both an ITSP and PBX side of the call and bridges them.
>>>
>>> This invite needs to be NAT aware as the SBC is behind a NAT firewall..
>>> all communications on Port 5060 inbound to 192.168.10.19 are ITSP
>>> communications. All communications on Port 5080 inbound to 192.168.10.19
>>> are PBX communications.
>>>
>>> And then finally i was under the impression that at NO TIME will an
>>> unmanaged Gateway have a WAN IP address put into any portion of the SIP
>>> messaging
>>>
>>>
>>> On Tue, Dec 11, 2012 at 9:09 PM, Joegen Baclor wrote:
>>>
>>>>  The root of all evil is in the INVITE.   It advertised itself as
>>>> NATed by providing a public contact and a public via.  Why is the gateway
>>>> doing this?
>>>>
>>>>
>>>> INVITE sip:7175463...@blueuc.com SIP/2.0
>>>> Via: SIP/2.0/UDP 24.229.51.68:5080;rport;branch=z9hG4bKSSZZ7Ht02mHSN
>>>> Max-Forwards: 11
>>>> From: "+16107413324" 
>>>> 
>>>> ;tag=rFp8vK6FN17Bj
>>>> To:  
>>>> Call-ID: 62b9c0f2-be62-1230-4aa1-005056a433a6
>>>> CSeq: 37288972 INVITE
>>>> Contact:
>>>> 
>>>> User-Agent: NetBorder Session Controller
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY
>>>> Supported: precondition, path, replaces
>>>> Allow-Events: talk, hold, refer
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 191
>>>> X-Inbound: TRUE
>>>> X-Account-Code: None
>>>> X-Account-Inbound: 3587
>>>> P-Acme-VSA: 202:3587
>>>> P-Acme-VSA-1: 201:None
>>>> X-Device: 24.229.51.68
>>>> X-FS-Support: update_display
>>>> Remote-Party-ID: "+16107413324" 
>>>> 
>>>> ;party=calling;screen=yes;privacy=off
>>>>
>>>>
>>>> v=0
>>>> o=nsc 1355222135 1355222136 IN IP4 192.168.10.19
>>>> s=nsc
>>>> c=IN IP4 192.168.10.19
>>>> t=0 0
>>>> m=audio 27938 RTP/AVP 0 8 101 13
>>>&g

Re: [sipx-dev] SDP NAT rewrite issue

[Tony Graziano]

Also, realize ANY SBC would be able to have its own dialplan and rules to
do this.

So where is sangoma on this issue?



On Wed, Dec 12, 2012 at 11:09 AM, Matt White wrote:

> True, but when the call goes from the ITSP to the SANGOMA, the SBC
> (sangoma) should be modifying the INVITE with the private info of the
> SANGOMAie 192.168.10.19
>
> But its not, its sending the public ITSP packet into sipx rather than
> modifying it, which makes sipxproxy think its not a local SIP invite and
> sipxrelay jumps into action to take over.
>
> The point of the SBC is to hide the public info when sending it to the
> proxy and preform the header modification between the proxy and the ITSP.
>
> -M
>
>
>
> >>> Chris Rawlings  12/12/12 9:11 AM >>>
>
> this is the initial invite. This is where the call is comming from the
> ITSP -> WAN IP 24.229.51.68 -> NAT FIREWALL -> SBC IP 192.168.10.19
> (FreeSWITCH / SANGOMA)
>
> The Sangoma SBC then hands this off to the PBX later in the chain where it
> creates both an ITSP and PBX side of the call and bridges them.
>
> This invite needs to be NAT aware as the SBC is behind a NAT firewall..
> all communications on Port 5060 inbound to 192.168.10.19 are ITSP
> communications. All communications on Port 5080 inbound to 192.168.10.19
> are PBX communications.
>
> And then finally i was under the impression that at NO TIME will an
> unmanaged Gateway have a WAN IP address put into any portion of the SIP
> messaging
>
>
> On Tue, Dec 11, 2012 at 9:09 PM, Joegen Baclor  wrote:
>
>>  The root of all evil is in the INVITE.   It advertised itself as NATed
>> by providing a public contact and a public via.  Why is the gateway doing
>> this?
>>
>>
>> INVITE sip:7175463...@blueuc.com SIP/2.0
>> Via: SIP/2.0/UDP 24.229.51.68:5080;rport;branch=z9hG4bKSSZZ7Ht02mHSN
>> Max-Forwards: 11
>> From: "+16107413324" 
>> 
>> ;tag=rFp8vK6FN17Bj
>> To:  
>> Call-ID: 62b9c0f2-be62-1230-4aa1-005056a433a6
>> CSeq: 37288972 INVITE
>> Contact:
>> 
>> User-Agent: NetBorder Session Controller
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>> Supported: precondition, path, replaces
>> Allow-Events: talk, hold, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 191
>> X-Inbound: TRUE
>> X-Account-Code: None
>> X-Account-Inbound: 3587
>> P-Acme-VSA: 202:3587
>> P-Acme-VSA-1: 201:None
>> X-Device: 24.229.51.68
>> X-FS-Support: update_display
>> Remote-Party-ID: "+16107413324" 
>> 
>> ;party=calling;screen=yes;privacy=off
>>
>>
>> v=0
>> o=nsc 1355222135 1355222136 IN IP4 192.168.10.19
>> s=nsc
>> c=IN IP4 192.168.10.19
>> t=0 0
>> m=audio 27938 RTP/AVP 0 8 101 13
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>>
>>
>>
>>
>>
>> On 12/12/2012 05:10 AM, Josh Patten wrote:
>>
>> Sorry I forgot to put this info.
>> Unmanaged Gateway.
>>
>>
>> On Tue, Dec 11, 2012 at 3:06 PM, Tony Graziano <
>> tgrazi...@myitdepartment.net> wrote:
>>
>>> Is this an update unmanaged gateway or sip trunk?
>>>  On Dec 11, 2012 3:08 PM, "Josh Patten"  wrote:
>>>
>>>>  Holy incoherent sentences Batman!
>>>>
>>>>  "The culprit, I believe, is sipX inserting conflicting connection
>>>> information into the SDP
>>>>  In the INVITE, and is unnecessarily inserting the WAN address into the
>>>> SDP even though the , causing the media relay to attempt to send the RTP
>>>> out of the WAN interface to the gateway."
>>>>
>>>>  Should read:
>>>>
>>>>  The culprit, I believe, is sipX inserting conflicting connection
>>>> information into the SDP in the INVITE and is unnecessarily inserting the
>>>> public IP address of sipX into the SDP even though the gateway is in the
>>>> same subnet as the sipX server, causing the media relay to attempt to send
>>>> the RTP through the media relay and out over the internet to the gateway.
>>>>
>>>>
>>>> On Tue, Dec 11, 2012 at 1:56 PM, Josh Patten  wrote:
>>>>
>>>>> Chris Rawlings of Blue Cloud Consulting and myself were on the phone
>>>>> with Sangoma attempting to determine why hairpinned calls were resulting 
>>>>> in
>>>>> no Audio, and believe we may have foun

Re: [sipx-users] Message Waiting Issue

[Tony Graziano]

If your DNS is properly configured, the registration should be at the SIP
DOMAIN (not the FQDN).

You can change this to ONE phone and send it one profile to verify.

Manual tinkering s a bad thing. Shame on you.

:>)

On Wed, Dec 12, 2012 at 11:05 AM, Tommy Laino  wrote:

>
>
> I think I found my problem. My registration server in my
> phone group is set to the FQDN. Does everyone concur with me
> before I resend 200 phone profiles?
> --
> Tommy Laino
> Dome Technologies
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>



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Re: [sipx-users] Incorrect Twinkle Softphone Docs on Wiki

[Tony Graziano]

Its not writable by the public. You can request an account to have edit
rights though.

I always have problems with the differing reports for softphones. I think
it needs to be stated whether the user is local or remote. In general,
using the IP or FQDN is discouraged but not all softphones are dutiful and
can resolve SRV records.

It might also depend on what "version" and/or OS your softphone is on,
because the same software package for different operating systems have
(sometimes) differing degrees of functionality. So if you report or edit
the wiki, please provide "OS" and whether you are local or remote. This
does, of course, assume a properly configured system and DNS records.

Thanks,

Tony

On Wed, Dec 12, 2012 at 10:55 AM, Adrien Guillon wrote:

> Hi,
>
> I don't think the sipfoundry wiki is writable by the public, but I do have
> a problem to report.
>
> On page:
> http://wiki.sipfoundry.org/display/sipXecs/Twinkle+1.4.2+Softphone
>
> Under user it states:
>
> Input a name, your User name (sipXecs extension or line or number), Domain
> (server IP or FQDN); then add your Authentication name (sipXecs user) and
> Password (sipXecs SIP Password).
>
> Unfortunately, this isn't correct.  The domain must be the FQDN, an IP
> address isn't good enough.  I did that, and got a bunch of 404 errors,
> until I tried an FQDN, then everything just worked.
>
> Thanks,
>
> AJ
>
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>



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Re: [sipx-dev] SDP NAT rewrite issue

[Tony Graziano]

Then is the server behind nat? Is sipx supporting remoe workers in this
instance?

On Wed, Dec 12, 2012 at 5:47 AM, Chris Rawlings wrote:

> this is the initial invite. This is where the call is comming from the
> ITSP -> WAN IP 24.229.51.68 -> NAT FIREWALL -> SBC IP 192.168.10.19
> (FreeSWITCH / SANGOMA)
>
> The Sangoma SBC then hands this off to the PBX later in the chain where it
> creates both an ITSP and PBX side of the call and bridges them.
>
> This invite needs to be NAT aware as the SBC is behind a NAT firewall..
> all communications on Port 5060 inbound to 192.168.10.19 are ITSP
> communications. All communications on Port 5080 inbound to 192.168.10.19
> are PBX communications.
>
> And then finally i was under the impression that at NO TIME will an
> unmanaged Gateway have a WAN IP address put into any portion of the SIP
> messaging
>
>
> On Tue, Dec 11, 2012 at 9:09 PM, Joegen Baclor  wrote:
>
>>  The root of all evil is in the INVITE.   It advertised itself as NATed
>> by providing a public contact and a public via.  Why is the gateway doing
>> this?
>>
>>
>> INVITE sip:7175463...@blueuc.com SIP/2.0
>> Via: SIP/2.0/UDP 24.229.51.68:5080;rport;branch=z9hG4bKSSZZ7Ht02mHSN
>> Max-Forwards: 11
>> From: "+16107413324" 
>> 
>> ;tag=rFp8vK6FN17Bj
>> To:  
>> Call-ID: 62b9c0f2-be62-1230-4aa1-005056a433a6
>> CSeq: 37288972 INVITE
>> Contact:
>> 
>> User-Agent: NetBorder Session Controller
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>> Supported: precondition, path, replaces
>> Allow-Events: talk, hold, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 191
>> X-Inbound: TRUE
>> X-Account-Code: None
>> X-Account-Inbound: 3587
>> P-Acme-VSA: 202:3587
>> P-Acme-VSA-1: 201:None
>> X-Device: 24.229.51.68
>> X-FS-Support: update_display
>> Remote-Party-ID: "+16107413324" 
>> 
>> ;party=calling;screen=yes;privacy=off
>>
>>
>> v=0
>> o=nsc 1355222135 1355222136 IN IP4 192.168.10.19
>> s=nsc
>> c=IN IP4 192.168.10.19
>> t=0 0
>> m=audio 27938 RTP/AVP 0 8 101 13
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>>
>>
>>
>>
>>
>> On 12/12/2012 05:10 AM, Josh Patten wrote:
>>
>> Sorry I forgot to put this info.
>> Unmanaged Gateway.
>>
>>
>> On Tue, Dec 11, 2012 at 3:06 PM, Tony Graziano <
>> tgrazi...@myitdepartment.net> wrote:
>>
>>> Is this an update unmanaged gateway or sip trunk?
>>>  On Dec 11, 2012 3:08 PM, "Josh Patten"  wrote:
>>>
>>>>  Holy incoherent sentences Batman!
>>>>
>>>>  "The culprit, I believe, is sipX inserting conflicting connection
>>>> information into the SDP
>>>>  In the INVITE, and is unnecessarily inserting the WAN address into the
>>>> SDP even though the , causing the media relay to attempt to send the RTP
>>>> out of the WAN interface to the gateway."
>>>>
>>>>  Should read:
>>>>
>>>>  The culprit, I believe, is sipX inserting conflicting connection
>>>> information into the SDP in the INVITE and is unnecessarily inserting the
>>>> public IP address of sipX into the SDP even though the gateway is in the
>>>> same subnet as the sipX server, causing the media relay to attempt to send
>>>> the RTP through the media relay and out over the internet to the gateway.
>>>>
>>>>
>>>> On Tue, Dec 11, 2012 at 1:56 PM, Josh Patten  wrote:
>>>>
>>>>> Chris Rawlings of Blue Cloud Consulting and myself were on the phone
>>>>> with Sangoma attempting to determine why hairpinned calls were resulting 
>>>>> in
>>>>> no Audio, and believe we may have found an issue in the way sipX modifies
>>>>> the SDP (Sangoma uses FreeSWITCH).
>>>>>
>>>>>  So here is the calling scenario:
>>>>>
>>>>>1. Call comes in from +16107413324 (PSTN CALLER) to 7175463006(Auto 
>>>>> Attendant)
>>>>>2. call is transferred to extension 212, which is set to perform
>>>>>call forwarding (at the same time) to 7174680293 (Cell Phone)
>>>>>3. If the call is answered on 212, no audio
>>>>>4. If the call is answered on cell phone, there is audio
>>>>>
>>>>&g

Re: [sipx-users] Message Waiting Issue

[Tony Graziano]

Ah. SWAP is fine then. You ought to check the sipxconfig/.log for any
errors, but... Only one system has the VM role enabled.  Why don't you set
a phone outside of the group and let sipxconfig manage and set the
parameters, send the phone its profile and see if MWI works.

I tend to think its a DNS/Phone config issue.

Can you explain "why" you set this manually on the phones? Have you ensured
the phones can see the DNS records and that these are correct?

On Wed, Dec 12, 2012 at 7:28 AM, Tommy Laino  wrote:

>
>
> Tony top is showing 12466741k RAM available and swap is at
> 6289436k. Which log would give the best information and I
> will send it to you off list. Also, forgot to mention that
> this is an HA system using load balancing. When I built the
> system I created the phone groups and I set the registration
> server and outbound proxy manually and didnt use the auto
> provision (another brain dead moment on my part).
> --
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> Dome Technologies
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Re: [sipx-users] Message Waiting Issue

[Tony Graziano]

No. As long as the phone is able to lookup the domain and resolve the host
records via SRV it should not matter. The phones (unless manually altered)
will subscribe using the user @ domain and not user @ hostname.domain).

>From the CLI (ssh or console) type in "top" and look at the memory usage
and report back if any swap is in use. If not, consider restarting the
sipXivr process to see if it clears it up. If you do have swap in use
(any), consider a maintenance reboot as it is indicative there may not be
enough RAM installed in the machine to keep this kind of thing from
happening.

In my experience, there are 5 things that can potentially break MWI on a
working system:

1. RAM
2. DNS
3. Moving phones with MWI on from another installation to a new system
(these MWI troubles would be related to these devices and not an overall
system MWI)
4. Phones not being sent the proper information in their config (DNS
related), where someone had the subscribe uri change to the hostname
instead of sipdomain which sipxconfig puts in manually
5. Expire certificates on sipx.

There may be others, but RAM and DNS are the most likely culprits. If
someone changed the DNS the phones get via DHCP then yes, that could break
it easily.

You could provide more information, perhaps a log.


On Tue, Dec 11, 2012 at 11:10 PM, Tommy Laino  wrote:

>
>
> Tony I am not sure what top is. I do not believe that we are
> using swap at all. Todd I would think that maybe you culd be
> right but this is all the phones and it was ll of a sudden.
> I did notice that somehow my FQDN ended up in the Domain
> Alias section not sure why it was in there or what I was
> doing that I thought it was a good idea to put it in there.
> But I removed it and resent all the profles. Do you think
> that could have been part of the problem?
> --
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[sipx-users] Patch 23 and call park

[Tony Graziano]

I have patch #23 installed, but have been having issues retrieving parked
calls.

I thought I heard the subscription forwarding was creating problems with
call park and some other services and that fix was being reverted.

Is it already reverted in patch #23 or is there another patch forthcoming
to revert this?

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Re: [sipx-users] Message Waiting Issue

[Tony Graziano]

I suggested top not the Mrtg statistics. Are toy using any swap?
On Dec 11, 2012 5:40 PM, "Tommy Laino"  wrote:

>
>
> I do not see either of those logs in my snapshot. I checked
> the statistics for the server and I am not having any issues
> with resources
> --
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Re: [sipx-users] Message Waiting Issue

[Tony Graziano]

Also co spider looking g at top and seeing if you have. Resource issues in
general.
On Dec 11, 2012 5:17 PM, "George Niculae"  wrote:

> On Wed, Dec 12, 2012 at 12:15 AM, Tommy Laino  wrote:
>
>>
>>
>> I got SipXecs 4.4 with all Polycom 335/550 phones with 3.2.6
>> firmware. Everything was working just fine until today.
>> Suddenly people would take their voicemails and the light
>> would not extinguish. On the other hand others are receiving
>> messages and not getting any MWI. What gives? I checked the
>> SUBSCRIBE under the messaging for the line and they all show
>> the users extension. What log am I looking for to see if the
>> user subscription is registering?
>>
>
> Check /var/log/sipdb/subscription.xml file for subscriptions,
> /var/log/sipxecs/sipstatus.log for MWI logs (I bet you get some
> unauthorized errors in there). Make sure phone registered with domain and
> not domain alias
>
> George
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Re: [sipx-dev] SDP NAT rewrite issue

[Tony Graziano]

Is this an update unmanaged gateway or sip trunk?
On Dec 11, 2012 3:08 PM, "Josh Patten"  wrote:

> Holy incoherent sentences Batman!
>
> "The culprit, I believe, is sipX inserting conflicting connection
> information into the SDP
> In the INVITE, and is unnecessarily inserting the WAN address into the SDP
> even though the , causing the media relay to attempt to send the RTP out of
> the WAN interface to the gateway."
>
> Should read:
>
> The culprit, I believe, is sipX inserting conflicting connection
> information into the SDP in the INVITE and is unnecessarily inserting the
> public IP address of sipX into the SDP even though the gateway is in the
> same subnet as the sipX server, causing the media relay to attempt to send
> the RTP through the media relay and out over the internet to the gateway.
>
>
> On Tue, Dec 11, 2012 at 1:56 PM, Josh Patten  wrote:
>
>> Chris Rawlings of Blue Cloud Consulting and myself were on the phone with
>> Sangoma attempting to determine why hairpinned calls were resulting in no
>> Audio, and believe we may have found an issue in the way sipX modifies the
>> SDP (Sangoma uses FreeSWITCH).
>>
>> So here is the calling scenario:
>>
>>1. Call comes in from +16107413324 (PSTN CALLER) to 7175463006 (Auto
>>Attendant)
>>2. call is transferred to extension 212, which is set to perform call
>>forwarding (at the same time) to 7174680293 (Cell Phone)
>>3. If the call is answered on 212, no audio
>>4. If the call is answered on cell phone, there is audio
>>
>>
>> The culprit, I believe, is sipX inserting conflicting connection
>> information into the SDP
>> In the INVITE, and is unnecessarily inserting the WAN address into the
>> SDP even though the , causing the media relay to attempt to send the RTP
>> out of the WAN interface to the gateway.
>>
>> Here is the SDP for step 1 (generated by the gateway):
>>
>> v=0
>> o=nsc 1355222135 1355222136 IN IP4 192.168.10.19
>> s=nsc
>> c=IN IP4 192.168.10.19
>> t=0 0
>> m=audio 27938 RTP/AVP 0 8 101 13
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> Here is the SDP that is generated internally by FreeSWITCH IVR and sent
>> to the Proxy:
>> v=0
>> o=FreeSWITCH 1355235729 1355235730 IN IP4 192.168.10.9
>> s=FreeSWITCH
>> c=IN IP4 192.168.10.9
>> t=0 0
>> m=audio 12894 RTP/AVP 9 101 13
>> a=rtpmap:9 G722/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>>
>> And here is what we send back in our 200 OK to the gateway after
>> traversing the Proxy:
>> v=0
>> o=FreeSWITCH 1355235655 1355235656 IN IP4 192.168.10.9
>> s=FreeSWITCH
>> *c=IN IP4 192.168.10.9*
>> t=0 0
>> m=audio 30496 RTP/AVP 0 101 13
>> *c=IN IP4 24.229.51.65 < THIS IS THE PUBLIC IP OF SIPX*
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>> a=x-sipx-ntap:X192.168.10.9-24.229.51.65;14
>>
>> Why is the proxy messing with the SDP like this when there is clearly no
>> need to rewrite it for a call that the RTP is terminating on devices on the
>> same subnet?
>>
>> I've attached a trace of the SDP example.
>>
>> --
>> Josh Patten
>> eZuce
>> Solutions Architect
>> O.978-296-1005 X2050
>> M.979-574-5699
>> http://www.ezuce.com
>>
>>
>
>
> --
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> eZuce
> Solutions Architect
> O.978-296-1005 X2050
> M.979-574-5699
> http://www.ezuce.com
>
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Re: [sipx-users] Multiple key presses

[Tony Graziano]

in a pcap, dtmf will appear to press multiple times and reading that is
actually quite confusing. are you using the same ITSP or gateway that she
is using? If not, you are not actually recreating her call.

I have seen this in the past with itsp's who use a different payload number
for dtmf. If it is a pstn gateway, verify the payload.

a pcap or siptrace speaks for itself.

On Tue, Dec 11, 2012 at 12:05 PM, Geoff Musgrave <
geoff.musgr...@cacionline.net> wrote:

>  I have 1 user with a softphone that calls into several automated systems
> throughout the day and for some reason she cannot navigate through them.
> From the captures I’ve done it appears that when she presses the numbers
> either on the keyboard or the dial pad built into the softphone the signal
> is sent multiple times. This is only happening with 1 user and I cannot
> recreate the issue from my computer using Bria, X-lite, Zoiper, or
> VOP-nano; all of which we have had her try and she continues to get the
> same results. Can someone give me something else to look for?
>
> ** **
>
> Thanks in advance.
>
> ** **
>
> --
>
> Geoff
>
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Re: [sipx-users] Latest Stage (4.4) outbound dialing issues with dialplan

[Tony Graziano]

Actually, I think it might be that this is a new ITSP and I fat fingered
something in the config.

The ITSP (appia) platform (cardinal using opensips) requires sending calls
to them on the SRV record OR I have to define the gateway(s) by A records
(hostname).

It seems I had the proxy entered in as the sip domain name for the ITSP.
Once I removed this it actually resolved itself.

Sorry for the false alarm!

On Tue, Dec 11, 2012 at 12:01 PM, Joegen Baclor  wrote:

>  Highly important that you confirm that this works for you pre dec-5
> update.
>
>
> On 12/12/2012 12:01 AM, Tony Graziano wrote:
>
> Is it just me?
>
>  I have a dialplan rule when I use "xx" plus 10 digits to strip the "xx"
> and send the 10 digits to a specified gateway. When I do this with the
> latest stage (dated dec. 5) I get a 500 internal server error from sipx.
>
>  Exception Info Unexpected error creating INVITE  at SipUtilities.java:936
>
>  The logging in debug really doesn't show anything I saw as meaningful.
> Can anyone replicate this?
>
>  --
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>


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Re: [sipx-users] Latest Stage (4.4) outbound dialing issues with dialplan

[Tony Graziano]

in my case permissions match in both dial rules.

On Tue, Dec 11, 2012 at 11:21 AM, Elwin Formsma wrote:

>  Hi Tony,
>
>  Mabye slightly related to this:
>
>  Rule 1: Dialplan rule with XX plus 10 => does not require permissions
> Rule 2: There is also a dialplan with 10 digits => does require
> permissions YY
>
>  User doesnt has permissions YY
> When you dial XX+10 digits (follows Rule 1) you can dial it without
> problem because you dont need permissions
> When you dial 10 digits without XX (should follow Rule 2) you can still
> dial it! This should not be possible because the rule 2 requires
> permissions.
>
>
>  Met vriendelijke groet,
>
>
> Elwin Formsma
> Telecats BV
> -
> Elwin Formsma | Telecats bv | KvK Enschede 06069106 | Tel:   053 488 99 44
> | Fax: 053 488 99 10 | E-mail: e.form...@telecats.nl |
>
>  Op 11 dec. 2012, om 17:01 heeft Tony Graziano <
> tgrazi...@myitdepartment.net> het volgende geschreven:
>
> Is it just me?
>
>  I have a dialplan rule when I use "xx" plus 10 digits to strip the "xx"
> and send the 10 digits to a specified gateway. When I do this with the
> latest stage (dated dec. 5) I get a 500 internal server error from sipx.
>
>  Exception Info Unexpected error creating INVITE  at SipUtilities.java:936
>
>  The logging in debug really doesn't show anything I saw as meaningful.
> Can anyone replicate this?
>
>  --
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[sipx-users] Latest Stage (4.4) outbound dialing issues with dialplan

[Tony Graziano]

Is it just me?

I have a dialplan rule when I use "xx" plus 10 digits to strip the "xx" and
send the 10 digits to a specified gateway. When I do this with the latest
stage (dated dec. 5) I get a 500 internal server error from sipx.

Exception Info Unexpected error creating INVITE  at SipUtilities.java:936

The logging in debug really doesn't show anything I saw as meaningful. Can
anyone replicate this?

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Re: [sipx-users] Call Park Presence

[Tony Graziano]

I think it is well known that you should not park or upoark calls using the
BLF button on a Polycom.
On Dec 7, 2012 1:07 PM, "Bryan Anderson"  wrote:

> So I noticed some talk in a previous email "Call forward fails to
> external number" about the Adtran 900 series.  I have a couple of comments
> and questions.
>
> We have a TA908e 2nd gen running AOS A5.02.00.E.  We currently have not
> noticed any issue with having an external caller forwarded to and 
> externalnumber.  cell => user for 30sec
> => external number.
>
> What we have had issues with is presence monitoring of call parking.   We
> have a Polycom Soundpoint IP 650 with a sing side car that monitors parklines 
> 6000
> - 6003.  We can park calls no problem, and so far have not had trouble
> retrieving calls.  Our problem is that once the call gets retrieved from
> the call park the BLF never stops blinking.  I have to restart the
> Park/Presence servers.
>
> This is with SipXecs 4.4.0.
>
> Thoughts and comments would be appreciated.
>
>
> -Bryan Anderson
>
>
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Re: [sipx-users] 4.6 Superuser Login

[Tony Graziano]

Does the wiki article need an explanation of how to reset superadmin if
login fails after a restoring a 4.4 version to 4.6?

Also, in 4.6, does it make sense for the restore to detect a version of 4.4
and provide the superadmin reset as part of the sipxconfig GUI?
On Dec 7, 2012 7:17 AM, "Laurentiu Ceausescu"  wrote:

> On Fri, Dec 7, 2012 at 2:10 PM, Douglas Hubler  wrote:
>
>> On Fri, Dec 7, 2012 at 6:50 AM, Tony Graziano
>>  wrote:
>> > Sounds like the wiki needs a how to page on 4.4 to 4.6 migration...
>> >
>> > Backup.
>> > Install 4.6
>> > Restore
>> > Perform superadmin password reset
>> > Login and use
>>
>> We have
>>
>>
>> http://wiki.sipfoundry.org/display/sipXecs/Upgrade+to+Latest+Stable+Version
>>
>> should be updated, however most folks will not RTFM so in addition, I
>> like Michael's suggestion of highlighting field may be best.  We
>> cannot make it mandatory because restoring from 4.6 it's not required.
>>  Config team got this?
>
>
> sure, we'll do
>
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Re: [sipx-users] Polycom outbound proxy

[Tony Graziano]

I was not indicating a possible path overcome a down gateway AND an
unavailable proxy.

However if both are down...
On Dec 7, 2012 7:08 AM, "Michael Picher"  wrote:

> well, isn't the point of SAS that you can't reach the proxy in the first
> place, so what's the sense in this discussion anyway...
>
>
> On Fri, Dec 7, 2012 at 6:36 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> I think it would help a lot if sipxbridge could send a failure message to
>> the proxy so the proxy could try another gateway (i.e. put these in a
>> dailplan). Though the gateways (if this is a pstn gateway) or sipxbridge
>> needs to be able to discern this is a transport issue and generate maybe a
>> 302 versus a 486 busy.
>>
>> I think if the proxy has the ability to detect "conclusively" that
>> something is down it can route calls. Conversely, the UA's can do the same
>> thing by registering one line to each (one line to sipx, the other to the
>> gateway).
>>
>>
>>
>> On Fri, Dec 7, 2012 at 6:03 AM, Michael Picher  wrote:
>>
>>> >From my recollection, SAS does not support using a SRV record.
>>>
>>> This is a bit of a problem as I see it.  Trading one point of failure
>>> for another.  I guess you just need to ask yourself, is it more likely that
>>> an ITSP will go down or your gateway will fail.  I think the former is more
>>> likely.
>>>
>>> Mike
>>>
>>>
>>> On Fri, Dec 7, 2012 at 5:48 AM, Marco Colaneri wrote:
>>>
>>>>
>>>>
>>>> Thank you Josh,
>>>>
>>>> I know that way to configure SAS.
>>>>
>>>> We didn't choose it because all sip signaling traffic goes
>>>> through the gateway.
>>>> So if the gateway fails or becomes unreachable, all the
>>>> phone can't register and reach SipXecs servers.
>>>>
>>>> Maybe we could get over this problem by defining a new SRV
>>>> record on DNS (e.g. gwsite1.sipdomain) which points
>>>> primarily to the audiocodes gateway and secondarily to the
>>>> sipxecs servers. This configuration should fix problems due
>>>> to Audiocodes failures. Am I right or I'm missing something?
>>>> If we didn't deploy a DNS server locally, would this
>>>> solution still work when the site WAN link fails? How could
>>>> phones resolve outbound proxy address? Perhaps using DNS
>>>> records caching?
>>>>
>>>> Thank you very much for help.
>>>>
>>>> Marco
>>>>
>>>>
>>>> ___
>>>> sipx-users mailing list
>>>> sipx-users@list.sipfoundry.org
>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>>>
>>>
>>>
>>>
>>> --
>>> Michael Picher, Director of Technical Services
>>> eZuce, Inc.
>>>
>>> 300 Brickstone Square
>>>
>>> Suite 201
>>>
>>> Andover, MA. 01810
>>> O.978-296-1005 X2015
>>> M.207-956-0262
>>> @mpicher <http://twitter.com/mpicher>
>>> linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
>>> www.ezuce.com
>>>
>>>
>>> 
>>> "The best way to predict the future is to invent it." - Alan Kay
>>>
>>>
>>> ___
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>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>>
>>
>>
>>
>> --
>> ~~
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>> Telephone: 434.984.8430
>> sip: tgrazi...@voice.myitdepartment.net
>> Fax: 434.465.6833
>> ~~
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>> Ask about our Internet Fax services!
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>>
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Re: [sipx-users] 4.6 Superuser Login

[Tony Graziano]

Sounds like the wiki needs a how to page on 4.4 to 4.6 migration...

Backup.
Install 4.6
Restore
Perform superadmin password reset
Login and use
On Dec 7, 2012 6:03 AM, "Michael Picher"  wrote:

> I am seeing that we are going to have to have giant flashing neon letters
> on the restore page...  ugh...
>
>
> On Fri, Dec 7, 2012 at 2:19 AM, Jan Fricke  wrote:
>
>> We had this problem too :-)
>> Did you set a default pin when restoring the 4.4 backup? If not the pin is
>> empty. I did a portforward on port 5432 and used pgadmin to set the new
>> pin in the user table of sipxconfig.
>>
>> _
>> Jan Fricke (B.Sc.)
>>
>> IANT -
>> APPLIED NGN-TECHNOLOGIES
>>
>> Turn-Key VoIP/UC Solutions and More...
>>
>>
>> Fon: +49 (5331) 6794 0
>> Fax: +49 (5331) 6794 499
>> Mail: jan.fri...@iant.de
>> Web: www.iant.de
>>
>>
>> IANT is eZuce Elite Partner for EMEA
>>
>> IANT is Member of GROUPLINK
>>
>> -Ursprüngliche Nachricht-
>> Von: sipx-users-boun...@list.sipfoundry.org
>> [mailto:sipx-users-boun...@list.sipfoundry.org] Im Auftrag von Charles
>> Chalekson MD
>> Gesendet: Freitag, 7. Dezember 2012 04:45
>> An: sipx-users@list.sipfoundry.org
>> Betreff: [sipx-users] 4.6 Superuser Login
>>
>> Updated today from 4.4 to 4.6 with a clean ISO CD installation.
>> Followed with yum update.
>> Able to sign into to GUI after establishing new superadmin password
>> Restore without issues from 4.4 config Unable to login under to superadmin
>> account.
>>
>> Advise?
>> Charles
>>
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>> ___
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>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>
>
>
> --
> Michael Picher, Director of Technical Services
> eZuce, Inc.
>
> 300 Brickstone Square
>
> Suite 201
>
> Andover, MA. 01810
> O.978-296-1005 X2015
> M.207-956-0262
> @mpicher 
> linkedin 
> www.ezuce.com
>
>
> 
> "The best way to predict the future is to invent it." - Alan Kay
>
>
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>

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Re: [sipx-users] Polycom outbound proxy

[Tony Graziano]

I think it would help a lot if sipxbridge could send a failure message to
the proxy so the proxy could try another gateway (i.e. put these in a
dailplan). Though the gateways (if this is a pstn gateway) or sipxbridge
needs to be able to discern this is a transport issue and generate maybe a
302 versus a 486 busy.

I think if the proxy has the ability to detect "conclusively" that
something is down it can route calls. Conversely, the UA's can do the same
thing by registering one line to each (one line to sipx, the other to the
gateway).



On Fri, Dec 7, 2012 at 6:03 AM, Michael Picher  wrote:

> >From my recollection, SAS does not support using a SRV record.
>
> This is a bit of a problem as I see it.  Trading one point of failure for
> another.  I guess you just need to ask yourself, is it more likely that an
> ITSP will go down or your gateway will fail.  I think the former is more
> likely.
>
> Mike
>
>
> On Fri, Dec 7, 2012 at 5:48 AM, Marco Colaneri  wrote:
>
>>
>>
>> Thank you Josh,
>>
>> I know that way to configure SAS.
>>
>> We didn't choose it because all sip signaling traffic goes
>> through the gateway.
>> So if the gateway fails or becomes unreachable, all the
>> phone can't register and reach SipXecs servers.
>>
>> Maybe we could get over this problem by defining a new SRV
>> record on DNS (e.g. gwsite1.sipdomain) which points
>> primarily to the audiocodes gateway and secondarily to the
>> sipxecs servers. This configuration should fix problems due
>> to Audiocodes failures. Am I right or I'm missing something?
>> If we didn't deploy a DNS server locally, would this
>> solution still work when the site WAN link fails? How could
>> phones resolve outbound proxy address? Perhaps using DNS
>> records caching?
>>
>> Thank you very much for help.
>>
>> Marco
>>
>>
>> ___
>> sipx-users mailing list
>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>
>
>
> --
> Michael Picher, Director of Technical Services
> eZuce, Inc.
>
> 300 Brickstone Square
>
> Suite 201
>
> Andover, MA. 01810
> O.978-296-1005 X2015
> M.207-956-0262
> @mpicher <http://twitter.com/mpicher>
> linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
> www.ezuce.com
>
>
> ----
> "The best way to predict the future is to invent it." - Alan Kay
>
>
> _______
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not available

[Tony Graziano]

 mind a little over the past few days, was this
>> just a typo from the 4.4 configuration for ITSP's using 5080 inbound, or is
>> the ITSP still sending to you on 5080?  We tested out inbound on 5060
>> (albeit weeks ago and many updates) and it was working fine.  I am guessing
>> its a typo, or you configured your system properly to still use 5080, but
>> it's sticking in my craw that maybe this isn't a typo or the ITSP is
>> sending to the wrong port?
>>
>>I have not tested the 5080 inbound on openUC 4.6 instead of 5060.
>>  Checking Devices -> SIP Trunk SBCs shows that sipXbridge-1 has port 5080
>> listed.  Features -> NAT Traversal under Server Config shows the private
>> IP,  public IP, and 5060 for SIP port and 5061 for TLS SIP port.  It could
>> very well work that 5080 inbound from ITSP's still functions as it did in
>> 4.4 and I think I remember someone saying you could use either or both?
>>  However I haven't bothered to test it yet since I was ecstatic to get
>> native 5060 for some picky providers and since that appeared to be working
>> over a month ago in our tests I just wanted to ask if the calls were really
>> sent to 5080 or 5060?
>>
>>  Trevor
>>
>>  On Nov 28, 2012, at 5:05 PM, Nicholas Drayer 
>> wrote:
>>
>> It’s only when a call comes in via the sipXecs server’s 5080 port that
>> the Bria client exhibits no audio sent/received.
>>
>>
>>
>> ___
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>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>
>
>
>  --
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> eZuce, Inc.
> 300 Brickstone Square
> Suite 201
> Andover, MA. 01810
>  O.978-296-1005 X2015
> M.207-956-0262
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> www.ezuce.com
>
>
> ----
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>
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>
>
>
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>



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Re: [sipx-users] Fax Service Question

[Tony Graziano]

I am not sure it matters. I should probably do a wiki page on this.

I have successfully deployed t.38 with and without siptrunks. With PRI's
and with paper and paperless fax machines behind both trunks and pri's.

Not all sip trunk providers can do t.38. Not all ATA's can do t.38. In my
use cases I put all the ATA's "behind" sipx (trunked or PRI) so that they
register and all CDR is handled and tracked through sipx.

t.38 providers must support reinvite with sdp in order to communicate with
FS on sipx. The Audiocodes MP-202B looks neat, but even with an efax
account the fax machine needs to support https faxing. At least with a
traditional gateway you can use it for phone calls too. Patton Sn4112's are
only 40 dollars more than the 202B. I'm sure AC has a similar price point
on the MP-112's.

One of the more interesting things about that device is that it supports
t.37, so if we decided to build t.37 support into sipxconfig for users in
unified messaging to send faxes (by email), it can potentially be useful.

On Sat, Dec 1, 2012 at 4:36 AM, Michael Picher  wrote:

> Definitely a hijack...  But a good option for a few fax users, just need
> an efax account.
>
>
> On Sat, Dec 1, 2012 at 4:28 AM, m...@mattkeys.net wrote:
>
>> I've been looking into buying the Audiocodes MP114 2fxo/2fxs for testing
>> t.38 when I ran across the MP202B with fax over HTTPS (
>> http://www.voipsupply.com/blog/audiocodes-mp-202b-httpsfax-adapter). I'm
>> curious if you (or anyone else reading) have tested the 202B while we're on
>> the topic. My apologies if that's considered thread jacking.
>>
>> ____
>> From: sipx-users-boun...@list.sipfoundry.org [
>> sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano [
>> tgrazi...@myitdepartment.net]
>> Sent: Saturday, December 01, 2012 3:43 AM
>> To: Discussion list for users of sipXecs software
>> Subject: Re: [sipx-users] Fax Service Question
>>
>> Realize if the ITSP does not offer t.38 on the initial invite FS will not
>> find a suitable codec for a fax transaction and off re-invite with t.38.
>> The provider must support re-invite with sdp if they do not send t.38
>> initially.
>>
>> IF t.38 is negotiated and the fax still fails, you will get a "ZERO" page
>> email with a TIFF attachment (blank). If it is successful you will get the
>> email as a pdf (assuming you are all patched up).
>>
>> I have tested against more than a few "t.38 compatible" providers, and
>> found many struggled with being able to offer guaranteed routes for t.38 in
>> general, so choose wisely or test thoroughly before you commit.
>>
>> ___
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>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>
>
>
> --
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> eZuce, Inc.
>
> 300 Brickstone Square
>
> Suite 201
>
> Andover, MA. 01810
> O.978-296-1005 X2015
> M.207-956-0262
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> linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
> www.ezuce.com
>
>
> ----
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>
>
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>



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Re: [sipx-users] Fax Service Question

[Tony Graziano]

Realize if the ITSP does not offer t.38 on the initial invite FS will not
find a suitable codec for a fax transaction and off re-invite with t.38.
The provider must support re-invite with sdp if they do not send t.38
initially.

IF t.38 is negotiated and the fax still fails, you will get a "ZERO" page
email with a TIFF attachment (blank). If it is successful you will get the
email as a pdf (assuming you are all patched up).

I have tested against more than a few "t.38 compatible" providers, and
found many struggled with being able to offer guaranteed routes for t.38 in
general, so choose wisely or test thoroughly before you commit.

On Fri, Nov 30, 2012 at 9:40 PM, Tommy Laino  wrote:

>
>
> Checked the logs. The sipXivr log shows that the Fax failed
> because the call dropped prematurely. I searched through the
> sipXproxy log and noticed that the incoming fax looked like
> it was coming through on port 5060. Im assuming that fax
> should use port 5080 like all the other incoming calls. I
> will get with my provider to get that resolved. Thanks for
> the help
> --
> Tommy Laino
> Dome Technologies
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>



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Re: [sipx-users] Fax Service Question

[Tony Graziano]

And put sipxivr and sipxproxy in debug log Lev, send a fax And inspect the
logs including sipxconfig.to see if there are errors.
On Nov 30, 2012 7:24 PM, "Tony Graziano" 
wrote:

> Do a search for the DID number and see if it is in multiple places on the
> system.
> On Nov 30, 2012 6:21 PM, "Tommy Laino"  wrote:
>
>>
>>
>> Yes I do hear the fax tone and the sending machine says that
>> the transmission was successful.
>> --
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>> Dome Technologies
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>>
>

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Re: [sipx-users] Fax Service Question

[Tony Graziano]

Do a search for the DID number and see if it is in multiple places on the
system.
On Nov 30, 2012 6:21 PM, "Tommy Laino"  wrote:

>
>
> Yes I do hear the fax tone and the sending machine says that
> the transmission was successful.
> --
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Re: [sipx-users] Fax Service Question

[Tony Graziano]

I.e.

Faxes don't go to user via paperless unless the DID is on the unified
messaging page of the used account, otherwise you would assume it is
sending to a fxs device which does t.38 (paper, not paperless).
A user can't receive paperless faxes with a did number unless it is entered
on the unified messaging page.
On Nov 30, 2012 5:39 PM, "Tony Graziano" 
wrote:

> The user is 12345 and their unifies messaging has a fax mailbox of 54321
> and a fax DID ON THE UNIFIED messaging page of 2223334567. Right?
> On Nov 30, 2012 5:33 PM, "Tommy Laino"  wrote:
>
>>
>>
>> The user is extension 5011. The fax extension is 6011. The
>> fax DID is set to the T.38 fax number. What do mean a setup
>> a fax box? I think I have everything setup the way you are
>> describing
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>

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Re: [sipx-users] Fax Service Question

[Tony Graziano]

The user is 12345 and their unifies messaging has a fax mailbox of 54321
and a fax DID ON THE UNIFIED messaging page of 2223334567. Right?
On Nov 30, 2012 5:33 PM, "Tommy Laino"  wrote:

>
>
> The user is extension 5011. The fax extension is 6011. The
> fax DID is set to the T.38 fax number. What do mean a setup
> a fax box? I think I have everything setup the way you are
> describing
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Re: [sipx-users] Fax Service Question

[Tony Graziano]

Depending on what patch version you are on this might actually go out as a
PDF file. The user account needs to have the D ID number from the provider
put on their fax account in Unified Messaging. This means you need to
create both a fax box and 50 ID number on the Unified Messaging page for
that user.
On Nov 30, 2012 4:53 PM, "Tommy Laino"  wrote:

>
>
> I have SipXecs 4.4 running. The unified messaging is working
> fine. All voicemails are delivered to the users email
> immediately without any issue. DNS is running and tested
> good. I have setup the Fax Extension and the Fax DID. The
> Fax DID is a T.38 DID from my SIP trunk provider.
>
> When I check the CDR I see the call completed but the logs
> are not showing the email go out. When I look at the maillog
> I can see the voicemail delivery going out but I do not see
> the fax going out. Its my understanding that the fax should
> be delivered to the users email as a TIFF file. Am I missing
> something? Shouldn't this work?
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Re: [sipx-users] CyberData

[Tony Graziano]

similarly you can use the valcom poe ip speakers. they are multicast so you
can dial from sipx like an extension and page MANY devices without any
network load OR you can use existing speaker (analog) and use a valcom sip
paging gateway to accomplish the same thing.

On Fri, Nov 30, 2012 at 2:19 PM, Todd Hodgen  wrote:

> You can also use a low cost ATA (Linksys, Grandstream, Audiocodes) that
> allows you to call an extension number, the ATA pots port connects to a
> valcom or bogen Page Adapter, and any number of standard speakers.   Old
> fashioned way of doing things, but it works very reliably.
>
> All three of those ATA/Gateways are managed by sipxecs, making
> configuration
> very simple.
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark Theis
> Sent: Friday, November 30, 2012 11:10 AM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] CyberData
>
> Yes. I like them. I have installed 6 or so. Only problem I have is that
> they
> need to be rebooted every 6 months or so. Not all of them, just random
> ones.
> Lol.
>
> I have daisy chained 6 of their non-ip speakers to one ip speaker and they
> worked great. Also daisy chained multiple existing warehouse speaker/horns
> and they worked perfectly! There must have been 10 of the horns in the
> giant
> warehouse. And the 1 ip ceiling speaker powered them all.
>
> Sent from my iPhone
>
> On Nov 30, 2012, at 11:03 AM, "Bryan Anderson" 
> wrote:
>
> > Does any one have any experience with the Cyberdata SIP Ceiling paging
> speakers?
> >
> >
> > -Bryan Anderson
> >
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Re: [sipx-users] Help/info on Paging SipXecs 4.4.0

[Tony Graziano]

I think the firmware on the phones limits this to (reliably) a max of 12
devices.
On Nov 30, 2012 11:41 AM, "Bryan Anderson"  wrote:

> Hello,
>
> We have an office of ~83 endpoints  all but one phone are Polycom
> SoundPoint IP 331's, the receptions uses a 650 with 1 sidecar.
>
> The firmware's are currently as shipped with a mix of firmwares 3.2.4 and
> 3.3.0 (please lets avoid the downgrade if we can, both version are having
> this issue).
>
> Every thing worked fine except paging.  Which in this office is a big
> thing.  The problem is some pages worked and some didn't.  The pages that
> didn't what happened is a large group of the phones would start to receive
> the page with the notification tone, then just stop even though the page
> was still in progress.  The phones this occurred on changed with each of
> the pages that failed.  I am at a loss for identifying the problem due to
> the inconsistency of the failures.
>
>
> My Questions are:
>
> 1) Has anyone else experience this and been able to resolve it,
>
> 2) Does any one else do a High volume of pages (30-50/day) and have this
> working.  If so, what equipment and sipx version are you using?
>
> 3) If it really is the 3.3.0 that's causing my problems, can it be
> downgraded without touching all of the 30 or so phones one by one?
>
> We still have SipXecs in place but are still using the old key system tell
> this is working.
>
>
> Thanks,
> Bryan Anderson
>
>
>
>
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Re: [sipx-users] Upgrade from 4.2.1 to 4.6

[Tony Graziano]

In 4.2-4.4 you should use the same hostname/IP/sip domain before doing your
restore.

Did you take those steps?
On Nov 28, 2012 7:31 AM, "Daniel Peinado"  wrote:

> Hello,
>
> I have installed the sipxecs 4.2.1. My setup has 1 master Sipxecs and one
> slave. I try to restore the backup and I do the next.
>
> 1) install sipxecs (1 master) 4.2.1 and yum update
> 2) install sipxecs (2 slave) 4.2.1 and yum update
> 3) add slave server in sipxecs interface and send profiles.
> 4) restart servers and everything runs good.
> 5) restore backup from the interface. It seems to restore right, but when
> I restart the servers ( I must restart because sipxecs requires) I have
> problems with the slave server. The restart option goes wrong. I tried to
> setup again the slave server, but I have the same problem. What I'm doing
> wrong?
>
> Thank you
>
> Daniel Peinado López
>
>
> El 28/11/2012, a las 00:29, Tony Graziano 
> escribió:
>
> > If it matters to anyone, I seem to recall there were structural or DB
> > changes between 4.2 and 4.4. If that is the case (I could be wrong but
> > I'm pretty sure there were changes to the DB) it would not work well.
> > If I am incorrect it would not work well.
> >
> > The update to 4.4 is pretty painless just make sure you send the
> > server its profiles and everything seems operational before backing up
> > to move forward if you take this route.
> >
> > On Tue, Nov 27, 2012 at 8:01 AM, Douglas Hubler 
> wrote:
> >> Restoring from 4.2.1 into a 4.6 system might just work.  You can try
> >> the restore in a virtual machine first just to make sure it works
> >> first.  What you'd do is install 4.6 first, then restore into that
> >> installation.  Tony's method is safest.
> >>
> >> On Tue, Nov 27, 2012 at 7:41 AM, Daniel Peinado Lopez
> >>  wrote:
> >>> Hello,
> >>>
> >>> I need to upgrade my SipXecs to 4.6. I have all data for the
> configuration
> >>> in one backup file. I think about doing the next:
> >>>
> >>> 1) Install Sipxecs 4.4 and restore the backup from 4.2.1.
> >>> 2) Make a new backup in Sipxecs 4.4
> >>> 3) Install Sipxecs 4.6 and restore the backup from 4.4.
> >>>
> >>> It´s possible to restore a 4.2.1 backup in sipxecs 4.6 (only one step,
> from
> >>> 4.2.1 to 4.6) and everything runs?
> >>>
> >>> Can you tell me the best way?
> >>>
> >>> Thank you very much
> >>>
> >>> --
> >>>
> >>>
> >>> Daniel Peinado López
> >>>
> >>> IANT - APPLIED NGN-TECHNOLOGIES
> >>>
> >>> Turn-Key VoIP/UC Solutions and More...
> >>>
> >>>
> >>> Fon: +49 (5331) 6794 450
> >>> Fax: +49 (5331) 6794 499
> >>> Mail: daniel.pein...@iant.de
> >>> Web: www.iant.de
> >>>
> >>>
> >>> IANT is eZuce Elite Partner for EMEA
> >>>
> >>> IANT is Member of GROUPLINK
> >>>
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> >
> >
> >
> > --
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Re: [sipx-users] Upgrade from 4.2.1 to 4.6

[Tony Graziano]

If it matters to anyone, I seem to recall there were structural or DB
changes between 4.2 and 4.4. If that is the case (I could be wrong but
I'm pretty sure there were changes to the DB) it would not work well.
If I am incorrect it would not work well.

The update to 4.4 is pretty painless just make sure you send the
server its profiles and everything seems operational before backing up
to move forward if you take this route.

On Tue, Nov 27, 2012 at 8:01 AM, Douglas Hubler  wrote:
> Restoring from 4.2.1 into a 4.6 system might just work.  You can try
> the restore in a virtual machine first just to make sure it works
> first.  What you'd do is install 4.6 first, then restore into that
> installation.  Tony's method is safest.
>
> On Tue, Nov 27, 2012 at 7:41 AM, Daniel Peinado Lopez
>  wrote:
>> Hello,
>>
>> I need to upgrade my SipXecs to 4.6. I have all data for the configuration
>> in one backup file. I think about doing the next:
>>
>> 1) Install Sipxecs 4.4 and restore the backup from 4.2.1.
>> 2) Make a new backup in Sipxecs 4.4
>> 3) Install Sipxecs 4.6 and restore the backup from 4.4.
>>
>> It´s possible to restore a 4.2.1 backup in sipxecs 4.6 (only one step, from
>> 4.2.1 to 4.6) and everything runs?
>>
>> Can you tell me the best way?
>>
>> Thank you very much
>>
>> --
>>
>>
>> Daniel Peinado López
>>
>> IANT - APPLIED NGN-TECHNOLOGIES
>>
>> Turn-Key VoIP/UC Solutions and More...
>>
>>
>> Fon: +49 (5331) 6794 450
>> Fax: +49 (5331) 6794 499
>> Mail: daniel.pein...@iant.de
>> Web: www.iant.de
>>
>>
>> IANT is eZuce Elite Partner for EMEA
>>
>> IANT is Member of GROUPLINK
>>
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Re: [sipx-users] Upgrade from 4.2.1 to 4.6

[Tony Graziano]

If it were me. I'd make a backup of 4.21 and do an in-place update to
4.4 (latest), then I would back it up and restore it to a 4.6 system.



On Tue, Nov 27, 2012 at 7:41 AM, Daniel Peinado Lopez
 wrote:
> Hello,
>
> I need to upgrade my SipXecs to 4.6. I have all data for the configuration
> in one backup file. I think about doing the next:
>
> 1) Install Sipxecs 4.4 and restore the backup from 4.2.1.
> 2) Make a new backup in Sipxecs 4.4
> 3) Install Sipxecs 4.6 and restore the backup from 4.4.
>
> It´s possible to restore a 4.2.1 backup in sipxecs 4.6 (only one step, from
> 4.2.1 to 4.6) and everything runs?
>
> Can you tell me the best way?
>
> Thank you very much
>
> --
>
>
> Daniel Peinado López
>
> IANT - APPLIED NGN-TECHNOLOGIES
>
> Turn-Key VoIP/UC Solutions and More...
>
>
> Fon: +49 (5331) 6794 450
> Fax: +49 (5331) 6794 499
> Mail: daniel.pein...@iant.de
> Web: www.iant.de
>
>
> IANT is eZuce Elite Partner for EMEA
>
> IANT is Member of GROUPLINK
>
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Re: [sipx-dev] Loop detected with ITSP

[Tony Graziano]

sipx says the call is in progress but no ack from the ITSP. You sent
the server its profiles after the update so the services could
restart?

if so, I'd say it is a bug. voip.ms has a few pops that are not as
easy to deal with. perhaps you could try a call through the NY POP
just to be sure? Otherwise I'd say its a but, but its also a dev
version so its kinda expected this kind of thing could happen.

On Mon, Nov 26, 2012 at 4:00 PM, Todd Hodgen  wrote:
>
>
> I’ve attached a PCAP of a call from a system running sipxecs 4.7.I’ve
> been using this particular ITSP for testing for about 3 ½ years (voip.ms),
> without seeing this particular issue.  I updated a current version of 4.6 to
> 4.7 via yum, and since that time have noticed that calls are failing at 20
> seconds.  It appears keep-alive is being looped back maybe?
>
>
>
> Call scenario – Polycom 550 phone running 3.2.7 places a call to
> 206-390-4689 (cell), call is answered with two way audio and drops at 20
> seconds.
>
>
>
> Wondering if something added recently for sipxbridge might be causing this
> issue?
>
>
>
> Todd R. Hodgen
>
>
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Re: [sipx-dev] Query on hunt group with fall back destination and allow call forwarding

[Tony Graziano]

I think its important to realize the hunt group is not a service and the
logic is very basic.

There are limitations, a lot of these limitations have to do with the way
it is implemented in the proxy. Also, please consult the wiki when
constructing a hunt group.

Forwarding has limitations.

Your description needs to be elaborated. User 207 has to be registered.
Actually, all users who are supposed to ring should be registered. You also
need to make sure the forwarding at the user is "at the same time" and not
"if not answer" because then you create conflicting statements and the
fallback destination is the only logical choice.

Please provide a better description.

On Nov 26, 2012 7:30 AM, "sangeetha.prem" 
wrote:

> **
> Hi All,
> *
> Scenario:*
> *The configuration is as below:*
> Hunt group #1 has extension 12345. It should ring user 209 for 30 seconds,
> "At the same time"  user 208 should ring for 30 seconds, set fall back
> destination to user 207 and Allow call forwarding enabled. For user 208 set
> call forwarding to user 212.
> *Actual behaviour:* User 209 and user 208 should ring for 30 sec no one
> answers then user 212 starts ringing no one answers then user 207 in fall
> back destination rings.
>
>  When none of the users in the hunt group answer the call, extension
> in fall back destination should ring or user level call forwarding should
> ring. What is the expected behavior?
>
> Thanks
> Sangeetha
>
>
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Re: [sipx-dev] [Fwd: Re: Query regarding hunt group]

[Tony Graziano]

The wiki is clear to not have the same user in the hunt group more than 1
time.
On Nov 26, 2012 7:41 AM, "George Niculae"  wrote:

> On Mon, Nov 26, 2012 at 2:39 PM, rekha.h  wrote:
>
>> **
>> Hi,
>>
>> If  a duplicate user is not supported in the hunt group, then we feel it
>> should give an error message when a duplicate user is added to the same
>> hunt group. So can we raise an  issue for the same please suggest.
>>
>>
> Right, UI shouldn't allow, though not 100% sure about my prev statement,
> anyone else?
>
> George
>
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Re: [sipx-users] Sendmail Issue

[Tony Graziano]

If they are not using any pots or catb services  is the actiontec, replace
it since it would be acting simply as an Ethernet router. Good luck!
On Nov 25, 2012 4:06 AM, "Todd Hodgen"  wrote:

> I suspect your issue is with Verizon, and not the router.   They are
> blocking your ability to send email through them as an SMTP gateway.
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
> Sent: Saturday, November 24, 2012 7:45 PM
> To: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] Sendmail Issue
>
>
>
> Just to update this thread. I took my new instance of SipXecs to the
> customer prem to start my install. Setup their firewall and SipXecs and
> when
> I was done I decided to try to see if sendmail would work their. Sure
> enough
> first try the test mail delivered to my email account and worked just fine.
> Tested UM and that worked flawlessly as well
>
> I am assuming that the Actiontec router provided by Verizon is not working
> properly. I have all the same ports open on both firewalls with differing
> results. I am going to swap out the router on Monday and see if that fixes
> my issue.
> Thank you for all your help guys
> --
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> Dome Technologies
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Re: [sipx-dev] Exception in media relay causes sipxbridge to end itsp calls with a BYE

[Tony Graziano]

It is a requirement of the offer-answer model that re-invite with no SDP
attribute return an offer. This offer can be the same as the original offer
but it is a protocol error to return OK with no SDP body.

It is likely If you dig a packet capture or siptrace out you will see this.
It would help a lot if you explained the call flow and UA's involved. This
most likely occurs during  hold or transfer.
On Nov 23, 2012 4:51 AM, "Kemal Eroğlu"  wrote:

> Hello,
>
> Recently I got an issue about sipxbridge and media relay in release-4.4. I
> did some analysis, but stucked at some point.
>
> To summarize the problem,
>
> sipxbridge is ending some trunk calls with a BYE message that contains:
> "sipxbridge;cause=204;text=Unexpected exception processing response"
> (Calls are ended during call setup, generally before sending "Trying" to
> the INVITE we received.)
>
> In sipxbridge log, the exception is:
> RtpTransmitterEndpoint:"Unexpected exception "
>  org.sipfoundry.sipxrelay.SymmitronException: Error in processing request
> Attempt to set transmitter but Receiver not set.
>
> Actually the exception occurs in media relay:
>  sipxrelay:"Exception setting destination "
> java.lang.IllegalStateException: Attempt to set transmitter but Receiver
> not set.
>
> After analyzing the logs, I saw that, while RTP session (bridge,
> symmitrons, session) is being constructed in relay, somehow the
> corresponding client transaction is terminated by bridge. This triggers
> relay to delete the resources constructed so far. And then, relay tries to
> achieve one of the deleted sources. (It attempts to set the transmitter of
> a sym, but it is deleted) That gives the exception and the call is finished
> by a BYE from sipxbridge.
>
> I could not find why the transaction is terminated, but in sipXbridge log,
> this line is the initial point:
> Timer-1::SipListenerImpl:"Transaction terminated event"
>
> Do you have any idea about this?
>
> Best Regards,
>
> --
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> Karel Electronics R&D Center
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Re: [sipx-users] sipXecs 4.6/bria password reset design bug & login screen wording

[Tony Graziano]

Could item be moved where folks could comment on it?

I've thought at first maybe the user could request a pull from the user
portal, but that didn't make sense since the UA does that already.

The only thing that makes sense is that the config gets auto pushed for
that phone type of a user change (password), then the user can logout/login
to force the config refresh. Perhaps an email message could be auto
generated.
On Nov 22, 2012 5:15 AM, "Mircea Carasel"  wrote:

>
>
> On Thu, Nov 22, 2012 at 12:06 PM, Michael Picher wrote:
>
>> We also have a request in our internal tracker for automatically sending
>> bria profiles after password change...
>>
>> This tracker item probably should have been in the open source tracker.
>>
> Yes, this should have been in the open source tracker, but anyway, I think
> we can use the internal one to address the problem
> mircea
>
>>
>> Mike
>>
>>
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Re: [sipx-users] Yealink SIP-T26P/T28P + EXP38

[Tony Graziano]

I would be less concerned with auto discovery (wasn't that dropped from
sipx already?) and more concerned with auto provisioning and firmware
management.

It is good to plug in the deice and have it grab a basic config from the
server so the admin can find the phone in unconfigured devices, assign a
line and send it profiles/reboot, etc. like we can with Polycom.

The "auto-discovery" term harkens back to the day when sipx actually could
scan network scopes looking for known mac address ranges. I think that was
neat but it was really problematic and it dropped off the development map
and was removed.

I am not sure if auto disocvery (the term) is used the same as I remember
it. sipx sees a known mac address range (like 0004f2) and says "polycom"
and makes it a manageable unprovisioned device. I think this is a good goal
for any UA if the vendors make it possible.

On Wed, Nov 21, 2012 at 9:58 AM, Михаил Родионов  wrote:

> Hello,
>
> Our current plugin does firmware management and provisioning, no
> auto-discovery yet.
>
> Yealink quality is very good right now - less than 1% units fail in first
> year.
>
> 2012/11/21 Matt White 
>
>> Have you guys done any work outside the plugin with
>> auto-discovery/provisioning?  That would be a killer feature that only the
>> polycoms have right now.
>>
>> About 2 years  ago (or has it been longer) we were working with yealink
>> and we too developed a plugin that worked fairly well and had just started
>> looking at the auto-provisioning plugin.  If I recall, back in the 4.2 days
>> it wasn't a huge change, the current provisioning model is hardcoded to
>> polycom mac address schemes, but could be updated with yealink mac's with
>> little effort.
>>
>>  At the time, I was very impressed with yealink support.  They would have
>> an engineer write custom firmware for feature xyz and email it to me the
>> next day.  But ultimately we decided to not purse it anymore because they
>> had QC issues with the hardware.   We had a sizable deployment of about 300
>> phones but about 25% would not grab dhcp.  Yealink discovered an issue with
>> the crystal used for timing and needed to go back to manufacturing.
>> Yealink provded the phone for free but it still left us weary until they
>> had matured.
>>
>> Maybe they have stabilized a bit these days.
>>
>> -M
>>
>>
>>
>>
>>
>> >>> Михаил Родионов 11/20/12 7:06 PM >>>
>> Well, I woke up and found this thread... I think I just have to defend
>> Yealink here. We support installations with hundreds of them and they work
>> flowlessly. Yealink's SIP support is outstanding - they are a real
>> competitor to Polycom here.
>>
>> We have 4.6-compatible version of plugin for phones with upcoming v70 t2x
>> firmware in beta, will release it as RPM next week (we have 4.6
>> installation for 100 yealinks planned there so time frame is already set
>> for this release).
>>
>> So go for Yealinks and please address all questions and issues to me
>> directly - we have direct support from Yealink R&D so even custom firmware
>> is often no problem.
>>
>> 2012/11/21 Michael Picher 
>>
>>> I think it's in 4.7 now...  somebody will pipe up and let me know if I'm
>>> wrong.
>>>
>>> It was getting pushed and prodded in and out of 4.6.
>>>
>>> If we can get enough testing with it we might back-port it to 4.6.0
>>> update 1 or update 2.  Not sure how that's all going to develop yet.
>>>
>>> Mike
>>>
>>>
>>> On Tue, Nov 20, 2012 at 3:43 PM, Tony Graziano <
>>> tgrazi...@myitdepartment.net> wrote:
>>>
>>>> Is vvx with firmware 4.x going to be on 4.6 or 4.7 initially or has it
>>>> made it that far yet?
>>>>  On Nov 20, 2012 3:35 PM, "Michael Picher"  wrote:
>>>>
>>>>> I think we'll get there with the others...  but they are still a
>>>>> compromise.  I think some firmware and template updates might be coming
>>>>> down post 4.6.0 for Yealink.
>>>>>
>>>>> I also know the Polycom vvx500 / ver 4.x firmware stuff should drop in
>>>>> mid to late December...  we'll want some testers on the 4.7 branch.  We'll
>>>>> probably be dog-fooding it once 4.6 drops too.
>>>>>
>>>>> There might be a couple other phones in the works too ;-)
>>>>>
>>>>> Mike
>>>>>
>>>>>
>>>>> 

Re: [sipx-users] Call transfer from AutoAttendant to PSTN

[Tony Graziano]

I clearly state that if you are not going to actively use it, to leave it
disabled.

It works, and would handle this use case if the directions were followed.

I agree its a security risk, but if that is the use case requirement, so be
it. At least the permissions can be forced at the ACC to disallow certain
call types if desired.
On Nov 21, 2012 7:46 AM, "Michael Picher"  wrote:

> huh, a new use I didn't know about :-)
>
> now supporting DISA !
>
> not sure if that's a good thing, but if folks want to open themselves up,
> so be it.
>
>
> On Wed, Nov 21, 2012 at 6:10 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> You use *81 as an AA destination for option "9". Then use the authcode
>> like anyone inside would. That is the while purpose of it. Read the wiki
>> page.
>> On Nov 20, 2012 10:41 PM, "Todd Hodgen"  wrote:
>>
>>> You could set up multiple extensions with call forwarding to a particular
>>> number.   Ext 111 - 206-4567890, Ext 112 - 206-2254589, etc.   From auto
>>> attendant, dial the extension.
>>>
>>> If you explain what you are trying to do in more detail from an
>>> application
>>> standpoint, I suspect there are other suggestions that could be provided.
>>>
>>>
>>>
>>> -Original Message-
>>> From: sipx-users-boun...@list.sipfoundry.org
>>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of ??
>>> Sent: Tuesday, November 20, 2012 5:45 PM
>>> To: Discussion list for users of sipXecs software
>>> Subject: [sipx-users] Call transfer from AutoAttendant to PSTN
>>>
>>> Is there any way to make call transfer from AA to external PSTN ?
>>>
>>> Now I dial 100 to AA from my ext. and then dial prefix 9 with PSTN, I
>>> always
>>> get the voice "that ext is not valid" from AA. I can dial prefix 9 with
>>> PSTN
>>> directly to my MGW without problem.
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>>>
>>
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>>
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>
>
>
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Re: [sipx-users] Call transfer from AutoAttendant to PSTN

[Tony Graziano]

You use *81 as an AA destination for option "9". Then use the authcode like
anyone inside would. That is the while purpose of it. Read the wiki page.
On Nov 20, 2012 10:41 PM, "Todd Hodgen"  wrote:

> You could set up multiple extensions with call forwarding to a particular
> number.   Ext 111 - 206-4567890, Ext 112 - 206-2254589, etc.   From auto
> attendant, dial the extension.
>
> If you explain what you are trying to do in more detail from an application
> standpoint, I suspect there are other suggestions that could be provided.
>
>
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of ??
> Sent: Tuesday, November 20, 2012 5:45 PM
> To: Discussion list for users of sipXecs software
> Subject: [sipx-users] Call transfer from AutoAttendant to PSTN
>
> Is there any way to make call transfer from AA to external PSTN ?
>
> Now I dial 100 to AA from my ext. and then dial prefix 9 with PSTN, I
> always
> get the voice "that ext is not valid" from AA. I can dial prefix 9 with
> PSTN
> directly to my MGW without problem.
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Re: [sipx-users] Call transfer from AutoAttendant to PSTN

[Tony Graziano]

Which is why I'm suggesting what you are asking for is not a transfer from
as to dual a pstn number.

See ... USERS>AUTHORIZATION CODES

the wiki has a page on it too.
On Nov 20, 2012 9:41 PM, "文军"  wrote:

> The PSTN I want to transfer is not a fixed number.
>
>
> 在 2012-11-21，上午10:07，"Tony Graziano"  写道：
>
> You would have to set the pstn # as the "option" for 9. In other words
> pressing 9 might transfer the call to 5551234.
>
> I think what you are trying to do is use the authorization codes and
> that's not part of the as. See the wiki if this is the case.
> On Nov 20, 2012 8:46 PM, "文军"  wrote:
>
>> Is there any way to make call transfer from AA to external PSTN ?
>>
>> Now I dial 100 to AA from my ext. and then dial prefix 9 with PSTN, I
>> always get the voice "that ext is not valid" from AA. I can dial prefix 9
>> with PSTN directly to my MGW without problem.
>> ___
>> sipx-users mailing list
>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>
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Re: [sipx-users] Call transfer from AutoAttendant to PSTN

[Tony Graziano]

You would have to set the pstn # as the "option" for 9. In other words
pressing 9 might transfer the call to 5551234.

I think what you are trying to do is use the authorization codes and that's
not part of the as. See the wiki if this is the case.
On Nov 20, 2012 8:46 PM, "文军"  wrote:

> Is there any way to make call transfer from AA to external PSTN ?
>
> Now I dial 100 to AA from my ext. and then dial prefix 9 with PSTN, I
> always get the voice "that ext is not valid" from AA. I can dial prefix 9
> with PSTN directly to my MGW without problem.
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>

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Re: [sipx-users] Yealink SIP-T26P/T28P + EXP38

[Tony Graziano]

Is vvx with firmware 4.x going to be on 4.6 or 4.7 initially or has it made
it that far yet?
On Nov 20, 2012 3:35 PM, "Michael Picher"  wrote:

> I think we'll get there with the others...  but they are still a
> compromise.  I think some firmware and template updates might be coming
> down post 4.6.0 for Yealink.
>
> I also know the Polycom vvx500 / ver 4.x firmware stuff should drop in mid
> to late December...  we'll want some testers on the 4.7 branch.  We'll
> probably be dog-fooding it once 4.6 drops too.
>
> There might be a couple other phones in the works too ;-)
>
> Mike
>
>
> On Tue, Nov 20, 2012 at 2:55 PM, Geoff Musgrave <
> geoff.musgr...@cacionline.net> wrote:
>
>>  Mike/Tony,
>>
>>
>> Thanks for the replies. I had a feeling that was going to be the kind of
>> response I got but wanted to check first. 
>>
>> ** **
>>
>> I completely agree that going with Polycoms would solve all of our
>> problems.
>>
>> ** **
>>
>> Thanks for the info/insight.
>>
>> ** **
>>
>> --
>>
>> Geoff
>>
>> ** **
>>
>> *From:* Tony Graziano [mailto:tgrazi...@myitdepartment.net]
>> *Sent:* Tuesday, November 20, 2012 1:25 PM
>> *To:* Discussion list for users of sipXecs software
>> *Subject:* Re: [sipx-users] Yealink SIP-T26P/T28P + EXP38
>>
>> ** **
>>
>> Shoot thyself in the foot. Snom or Yealink, both are problematic. Get
>> thee to a Polycom...
>>
>> On Tue, Nov 20, 2012 at 2:11 PM, Geoff Musgrave <
>> geoff.musgr...@cacionline.net> wrote:
>>
>> Has anyone used the Yealink SIP-T26P or T28P with or without the EXP38
>> with sipXecs  4.6? If so, I would like to hear your experience and/or
>> opinion.
>>
>>  
>>
>> I’m considering placing 2 of these with the EXP38 at another office due
>> to the owners not wanting to spend Polycom prices. My other option is to go
>> with snom 3xx with expansion modules again but I’m still having some
>> transferring issues with the current snom 370s at the office we did the
>> initial deployment.
>>
>>  
>>
>> Thanks in advance.
>>
>>  
>>
>> --
>>
>> Geoff
>>
>>  
>>
>>
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>>
>>
>>
>> 
>>
>> ** **
>>
>> --
>> ~~
>> Tony Graziano, Manager
>> Telephone: 434.984.8430
>> sip: tgrazi...@voice.myitdepartment.net
>> Fax: 434.465.6833
>> ~~
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>> Ask about our Internet Fax services!
>> ~~
>>
>> ** **
>>
>> <http://sipxcolab2013.eventbrite.com/?discount=tony2013>
>> Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
>> 2013!
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>> [image: Description: Image removed by 
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>> 
>>
>>  ** **
>>
>> LAN/Telephony/Security and Control Systems Helpdesk:
>>
>> Telephone: 434.984.8426
>>
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>>
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>>
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>
>
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> M.207-956-0262
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> linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
> www.ezuce.com
>
>
> 
> There are 10 kinds of people in the world, those who understand binary and
> those who don't.
>
>
> ___
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> sipx-users@list.sipfoundry.org
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>

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Re: [sipx-users] Yealink SIP-T26P/T28P + EXP38

[Tony Graziano]

Shoot thyself in the foot. Snom or Yealink, both are problematic. Get thee
to a Polycom...

On Tue, Nov 20, 2012 at 2:11 PM, Geoff Musgrave <
geoff.musgr...@cacionline.net> wrote:

>  Has anyone used the Yealink SIP-T26P or T28P with or without the EXP38
> with sipXecs  4.6? If so, I would like to hear your experience and/or
> opinion.
>
> ** **
>
> I’m considering placing 2 of these with the EXP38 at another office due to
> the owners not wanting to spend Polycom prices. My other option is to go
> with snom 3xx with expansion modules again but I’m still having some
> transferring issues with the current snom 370s at the office we did the
> initial deployment.
>
> ** **
>
> Thanks in advance.
>
> ** **
>
> --
>
> Geoff
>
> ** **
>
> ___
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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Re: [sipx-users] voip.ms impression

[Tony Graziano]

Today alone I had two sites with a authentication issue. In one instance I
had two registrations that sipx said were authenticated. voip.ms says none
were. Restrating the gateways said they were authenticated, voip.ms still
says no. Incoming still worked even though it was not registered, which is
"wrong" but outbound would not work until we moved from seattle.

Moving one registration from seattle to NY and restart gateways... now
voip.ms agreed.

Same thing in Dallas, sipx says YES voip.ms says NO. Restart gateway, no.
Move to NY and restart, now voip.ms says yes.

I'm not particularly fond of their tech support. "It must be you". The
one constant in all this is voip.ms...

On Tue, Nov 20, 2012 at 1:22 PM, Burleigh, Matt <
matt.burle...@eiisolutions.net> wrote:

> I’ve recently(2 months) started using voip.ms and my support experience
> has been similar. Ever since Hurricane Sandy I’ve had numerous issues. I
> can usually restart SIP trunking to restore service and I don’t always get
> an alarm from sipx. I’ve had some recent complaints of busy signals as
> well...  
>
> ** **
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Mark Wood
> *Sent:* Monday, November 19, 2012 19:26
> *To:* 'sipx-users@list.sipfoundry.org'
> *Subject:* [sipx-users] voip.ms impression
>
> ** **
>
> We have been using voip.ms for 6-8 months so far and I want to get some
> feedback from users that have been at it for longer. 
>
> ** **
>
> Specifically we (main and subaccounts) experience times where our outbound
> calls just hang after dialing and sometimes abruptly connect, or sometimes
> not at all. When a subaccount calls to report problems to us and we check
> our home page it will show all of our accounts as ‘not registered’, and
> then slowly one by one they will show as ‘registered’. We had an incident
> over the weekend with a security office that couldn’t receive any inbound
> calls. We logged in to the voip.ms site to check the registrations and
> initiate a support ticket and the site again said ‘not registered’. The
> instructions had us do and ‘echo test’ procedure and the results were the
> same as when they were routed to the subaccount. The support response 12
> hours later was ‘works for us’ and then ‘check your routers and firewalls’.
> 
>
> ** **
>
> Comments? Who are other good candidates for reselling VoIP like this model?
> 
>
> ** **
>
> *Thanks,*
>
> *Mark W. Wood*
>
> *office:* (760)202-0224   X2010
>
> *Direct: *(760)459-1981
>
> *[image: New Image.BMP]*
>
> www.redphonetech.com
>
> ** **
>
> ** **
>
> * *
>
> ** **
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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~~
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Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
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~~
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Re: [sipx-users] voip.ms impression

[Tony Graziano]

We use voip.ms for small occasional use and its very good for testing. Our
production systems use Appia (we are a reseller) who can also provide t.38
service which we have found to work well in a production environment.

At the same time, most of our sip servers sit behind application aware
firewalls and ensure connections using SIP the protocol are in now way
impaired by other applications on the network. These application aware
devices also have IDS and are completely cloud managed and monitored.

I would agree voip.ms works but not always that well at every POP and they
do seem to have occasional problems with call quality and completion
depending on the number you call. Whether this is ISP routing, edge or
internal issues at their POP is sometimes a little unclear, but they are
good enough to use for failover, backup or testing but not particulary in
large production environments (IMO).

On Mon, Nov 19, 2012 at 7:25 PM, Mark Wood wrote:

> We have been using voip.ms for 6-8 months so far and I want to get some
> feedback from users that have been at it for longer. 
>
> ** **
>
> Specifically we (main and subaccounts) experience times where our outbound
> calls just hang after dialing and sometimes abruptly connect, or sometimes
> not at all. When a subaccount calls to report problems to us and we check
> our home page it will show all of our accounts as ‘not registered’, and
> then slowly one by one they will show as ‘registered’. We had an incident
> over the weekend with a security office that couldn’t receive any inbound
> calls. We logged in to the voip.ms site to check the registrations and
> initiate a support ticket and the site again said ‘not registered’. The
> instructions had us do and ‘echo test’ procedure and the results were the
> same as when they were routed to the subaccount. The support response 12
> hours later was ‘works for us’ and then ‘check your routers and firewalls’.
> 
>
> ** **
>
> Comments? Who are other good candidates for reselling VoIP like this model?
> 
>
> ** **
>
> *Thanks,*
>
> *Mark W. Wood*
>
> *office:* (760)202-0224   X2010
>
> *Direct: *(760)459-1981
>
> *[image: New Image.BMP]*
>
> www.redphonetech.com
>
> ** **
>
> ** **
>
> * *
>
> ** **
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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sip: tgrazi...@voice.myitdepartment.net
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~~
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Re: [sipx-users] Crashing proxy

[Tony Graziano]

It was and pushed into patch 23 posted today...

On Mon, Nov 19, 2012 at 11:15 AM, Michael Picher  wrote:

> I'm not so sure that those were pushed back to 4.4...
>
>
> On Mon, Nov 19, 2012 at 10:29 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> Have you looked at the recent thread regarding RLS and forwards, to which
>> there was a patch posted on the list last week? I am wondering if this
>> applies to your situation.
>>
>>
>> On Mon, Nov 19, 2012 at 10:22 AM, Elwin Formsma wrote:
>>
>>>  Hi George,
>>>
>>>  In that case, about 500 RLS users.
>>>
>>>
>>>  Kind regards,
>>>  Met vriendelijke groet,
>>>
>>>
>>> Elwin Formsma
>>> Telecats BV
>>> -
>>> Elwin Formsma | Telecats bv | KvK Enschede 06069106 | Tel:   053 488 99
>>> 44 | Fax: 053 488 99 10 | E-mail: e.form...@telecats.nl |
>>>
>>>  Op 19 nov. 2012, om 16:18 heeft George Niculae  het
>>> volgende geschreven:
>>>
>>>  On Mon, Nov 19, 2012 at 5:11 PM, Elwin Formsma 
>>> wrote:
>>>
>>>> Hi Michael,
>>>>
>>>>  4.4.0 #16
>>>> +- 550 users
>>>> There shouldnt be much RLS usage since no devices have been configured
>>>> to use it.
>>>>
>>>
>>>  Well if you have IM enabled for all users you'll have RLS subscribing
>>> for all that extensions
>>>
>>>  George
>>>  ___
>>> sipx-users mailing list
>>> sipx-users@list.sipfoundry.org
>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>>
>>>
>>>
>>> ___
>>> sipx-users mailing list
>>> sipx-users@list.sipfoundry.org
>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>>
>>
>>
>>
>> --
>> ~~
>> Tony Graziano, Manager
>> Telephone: 434.984.8430
>> sip: tgrazi...@voice.myitdepartment.net
>> Fax: 434.465.6833
>> ~~
>> Linked-In Profile:
>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>> Ask about our Internet Fax services!
>> ~~
>>
>> Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
>> 2013!
>>  <http://sipxcolab2013.eventbrite.com/?discount=tony2013>
>>
>>
>> LAN/Telephony/Security and Control Systems Helpdesk:
>> Telephone: 434.984.8426
>> sip: helpdesk@voice.myitdepartment.**net
>>
>> Helpdesk Customers: 
>> http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
>> Blog: http://blog.myitdepartment.net
>>
>> ___
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>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>
>
>
> --
> Michael Picher, Director of Technical Services
> eZuce, Inc.
>
> 300 Brickstone Square
>
> Suite 201
>
> Andover, MA. 01810
> O.978-296-1005 X2015
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> linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
> www.ezuce.com
>
>
> 
> There are 10 kinds of people in the world, those who understand binary and
> those who don't.
>
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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Re: [sipx-users] Crashing proxy

[Tony Graziano]

Have you looked at the recent thread regarding RLS and forwards, to which
there was a patch posted on the list last week? I am wondering if this
applies to your situation.

On Mon, Nov 19, 2012 at 10:22 AM, Elwin Formsma wrote:

>  Hi George,
>
>  In that case, about 500 RLS users.
>
>
>  Kind regards,
>  Met vriendelijke groet,
>
>
> Elwin Formsma
> Telecats BV
> -
> Elwin Formsma | Telecats bv | KvK Enschede 06069106 | Tel:   053 488 99 44
> | Fax: 053 488 99 10 | E-mail: e.form...@telecats.nl |
>
>  Op 19 nov. 2012, om 16:18 heeft George Niculae  het
> volgende geschreven:
>
>  On Mon, Nov 19, 2012 at 5:11 PM, Elwin Formsma wrote:
>
>> Hi Michael,
>>
>>  4.4.0 #16
>> +- 550 users
>> There shouldnt be much RLS usage since no devices have been configured to
>> use it.
>>
>
>  Well if you have IM enabled for all users you'll have RLS subscribing
> for all that extensions
>
>  George
>  ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
>
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

Thanks for the JIRA.
On Nov 16, 2012 9:16 PM, "Noah Mehl"  wrote:

>  Tony,
>
>  You know what?  I think everyone is clear on *YOUR* opinion on the
> matter.
>
>  In *MY* opinion, this is a serious bug.  I have created a Jira story:
>
>  http://track.sipfoundry.org/browse/XX-10529
>
>  Next time, I would appreciate constructive comments instead of: "This is
> only a problem for you...  You must be doing something wrong…  You're not
> setting a firewall/ids up correctly…."  I know I am not the only person who
> thinks this is a serious issue.
>
>  ~Noah
>
>
>  On Nov 16, 2012, at 7:30 PM, Tony Graziano 
> wrote:
>
>  That is with ssh open or available from the outside.
>
> I still suggest a JIRA...
> On Nov 16, 2012 6:41 PM, "Noah Mehl"  wrote:
>
>> I would also like to mention:
>>
>>  This works for any port, including SIP.  There might be huge amounts of
>> SIP piracy across peoples servers.
>>
>>  ~Noah
>>
>>  On Nov 16, 2012, at 6:27 PM, Alan Worstell 
>> wrote:
>>
>>  What Noah is posting about is correct. SMTP is listening on 127.0.0.1.
>> However, if you use SSH port redirection, from an outside host you can
>> forward your remote 127.0.0.1:25 to your own 127.0.0.1:25. I just tested
>> this with a development 4.6 server we have, from a system completely
>> off-network:
>> ssh -vN PlcmSpIp@{IP_OF_SIPX_SERVER} -L 25:127.0.0.1:25
>> After entering the password PlcmSpIp, I could telnet 127.0.0.1 25 and
>> send mail. I would consider that to be a pretty large security flaw, as
>> every sipx server out there that has SSH Password logins allowed to the
>> world can be used as spam relays.
>>
>> Regards,
>>
>> Alan Worstell
>> A1 Networks - Systems Administrator
>> VTSP, dCAA, LPIC-1, Linux+, CLA, DCTS
>> (707)570-2021 x204
>> For support issues please email supp...@a-1networks.com or call 707-703-1050
>>
>> On 11/16/12 3:17 PM, Tony Graziano wrote:
>>
>> can you provide the output of: lsof -i | grep LISTEN
>>
>>  and post what SMTP is listening to?
>>
>>
>>
>> On Fri, Nov 16, 2012 at 6:11 PM, Noah Mehl wrote:
>>
>>>  This is my problem:
>>>
>>>  You are arguing with me when you don't understand how SSH port
>>> forwarding works.
>>>
>>>  In the exploit I've illustrated, the port is tunneled via SSH. Then on
>>> the remote machine (the sipxecs server) the traffic originates as
>>> LOCALHOST. That's why it's a OOTB security flaw.
>>>
>>>  I have not made changes to the smtp config.
>>>
>>> ~Noah
>>>
>>> On Nov 16, 2012, at 6:02 PM, "Tony Graziano" <
>>> tgrazi...@myitdepartment.net> wrote:
>>>
>>>  There is that too. I keep bringing it up but he skips over it.
>>>
>>> In a default sipx installation, the output shows:
>>>
>>>  sendmail TCP localhost.localdomain:smtp (LISTEN)
>>>
>>>  and there are no other entries related to SMTP. So again, something is
>>> different here than in all the others (remember that kids game?). Why is
>>> your installation different? Why is SMTP open to begin with? Why is SMTP
>>> open on your system and noone else's?
>>>
>>>  I still don't agree with your assessment. It is the way your firewall
>>> and/or sendmail is configured to begin with that is not consistent with the
>>> way the system is used. Security is the admin's and certainly port SSH
>>> forward can be turned off and the user can be denied. I don't think it very
>>> helpful to make changes to secure a system if someone keeps opening holes
>>> or changing smtp configs and then opening another case that the system is
>>> not secure enough... I'm just saying. You still have neglected to explain
>>> why SMTP is open from w back in this thread.
>>>
>>>  Realize the developers list are some of the same people here (I won't
>>> dissuade you from posting to it to that list, or opening a JIRA) but
>>> realize it can be discussed and decided there is no problem and a change is
>>> not warranted, only an implementation decision gone awry.  On the other
>>> hand, if enough people agree those are two things that can be done by
>>> default "in the event someone decides to open SMTP". I'm not a fortune
>>> teller.
>>>
>>>  I think it took a lot of your time to find it and to bring it up, and
>>> 

Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

That is with ssh open or available from the outside.

I still suggest a JIRA...
On Nov 16, 2012 6:41 PM, "Noah Mehl"  wrote:

>  I would also like to mention:
>
>  This works for any port, including SIP.  There might be huge amounts of
> SIP piracy across peoples servers.
>
>  ~Noah
>
>  On Nov 16, 2012, at 6:27 PM, Alan Worstell 
> wrote:
>
>  What Noah is posting about is correct. SMTP is listening on 127.0.0.1.
> However, if you use SSH port redirection, from an outside host you can
> forward your remote 127.0.0.1:25 to your own 127.0.0.1:25. I just tested
> this with a development 4.6 server we have, from a system completely
> off-network:
> ssh -vN PlcmSpIp@{IP_OF_SIPX_SERVER} -L 25:127.0.0.1:25
> After entering the password PlcmSpIp, I could telnet 127.0.0.1 25 and send
> mail. I would consider that to be a pretty large security flaw, as every
> sipx server out there that has SSH Password logins allowed to the world can
> be used as spam relays.
>
> Regards,
>
> Alan Worstell
> A1 Networks - Systems Administrator
> VTSP, dCAA, LPIC-1, Linux+, CLA, DCTS
> (707)570-2021 x204
> For support issues please email supp...@a-1networks.com or call 707-703-1050
>
> On 11/16/12 3:17 PM, Tony Graziano wrote:
>
> can you provide the output of: lsof -i | grep LISTEN
>
>  and post what SMTP is listening to?
>
>
>
> On Fri, Nov 16, 2012 at 6:11 PM, Noah Mehl  wrote:
>
>>  This is my problem:
>>
>>  You are arguing with me when you don't understand how SSH port
>> forwarding works.
>>
>>  In the exploit I've illustrated, the port is tunneled via SSH. Then on
>> the remote machine (the sipxecs server) the traffic originates as
>> LOCALHOST. That's why it's a OOTB security flaw.
>>
>>  I have not made changes to the smtp config.
>>
>> ~Noah
>>
>> On Nov 16, 2012, at 6:02 PM, "Tony Graziano" <
>> tgrazi...@myitdepartment.net> wrote:
>>
>>  There is that too. I keep bringing it up but he skips over it.
>>
>> In a default sipx installation, the output shows:
>>
>>  sendmail TCP localhost.localdomain:smtp (LISTEN)
>>
>>  and there are no other entries related to SMTP. So again, something is
>> different here than in all the others (remember that kids game?). Why is
>> your installation different? Why is SMTP open to begin with? Why is SMTP
>> open on your system and noone else's?
>>
>>  I still don't agree with your assessment. It is the way your firewall
>> and/or sendmail is configured to begin with that is not consistent with the
>> way the system is used. Security is the admin's and certainly port SSH
>> forward can be turned off and the user can be denied. I don't think it very
>> helpful to make changes to secure a system if someone keeps opening holes
>> or changing smtp configs and then opening another case that the system is
>> not secure enough... I'm just saying. You still have neglected to explain
>> why SMTP is open from w back in this thread.
>>
>>  Realize the developers list are some of the same people here (I won't
>> dissuade you from posting to it to that list, or opening a JIRA) but
>> realize it can be discussed and decided there is no problem and a change is
>> not warranted, only an implementation decision gone awry.  On the other
>> hand, if enough people agree those are two things that can be done by
>> default "in the event someone decides to open SMTP". I'm not a fortune
>> teller.
>>
>>  I think it took a lot of your time to find it and to bring it up, and I
>> think its worthy of consideration though.
>>
>> On Fri, Nov 16, 2012 at 5:50 PM, Noah Mehl wrote:
>>
>>> Hey!  FINALLY, I got some information that's actually usefully to me!!!
>>> Where is the JIRA link where I can post a bug?  Is there a different
>>> mailing list for Sipxecs dev?
>>>
>>>  No, my argument is that two users are created by the SipXecs install:
>>> PlcmSIp and lvp2890.  These user have passwords set in the /etc/shadow from
>>> the install script.
>>>
>>>  I do not believe that this is a Redhat/Centos problem, because they DO
>>> NOT ship system users with passwords in /etc/shadow. Or any user with a
>>> password in /etc/shadow except for the password one sets for root during
>>> install, and the password for the first user during install.
>>>
>>>  Since SipXecs install creates these users, and thereby creates the
>>> security issue, part of the user creation should deny those users a

Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

can you provide the output of: lsof -i | grep LISTEN

and post what SMTP is listening to?



On Fri, Nov 16, 2012 at 6:11 PM, Noah Mehl  wrote:

>  This is my problem:
>
>  You are arguing with me when you don't understand how SSH port
> forwarding works.
>
>  In the exploit I've illustrated, the port is tunneled via SSH. Then on
> the remote machine (the sipxecs server) the traffic originates as
> LOCALHOST. That's why it's a OOTB security flaw.
>
>  I have not made changes to the smtp config.
>
> ~Noah
>
> On Nov 16, 2012, at 6:02 PM, "Tony Graziano" 
> wrote:
>
>  There is that too. I keep bringing it up but he skips over it.
>
> In a default sipx installation, the output shows:
>
>  sendmail TCP localhost.localdomain:smtp (LISTEN)
>
>  and there are no other entries related to SMTP. So again, something is
> different here than in all the others (remember that kids game?). Why is
> your installation different? Why is SMTP open to begin with? Why is SMTP
> open on your system and noone else's?
>
>  I still don't agree with your assessment. It is the way your firewall
> and/or sendmail is configured to begin with that is not consistent with the
> way the system is used. Security is the admin's and certainly port SSH
> forward can be turned off and the user can be denied. I don't think it very
> helpful to make changes to secure a system if someone keeps opening holes
> or changing smtp configs and then opening another case that the system is
> not secure enough... I'm just saying. You still have neglected to explain
> why SMTP is open from w back in this thread.
>
>  Realize the developers list are some of the same people here (I won't
> dissuade you from posting to it to that list, or opening a JIRA) but
> realize it can be discussed and decided there is no problem and a change is
> not warranted, only an implementation decision gone awry.  On the other
> hand, if enough people agree those are two things that can be done by
> default "in the event someone decides to open SMTP". I'm not a fortune
> teller.
>
>  I think it took a lot of your time to find it and to bring it up, and I
> think its worthy of consideration though.
>
> On Fri, Nov 16, 2012 at 5:50 PM, Noah Mehl  wrote:
>
>> Hey!  FINALLY, I got some information that's actually usefully to me!!!
>> Where is the JIRA link where I can post a bug?  Is there a different
>> mailing list for Sipxecs dev?
>>
>>  No, my argument is that two users are created by the SipXecs install:
>> PlcmSIp and lvp2890.  These user have passwords set in the /etc/shadow from
>> the install script.
>>
>>  I do not believe that this is a Redhat/Centos problem, because they DO
>> NOT ship system users with passwords in /etc/shadow. Or any user with a
>> password in /etc/shadow except for the password one sets for root during
>> install, and the password for the first user during install.
>>
>>  Since SipXecs install creates these users, and thereby creates the
>> security issue, part of the user creation should deny those users access to
>> ssh in the sshd_config.  That's the only part of this scenario that isn't
>> secure.  I will be happy to submit a bug, etc...
>>
>>  As it happens, I'm not the first person to be hacked because of this:
>> http://www.mail-archive.com/sipx-users@list.sipfoundry.org/msg04471.html And 
>> it's highly likely that many people have been bitten by this, and no
>> one knew what the cause was.
>>
>>  This serves as a warning to ALL SipXecs 4.4.x users:
>>
>>  1. If you have SipXecs 4.4.x
>> 2. You still have the PlcmSIp and lvp2890 users, with unchanged password
>> (which you would by default, not knowing they had been added to your server)
>> 3. Anyone has SSH port access to the server
>> 4. Then you are wide open
>>
>>  I don't care how one solves the issue, we have 3 solutions so far:
>>
>>  1. Disable or heavily restrict all ssh access to the machine
>> 2. DenyUsers PlcmSIp lvp2890 in /etc/ssh/sshd_config
>> 3. AllowTcpForwarding no in /etc/ssh/sshd_config
>>
>>  I prefer method 2 because I don't want to remove a useful tool in my
>> arsenal (ssh port forwarding), and I don't want to change the default
>> passwords (because of provision stock phones).  But I HIGHLY suggest
>> everyone takes a quick look at their settings, because I bet a lot of
>> people are susceptible to this.  Thanks.
>>
>>  ~Noah
>>
>>   On Nov 16, 2012, at 5:37 PM, Tony Graziano <
>> tgrazi...@myitdepartment.net>
>&g

Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

There is that too. I keep bringing it up but he skips over it.

In a default sipx installation, the output shows:

sendmail TCP localhost.localdomain:smtp (LISTEN)

and there are no other entries related to SMTP. So again, something is
different here than in all the others (remember that kids game?). Why is
your installation different? Why is SMTP open to begin with? Why is SMTP
open on your system and noone else's?

I still don't agree with your assessment. It is the way your firewall
and/or sendmail is configured to begin with that is not consistent with the
way the system is used. Security is the admin's and certainly port SSH
forward can be turned off and the user can be denied. I don't think it very
helpful to make changes to secure a system if someone keeps opening holes
or changing smtp configs and then opening another case that the system is
not secure enough... I'm just saying. You still have neglected to explain
why SMTP is open from w back in this thread.

Realize the developers list are some of the same people here (I won't
dissuade you from posting to it to that list, or opening a JIRA) but
realize it can be discussed and decided there is no problem and a change is
not warranted, only an implementation decision gone awry.  On the other
hand, if enough people agree those are two things that can be done by
default "in the event someone decides to open SMTP". I'm not a fortune
teller.

I think it took a lot of your time to find it and to bring it up, and I
think its worthy of consideration though.

On Fri, Nov 16, 2012 at 5:50 PM, Noah Mehl  wrote:

>  Hey!  FINALLY, I got some information that's actually usefully to me!!!
> Where is the JIRA link where I can post a bug?  Is there a different
> mailing list for Sipxecs dev?
>
>  No, my argument is that two users are created by the SipXecs install:
> PlcmSIp and lvp2890.  These user have passwords set in the /etc/shadow from
> the install script.
>
>  I do not believe that this is a Redhat/Centos problem, because they DO
> NOT ship system users with passwords in /etc/shadow. Or any user with a
> password in /etc/shadow except for the password one sets for root during
> install, and the password for the first user during install.
>
>  Since SipXecs install creates these users, and thereby creates the
> security issue, part of the user creation should deny those users access to
> ssh in the sshd_config.  That's the only part of this scenario that isn't
> secure.  I will be happy to submit a bug, etc...
>
>  As it happens, I'm not the first person to be hacked because of this:
> http://www.mail-archive.com/sipx-users@list.sipfoundry.org/msg04471.html And 
> it's highly likely that many people have been bitten by this, and no
> one knew what the cause was.
>
>  This serves as a warning to ALL SipXecs 4.4.x users:
>
>  1. If you have SipXecs 4.4.x
> 2. You still have the PlcmSIp and lvp2890 users, with unchanged password
> (which you would by default, not knowing they had been added to your server)
> 3. Anyone has SSH port access to the server
> 4. Then you are wide open
>
>  I don't care how one solves the issue, we have 3 solutions so far:
>
>  1. Disable or heavily restrict all ssh access to the machine
> 2. DenyUsers PlcmSIp lvp2890 in /etc/ssh/sshd_config
> 3. AllowTcpForwarding no in /etc/ssh/sshd_config
>
>  I prefer method 2 because I don't want to remove a useful tool in my
> arsenal (ssh port forwarding), and I don't want to change the default
> passwords (because of provision stock phones).  But I HIGHLY suggest
> everyone takes a quick look at their settings, because I bet a lot of
> people are susceptible to this.  Thanks.
>
>  ~Noah
>
>   On Nov 16, 2012, at 5:37 PM, Tony Graziano  >
>  wrote:
>
> You do realize the other side of this argument is that SSH forwarding is
> enabled by default on Redhat/Centos and that since you have SSH available
> to the public at large it also makes this an effective use of your system.
>
>  I think the place for you to ask for a change is submitting a JIRA and
> posting a link on the users and dev groups so people can comment and/or
> vote for this change...
>
>  add in /etc/ssh/sshd_config by default:
>
>  AllowTcpForwarding no
> DenyUsers PlcmSpIp
>
>
>
>
> On Fri, Nov 16, 2012 at 5:24 PM, Noah Mehl  wrote:
>
>> Shall I make a screencast to explain?
>>
>>  ~Noah
>>
>>  On Nov 16, 2012, at 5:20 PM, Noah Mehl  wrote:
>>
>>  Gerald.
>>
>>  That's the security hole.  I AM ABLE TO CONNECT TO THE LOCAL SMTP
>> SERVICE ON THE SIPXECS SERVER via SSH remotely using the default user/pass
>> of PlcmSIp, utilizing ssh port forwarding.
>>
>>  ~N

Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

You do realize the other side of this argument is that SSH forwarding is
enabled by default on Redhat/Centos and that since you have SSH available
to the public at large it also makes this an effective use of your system.

I think the place for you to ask for a change is submitting a JIRA and
posting a link on the users and dev groups so people can comment and/or
vote for this change...

add in /etc/ssh/sshd_config by default:

AllowTcpForwarding no
DenyUsers PlcmSpIp




On Fri, Nov 16, 2012 at 5:24 PM, Noah Mehl  wrote:

>  Shall I make a screencast to explain?
>
>  ~Noah
>
>  On Nov 16, 2012, at 5:20 PM, Noah Mehl  wrote:
>
>  Gerald.
>
>  That's the security hole.  I AM ABLE TO CONNECT TO THE LOCAL SMTP
> SERVICE ON THE SIPXECS SERVER via SSH remotely using the default user/pass
> of PlcmSIp, utilizing ssh port forwarding.
>
>  ~Noah
>
>  On Nov 16, 2012, at 5:17 PM, Gerald Drouillard 
> wrote:
>
>  On 11/16/2012 1:57 PM, Noah Mehl wrote:
>
> Does nobody on the list know what SSH port forwarding is?  I am running
> the first two commands from a remote machine (connecting to the sipxecs
> machine) in separate terminals to forward my local 25 port to the sipxecs
> box, and the 25 port on the sipxecs box locally.  The third command is run
> locally on the remote machine.  This exploit gives the remote machine
> access to port 25 on the SipXecs box even if all other ports are blocked.
>  This could be used for any port that is blocked by firewall, ids, etc, if
> the remote machine has ssh access to the sipxecs box.
>
>  ~Noah
>
> Do you understand that if your sipx smtp server is only running on
> localhost that you will not be able to connect to it via
> telnet/ssh/whatever?
>
>
> --
> Regards
> --
> Gerald Drouillard
> Technology Architect
> Drouillard & Associates, Inc.http://www.Drouillard.biz 
> <http://www.drouillard.biz/>
>
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>
>
>
>   ­­
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>
>
>
>   ­­
>
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>



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Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

It would mean the user would no longer be present on the system because it
would not be required.

On Fri, Nov 16, 2012 at 2:08 PM, Noah Mehl  wrote:

>  I don't understand:
>
>  "so polycom provisioning in Sipx will cease using ftp and the user
> account will be removed at that time and move to http/HTTPS."
>
>  Why would denying the PlcmSpIp user in the sshd config affect provision?
>
>  Honestly, the exploit is the ability to use SSH port forwarding with the
> default PlcmSpIp user/pass.  I doubt your ids will stop that if you have
> ssh access to the machine.
>
>  ~Noah
>
>  On Nov 16, 2012, at 1:48 PM, Tony Graziano 
> wrote:
>
>  Fwiw I can test the exploit and my ids (commercial snort rules).
>
> so polycom provisioning in Sipx will cease using ftp and the user account
> will be removed at that time and move to http/HTTPS.
> On Nov 16, 2012 12:52 PM, "Noah Mehl"  wrote:
>
>> I can confirm that adding:
>>
>>  DenyUsers PlcmSpIp
>>
>>  to /etc/ssh/sshd_config solves this exploit.
>>
>>  I'm back to my original opinion that if this user is installed
>> automatically, without my intervention, then that line should be added to
>> the sshd_config.
>>
>>  ~Noah
>>
>>  On Nov 16, 2012, at 12:46 PM, Noah Mehl  wrote:
>>
>>  Tony,
>>
>>  I just figured out an exploit in 15 minutes with the help of Google
>> http://www.semicomplete.com/articles/ssh-security/:
>>
>>  $sudo ssh -vN -L25:localhost:25 PlcmSpIp@sipxecsip
>> $sudo ssh -vN -R25:localhost:25 PlcmSpIp@sipxecsip
>> $telnet localhost 25
>>
>>  Tell me if your ids stops that?
>>
>>  This works on a stock SipXecs 4.4.0 install.
>>
>>  ~Noah
>>
>>  On Nov 16, 2012, at 11:46 AM, Tony Graziano <
>> tgrazi...@myitdepartment.net>
>>  wrote:
>>
>>  The user doesn't have login via ssh. Ssh in and of itself is not
>> protected and it is exposed.
>>
>> It is trivial to change the user password and/or delete it. We typically
>> don't expose ssh at all. You haven't provides any real evidence that a
>> dictionary attack didn't overwhelm the pam service either.
>>
>> I don't share your opinion here. My firewall protects against all kinds
>> of ids stuff even if I had ssh open. Just because you have iptables running
>> it doesn't mean you are inherently secure at all.
>>
>> Our firewalls sitting in front of sipx had ids rules running that would
>> protect anything behind it from a known attack against a well known service
>> like ssh. Ssh has lots of options which should be exercised according to
>> your security border device.
>> On Nov 16, 2012 11:36 AM, "Noah Mehl"  wrote:
>>
>>> The only hardening required to solve this particular problem would be an
>>> addition to the sshd config:
>>>
>>>  DenyUsers PlcmSpIp
>>>
>>>  I think this should be included in the default distribution of SipXecs
>>> isos and/or packages (I've only ever used the iso) because this is
>>> something that is specific to the distribution.  That user, and its
>>> password and access, are created by SipXecs, and that addition to the sshd
>>> config should be made OOTB.  Unless someone has a reason that PlcmSpIp
>>> should be able to have any ssh access?
>>>
>>>  I'd really like some input from someone from eZuce, as this is an easy
>>> solution and protects the entire community.
>>>
>>>  This was NOT a DDOS attack.  This it that the PlcmSpIp user has a
>>> default password of PlcmSpIp, and there's something about the default
>>> access of that user that allow remote execution via SSH OOTB, and that *
>>> IS* a security issue.  You know why?  Because as far as I know, no
>>> other default linux service account is susceptible to this attack.
>>>  Probably because linux system accounts DON'T HAVE PASSWORDS!  In other
>>> words, if you're creating service users with default passwords, they
>>> probably should be denied from ssh OOTB.  This is also, not documented as
>>> far as I can tell...
>>>
>>>  ~Noah
>>>
>>>  On Nov 16, 2012, at 11:26 AM, Tony Graziano <
>>> tgrazi...@myitdepartment.net> wrote:
>>>
>>>  It really sounds like you don't have a method to harden your server if
>>> you are exposing it. Its entirely possible you were targeted with a ddos
>>> attack that overwhelmed the Linux system. If you had properly crafted
>>> 

Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

Fwiw I can test the exploit and my ids (commercial snort rules).

so polycom provisioning in Sipx will cease using ftp and the user account
will be removed at that time and move to http/HTTPS.
On Nov 16, 2012 12:52 PM, "Noah Mehl"  wrote:

>  I can confirm that adding:
>
>  DenyUsers PlcmSpIp
>
>  to /etc/ssh/sshd_config solves this exploit.
>
>  I'm back to my original opinion that if this user is installed
> automatically, without my intervention, then that line should be added to
> the sshd_config.
>
>  ~Noah
>
>  On Nov 16, 2012, at 12:46 PM, Noah Mehl  wrote:
>
>  Tony,
>
>  I just figured out an exploit in 15 minutes with the help of Google
> http://www.semicomplete.com/articles/ssh-security/:
>
>  $sudo ssh -vN -L25:localhost:25 PlcmSpIp@sipxecsip
> $sudo ssh -vN -R25:localhost:25 PlcmSpIp@sipxecsip
> $telnet localhost 25
>
>  Tell me if your ids stops that?
>
>  This works on a stock SipXecs 4.4.0 install.
>
>  ~Noah
>
>  On Nov 16, 2012, at 11:46 AM, Tony Graziano  >
>  wrote:
>
>  The user doesn't have login via ssh. Ssh in and of itself is not
> protected and it is exposed.
>
> It is trivial to change the user password and/or delete it. We typically
> don't expose ssh at all. You haven't provides any real evidence that a
> dictionary attack didn't overwhelm the pam service either.
>
> I don't share your opinion here. My firewall protects against all kinds of
> ids stuff even if I had ssh open. Just because you have iptables running it
> doesn't mean you are inherently secure at all.
>
> Our firewalls sitting in front of sipx had ids rules running that would
> protect anything behind it from a known attack against a well known service
> like ssh. Ssh has lots of options which should be exercised according to
> your security border device.
> On Nov 16, 2012 11:36 AM, "Noah Mehl"  wrote:
>
>> The only hardening required to solve this particular problem would be an
>> addition to the sshd config:
>>
>>  DenyUsers PlcmSpIp
>>
>>  I think this should be included in the default distribution of SipXecs
>> isos and/or packages (I've only ever used the iso) because this is
>> something that is specific to the distribution.  That user, and its
>> password and access, are created by SipXecs, and that addition to the sshd
>> config should be made OOTB.  Unless someone has a reason that PlcmSpIp
>> should be able to have any ssh access?
>>
>>  I'd really like some input from someone from eZuce, as this is an easy
>> solution and protects the entire community.
>>
>>  This was NOT a DDOS attack.  This it that the PlcmSpIp user has a
>> default password of PlcmSpIp, and there's something about the default
>> access of that user that allow remote execution via SSH OOTB, and that *
>> IS* a security issue.  You know why?  Because as far as I know, no other
>> default linux service account is susceptible to this attack.  Probably
>> because linux system accounts DON'T HAVE PASSWORDS!  In other words, if
>> you're creating service users with default passwords, they probably should
>> be denied from ssh OOTB.  This is also, not documented as far as I can
>> tell...
>>
>>  ~Noah
>>
>>  On Nov 16, 2012, at 11:26 AM, Tony Graziano <
>> tgrazi...@myitdepartment.net> wrote:
>>
>>  It really sounds like you don't have a method to harden your server if
>> you are exposing it. Its entirely possible you were targeted with a ddos
>> attack that overwhelmed the Linux system. If you had properly crafted
>> iptables rules I and ssh protection mechanisms it would most likely not
>> have happened.
>>
>> Any did or ddos can overwhelm system services to the point of failure
>> this allowing (by unavailability) internal logging or protection
>> mechanisms. Put the served behind a firewall and protect the vulnerable
>> service (ssh) by limiting the footprint. Backup the system, wipe and
>> restore it in the event a root kit was planted.
>>
>> I don't think iptables was adequately configured. I don't think there is
>> anything inherently wrong with Sipx here either.
>>
>> It is a phone system. It is up to you to protect and/or harden it. Any
>> vulnerabilities exposed are really Linux vulnerabilities and Linux is not
>> hack proof.
>>
>> Good luck.
>> On Nov 16, 2012 10:07 AM, "Noah Mehl"  wrote:
>>
>>> Todd,
>>>
>>> The private subnet is: 172.16.0.0 - 172.31.255.255.  That IP is a public
>>> IP address, which is part of AOL in Nevada I

Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

But there again SMTP is for some reason open on that machine and unless you
are also using it as a mail server I don't see the point in making it
available to the public at large. Send mail does not need to have SMTP open
in order to send. This is yet another thing that confuses me about your
firewall arrangements.
On Nov 16, 2012 1:34 PM, "Gerald Drouillard" 
wrote:

>  On 11/16/2012 12:45 PM, Noah Mehl wrote:
>
> Tony,
>
>  I just figured out an exploit in 15 minutes with the help of Google
> http://www.semicomplete.com/articles/ssh-security/:
>
>  $sudo ssh -vN -L25:localhost:25 PlcmSpIp@sipxecsip
> $sudo ssh -vN -R25:localhost:25 PlcmSpIp@sipxecsip
> $telnet localhost 25
>
>
>  Of course you can telnet to port 25 (smtp) on the server to localhost.
> You have sendmail running on local host.  If your sendmail is configured
> properly you will not be able to access port 25 for another machine or the
> real ip address of the server.
>
> --
> Regards
> --
> Gerald Drouillard
> Technology Architect
> Drouillard & Associates, Inc.http://www.Drouillard.biz
>
>
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Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

Fwiw I can test the exploit and my ids (commercial snort rules).

Polycom provisioning in Sipx will cease using ftp and the user account will
be removed (most likely) when this is done.

Your exploit though appears to originate from inside your network though
doesn't it? If it originates inside it not passing any firewall except
iptables which implicitly allows the connection, right?
On Nov 16, 2012 12:52 PM, "Noah Mehl"  wrote:

>  I can confirm that adding:
>
>  DenyUsers PlcmSpIp
>
>  to /etc/ssh/sshd_config solves this exploit.
>
>  I'm back to my original opinion that if this user is installed
> automatically, without my intervention, then that line should be added to
> the sshd_config.
>
>  ~Noah
>
>  On Nov 16, 2012, at 12:46 PM, Noah Mehl  wrote:
>
>  Tony,
>
>  I just figured out an exploit in 15 minutes with the help of Google
> http://www.semicomplete.com/articles/ssh-security/:
>
>  $sudo ssh -vN -L25:localhost:25 PlcmSpIp@sipxecsip
> $sudo ssh -vN -R25:localhost:25 PlcmSpIp@sipxecsip
> $telnet localhost 25
>
>  Tell me if your ids stops that?
>
>  This works on a stock SipXecs 4.4.0 install.
>
>  ~Noah
>
>  On Nov 16, 2012, at 11:46 AM, Tony Graziano  >
>  wrote:
>
>  The user doesn't have login via ssh. Ssh in and of itself is not
> protected and it is exposed.
>
> It is trivial to change the user password and/or delete it. We typically
> don't expose ssh at all. You haven't provides any real evidence that a
> dictionary attack didn't overwhelm the pam service either.
>
> I don't share your opinion here. My firewall protects against all kinds of
> ids stuff even if I had ssh open. Just because you have iptables running it
> doesn't mean you are inherently secure at all.
>
> Our firewalls sitting in front of sipx had ids rules running that would
> protect anything behind it from a known attack against a well known service
> like ssh. Ssh has lots of options which should be exercised according to
> your security border device.
> On Nov 16, 2012 11:36 AM, "Noah Mehl"  wrote:
>
>> The only hardening required to solve this particular problem would be an
>> addition to the sshd config:
>>
>>  DenyUsers PlcmSpIp
>>
>>  I think this should be included in the default distribution of SipXecs
>> isos and/or packages (I've only ever used the iso) because this is
>> something that is specific to the distribution.  That user, and its
>> password and access, are created by SipXecs, and that addition to the sshd
>> config should be made OOTB.  Unless someone has a reason that PlcmSpIp
>> should be able to have any ssh access?
>>
>>  I'd really like some input from someone from eZuce, as this is an easy
>> solution and protects the entire community.
>>
>>  This was NOT a DDOS attack.  This it that the PlcmSpIp user has a
>> default password of PlcmSpIp, and there's something about the default
>> access of that user that allow remote execution via SSH OOTB, and that *
>> IS* a security issue.  You know why?  Because as far as I know, no other
>> default linux service account is susceptible to this attack.  Probably
>> because linux system accounts DON'T HAVE PASSWORDS!  In other words, if
>> you're creating service users with default passwords, they probably should
>> be denied from ssh OOTB.  This is also, not documented as far as I can
>> tell...
>>
>>  ~Noah
>>
>>  On Nov 16, 2012, at 11:26 AM, Tony Graziano <
>> tgrazi...@myitdepartment.net> wrote:
>>
>>  It really sounds like you don't have a method to harden your server if
>> you are exposing it. Its entirely possible you were targeted with a ddos
>> attack that overwhelmed the Linux system. If you had properly crafted
>> iptables rules I and ssh protection mechanisms it would most likely not
>> have happened.
>>
>> Any did or ddos can overwhelm system services to the point of failure
>> this allowing (by unavailability) internal logging or protection
>> mechanisms. Put the served behind a firewall and protect the vulnerable
>> service (ssh) by limiting the footprint. Backup the system, wipe and
>> restore it in the event a root kit was planted.
>>
>> I don't think iptables was adequately configured. I don't think there is
>> anything inherently wrong with Sipx here either.
>>
>> It is a phone system. It is up to you to protect and/or harden it. Any
>> vulnerabilities exposed are really Linux vulnerabilities and Linux is not
>> hack proof.
>>
>> Good luck.
>> On Nov 16, 2012 10:07 AM, "Noah Meh

Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

The user doesn't have login via ssh. Ssh in and of itself is not protected
and it is exposed.

It is trivial to change the user password and/or delete it. We typically
don't expose ssh at all. You haven't provides any real evidence that a
dictionary attack didn't overwhelm the pam service either.

I don't share your opinion here. My firewall protects against all kinds of
ids stuff even if I had ssh open. Just because you have iptables running it
doesn't mean you are inherently secure at all.

Our firewalls sitting in front of sipx had ids rules running that would
protect anything behind it from a known attack against a well known service
like ssh. Ssh has lots of options which should be exercised according to
your security border device.
On Nov 16, 2012 11:36 AM, "Noah Mehl"  wrote:

>  The only hardening required to solve this particular problem would be an
> addition to the sshd config:
>
>  DenyUsers PlcmSpIp
>
>  I think this should be included in the default distribution of SipXecs
> isos and/or packages (I've only ever used the iso) because this is
> something that is specific to the distribution.  That user, and its
> password and access, are created by SipXecs, and that addition to the sshd
> config should be made OOTB.  Unless someone has a reason that PlcmSpIp
> should be able to have any ssh access?
>
>  I'd really like some input from someone from eZuce, as this is an easy
> solution and protects the entire community.
>
>  This was NOT a DDOS attack.  This it that the PlcmSpIp user has a
> default password of PlcmSpIp, and there's something about the default
> access of that user that allow remote execution via SSH OOTB, and that *IS
> * a security issue.  You know why?  Because as far as I know, no other
> default linux service account is susceptible to this attack.  Probably
> because linux system accounts DON'T HAVE PASSWORDS!  In other words, if
> you're creating service users with default passwords, they probably should
> be denied from ssh OOTB.  This is also, not documented as far as I can
> tell...
>
>  ~Noah
>
>  On Nov 16, 2012, at 11:26 AM, Tony Graziano 
> wrote:
>
>  It really sounds like you don't have a method to harden your server if
> you are exposing it. Its entirely possible you were targeted with a ddos
> attack that overwhelmed the Linux system. If you had properly crafted
> iptables rules I and ssh protection mechanisms it would most likely not
> have happened.
>
> Any did or ddos can overwhelm system services to the point of failure this
> allowing (by unavailability) internal logging or protection mechanisms. Put
> the served behind a firewall and protect the vulnerable service (ssh) by
> limiting the footprint. Backup the system, wipe and restore it in the event
> a root kit was planted.
>
> I don't think iptables was adequately configured. I don't think there is
> anything inherently wrong with Sipx here either.
>
> It is a phone system. It is up to you to protect and/or harden it. Any
> vulnerabilities exposed are really Linux vulnerabilities and Linux is not
> hack proof.
>
> Good luck.
> On Nov 16, 2012 10:07 AM, "Noah Mehl"  wrote:
>
>> Todd,
>>
>> The private subnet is: 172.16.0.0 - 172.31.255.255.  That IP is a public
>> IP address, which is part of AOL in Nevada I think.  I actually have over
>> 80 different public IP address entries in my log using that user to SSH to
>> my SipXecs box.
>>
>> I understand that it's a phone system and not a firewall.  However it's a
>> linux server, and IPtables is the best firewall in world, IMHO.  I did have
>> SSH access open to the world, that was my choice.  I have never been bitten
>> by this before.  Either way, you should not be able to execute anything by
>> SSH'ing with the PlcmSpIp user, whether it's a public IP or not.
>>
>> ~Noah
>>
>> On Nov 15, 2012, at 7:07 PM, Todd Hodgen  wrote:
>>
>> > Here is a question I would have as well - 172.129.67.195 seems to be an
>> > address that is local to your network.   Who has that IP address, why
>> are
>> > they attempting to breach that server.   If they are not a part of your
>> > network, how are they getting to that server from outside your network -
>> > there has to be an opening in a firewall somewhere that is allowing it.
>> >
>> > Remember, this is a phone system, not a firewall, not a router.   It's a
>> > phone system with pretty standard authentication requirements, it's up
>> to
>> > the administrator to keep others off of the network.
>> >
>> > -Original Message-
>>

Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

icmp --  anywhere anywhereicmp any
> >> ACCEPT esp  --  anywhere anywhere
> >> ACCEPT ah   --  anywhere anywhere
> >> ACCEPT udp  --  anywhere 224.0.0.251 udp
> dpt:mdns
> >> ACCEPT udp  --  anywhere anywhereudp dpt:ipp
> >> ACCEPT tcp  --  anywhere anywheretcp dpt:ipp
> >> ACCEPT all  --  anywhere anywherestate
> >> RELATED,ESTABLISHED
> >> ACCEPT tcp  --  anywhere anywherestate NEW
> tcp
> >> dpt:pcsync-https
> >> ACCEPT tcp  --  anywhere anywherestate NEW
> tcp
> >> dpt:http
> >> ACCEPT tcp  --  anywhere anywherestate NEW
> tcp
> >> dpt:xmpp-client
> >> ACCEPT tcp  --  anywhere anywherestate NEW
> tcp
> >> dpt:5223
> >> ACCEPT all  --  192.168.0.0/16   anywhere
> >> ACCEPT udp  --  anywhere anywherestate NEW
> udp
> >> dpt:sip
> >> ACCEPT tcp  --  anywhere anywherestate NEW
> tcp
> >> dpt:sip
> >> ACCEPT tcp  --  anywhere anywherestate NEW
> tcp
> >> dpt:sip-tls
> >> ACCEPT udp  --  sip02.gafachi.comanywherestate NEW
> udp
> >> dpts:sip:5080
> >> ACCEPT udp  --  204.11.192.0/22  anywherestate NEW
> udp
> >> dpts:sip:5080
> >> REJECT all  --  anywhere anywherereject-with
> >> icmp-host-prohibited
> >>
> >> As far as I can tell, no one should be able to use port 25 from the
> world.
> >> Also, sendmail is only configured to allow relay from localhost:
> >>
> >> [root@sipx1 ~]# cat /etc/mail/access
> >> # Check the /usr/share/doc/sendmail/README.cf file for a description #
> >> of the format of this file. (search for access_db in that file) # The
> >> /usr/share/doc/sendmail/README.cf is part of the sendmail-doc # package.
> >> #
> >> # by default we allow relaying from localhost...
> >> Connect:localhost.localdomain   RELAY
> >> Connect:localhost   RELAY
> >> Connect:127.0.0.1   RELAY
> >>
> >> Can someone please help me figure out where this spam is coming from?
> >> Thanks.
> >>
> >> ~Noah
> >>
> >> On Oct 13, 2012, at 10:17 AM, Noah Mehl  wrote:
> >>
> >>> I did not change the configuration of anything related to the
> >>> PlcmSpIp
> >> user.  It does however make me feel better that it is related to the
> >> vsftpd service and the polycom phones.
> >>>
> >>>> From /etc/passwd:
> >>>
> >>> PlcmSpIp:x:500:500::/var/sipxdata/configserver/phone/profile/tftproot:
> >>> /sbin/nologin
> >>>
> >>> So, that user cannot ssh to a shell. So I don't think it was that.
> >>>
> >>> ~Noah
> >>>
> >>> On Oct 12, 2012, at 9:05 AM, Tony Graziano
> >>> 
> >> wrote:
> >>>
> >>>> ... more -- its a user that does not have login to the OS itself,
> >>>> just vsftpd, which is restricted to certain commands and must
> >>>> present a request for its mac address in order to get a configuration
> > file.
> >>>> It is not logging into linux unless someone changed the rights of
> >>>> the user.
> >>>>
> >>>> On Fri, Oct 12, 2012 at 7:30 AM, George Niculae 
> > wrote:
> >>>>> On Fri, Oct 12, 2012 at 2:26 PM, Tony Graziano
> >>>>>  wrote:
> >>>>>> this is not a valid system user unless you have manually added it
> >>>>>> to the system. I do think the logs would show more if access was
> >>>>>> granted. Why are you exposing sshd to the outside world with an
> >>>>>> acl or by protecting it at your firewall?
> >>>>>>
> >>>>>
> >>>>> PlcmSpIp is the user used by polycom phones for fetching config
> >>>>> from server
> >>>>>
> >>>>> George
> >>>>> ___
> >>>>> sipx-users mailing list
> >>>>> sipx-u

Re: [sipx-dev] Query on Call Status for a call from PSTN phone to Mobile number

[Tony Graziano]

Right, but the PSTN connects the call and the PSTN is sending AUDIO packets
in the form or ringing. As far as the audiocodes is concerned (whether it
is a busy signal, error recording or ringing or a person answering the
phone) it has successfully completed the call and connected it. It does not
and cannot know the difference accordingly (especially if it is an analog
line).

On Fri, Nov 16, 2012 at 6:28 AM, Kumaran <
thiru.venkateshwa...@ttplservices.com> wrote:

>  I just checked the scenario and send the behavior...No interop between
> with ITSP and gateway...I want to inform that Audiocodes sends 200OK before
> external users answer the call or not..
>
> Regards,
> Kumaran T
>
>
> On 11/16/2012 4:42 PM, Joegen Baclor wrote:
>
> Then make up your mind.  Are you reporting an audio code bug or
> investigating why it is connected in CDRs?  It is conencted in CDR because
> IT IS CONNECTED.  If you are investigating audiocodes interop with ITSP,
> please send a packet capture directly showing packet exhcnge between this
> two components and not from home.
>
> On 11/16/2012 06:57 PM, Kumaran wrote:
>
> No Joegen,its not successful call.Audiocodes sends 200 OK.Either external
> users  answers or not  there will 200OK from Audiocodes.So CDR shows
> Completed instead of Abandon.But this behavior is not seen in ITSP...
>
> Regards,
> Kumaran T
>
> On 11/16/2012 1:37 PM, Joegen Baclor wrote:
>
> This is a successful call.  Why do you expect this to be flagged as
> abandoned?
>
>
> On 11/16/2012 01:11 PM, sangeetha.prem wrote:
>
> Hi,
>
> Please find the pcap file attached.
>
> Thanks
> Sangeetha
>
> --
>
>
>  Original Message   Subject: Re: [sipx-dev] [Fwd: Re:
> Query on Call Status for a call from PSTN phone to Mobile number]  Date: Thu,
> 15 Nov 2012 19:21:39 +0800  From: Joegen Baclor 
>   Reply-To:
> sipXecs developer discussions 
>   To:
> sipXecs developer discussions 
>   References:
> <50a4ca75.5010...@qantom.com> <50a4ca75.5010...@qantom.com>
> 
>
> There is a chance that this is a race condition on call cancellation where
> CANCEL is being sent while 200 OK is already traversing the wire.  In this
> case, the call-termination mechanism will be a time-out in ACK.  However,
> CDR will show sucess because it base the connected flag on receipt of 200
> OK.  If this is the case, then it is not a bug.  Too many assumptions
> here.  a packet capture will shed light.
>
> On 11/15/2012 07:11 PM, Tony Graziano wrote:
>
> correct
>
> On Thu, Nov 15, 2012 at 5:56 AM, sangeetha.prem  > wrote:
>
>> Hi All,
>>
>> Thanks for the reply, so can i raise an issue since the call status is
>> displayed as "Completed" instead of "Abondoned" when the subscriber hung up
>> before there was an answer.
>>
>> Thanks
>> Sangeetha
>>
>>
>>
>> Tony Graziano wrote:
>>
>> if there was no answer
>>
>>  Subscriber --> PSTN
>>
>>  and the subscriber hung up before there was an answer, then the cdr
>> should say "abandoned".
>>
>> On Thu, Nov 15, 2012 at 5:30 AM, sangeetha.prem <
>> sangeetha.p...@qantom.com> wrote:
>>
>>> Hi All,
>>>
>>>Make a call from Phone 1 to Mobile number and do not answer the call
>>>  on Mobile or reject the call from mobile, In CDR historic page call
>>> status is displayed as "Completed" instead of "Abondoned" . What is the
>>> expected Call Status?
>>>
>>> Thanks
>>> Sangeetha
>>> ___
>>> sipx-dev mailing list
>>> sipx-dev@list.sipfoundry.org
>>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>>>
>>
>>
>>
>>  --
>> ~~
>> Tony Graziano, Manager
>> Telephone: 434.984.8430
>> sip: tgrazi...@voice.myitdepartment.net
>> Fax: 434.465.6833
>> ~~
>> Linked-In Profile:
>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>> Ask about our Internet Fax services!
>> ~~
>>
>>  Using or developing for sipXecs from SIPFoundry? Ask me about
>> sipX-CoLab 2013!
>>  <http://sipxcolab2013.eventbrite.com/?discount=tony2013>
>>
>>
>> LAN/Telephony/Security and Control Systems Helpdesk:
>> Telephone: 434.984.8426
>> sip: helpd...@voice.myitdepartment.net
>>
>>  Helpdesk Customers: http://myhelp.myitdepartment.net
>> Blog: http://blog.myitdepartment.net
>>
>> ---

Re: [sipx-dev] Query on Call Status for a call from PSTN phone to Mobile number

[Tony Graziano]

I think the difference is a behavioral one with the two sets of
gateways/providers.

The audiocodes sends the call to the PSTN and gets a "ringing" which it
sends back as "audio" which means the call has been established with the
outbound carrier. This is correct because there is media, even if it is
ringing.

The ITSP is sending plainer sip signalling because they (the provider) are
also the gateway. So the layer of visibility changes.

Neither is incorrect. I don't think there is an issue here.

If the audiocodes gateway was a PRI gateway using ISDN then maybe the
experience would be different, but I'm not sure it matters.

On Fri, Nov 16, 2012 at 5:57 AM, Kumaran <
thiru.venkateshwa...@ttplservices.com> wrote:

>  No Joegen,its not successful call.Audiocodes sends 200 OK.Either external
> users  answers or not  there will 200OK from Audiocodes.So CDR shows
> Completed instead of Abandon.But this behavior is not seen in ITSP...
>
> Regards,
> Kumaran T
>
> On 11/16/2012 1:37 PM, Joegen Baclor wrote:
>
> This is a successful call.  Why do you expect this to be flagged as
> abandoned?
>
>
> On 11/16/2012 01:11 PM, sangeetha.prem wrote:
>
> Hi,
>
> Please find the pcap file attached.
>
> Thanks
> Sangeetha
>
> --
>
>
>  Original Message   Subject: Re: [sipx-dev] [Fwd: Re:
> Query on Call Status for a call from PSTN phone to Mobile number]  Date: Thu,
> 15 Nov 2012 19:21:39 +0800  From: Joegen Baclor 
>   Reply-To:
> sipXecs developer discussions 
>   To:
> sipXecs developer discussions 
>   References:
> <50a4ca75.5010...@qantom.com> <50a4ca75.5010...@qantom.com>
> 
>
> There is a chance that this is a race condition on call cancellation where
> CANCEL is being sent while 200 OK is already traversing the wire.  In this
> case, the call-termination mechanism will be a time-out in ACK.  However,
> CDR will show sucess because it base the connected flag on receipt of 200
> OK.  If this is the case, then it is not a bug.  Too many assumptions
> here.  a packet capture will shed light.
>
> On 11/15/2012 07:11 PM, Tony Graziano wrote:
>
> correct
>
> On Thu, Nov 15, 2012 at 5:56 AM, sangeetha.prem  > wrote:
>
>> Hi All,
>>
>> Thanks for the reply, so can i raise an issue since the call status is
>> displayed as "Completed" instead of "Abondoned" when the subscriber hung up
>> before there was an answer.
>>
>> Thanks
>> Sangeetha
>>
>>
>>
>> Tony Graziano wrote:
>>
>> if there was no answer
>>
>>  Subscriber --> PSTN
>>
>>  and the subscriber hung up before there was an answer, then the cdr
>> should say "abandoned".
>>
>> On Thu, Nov 15, 2012 at 5:30 AM, sangeetha.prem <
>> sangeetha.p...@qantom.com> wrote:
>>
>>> Hi All,
>>>
>>>Make a call from Phone 1 to Mobile number and do not answer the call
>>>  on Mobile or reject the call from mobile, In CDR historic page call
>>> status is displayed as "Completed" instead of "Abondoned" . What is the
>>> expected Call Status?
>>>
>>> Thanks
>>> Sangeetha
>>> ___
>>> sipx-dev mailing list
>>> sipx-dev@list.sipfoundry.org
>>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>>>
>>
>>
>>
>>  --
>> ~~
>> Tony Graziano, Manager
>> Telephone: 434.984.8430
>> sip: tgrazi...@voice.myitdepartment.net
>> Fax: 434.465.6833
>> ~~
>> Linked-In Profile:
>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>> Ask about our Internet Fax services!
>> ~~
>>
>>  Using or developing for sipXecs from SIPFoundry? Ask me about
>> sipX-CoLab 2013!
>>  <http://sipxcolab2013.eventbrite.com/?discount=tony2013>
>>
>>
>> LAN/Telephony/Security and Control Systems Helpdesk:
>> Telephone: 434.984.8426
>> sip: helpd...@voice.myitdepartment.net
>>
>>  Helpdesk Customers: http://myhelp.myitdepartment.net
>> Blog: http://blog.myitdepartment.net
>>
>> --
>>
>> ___
>> sipx-dev mailing listsipx-...@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>>
>>
>> ___
>> sipx-dev mailing list
>> sipx-dev@list.sipfoundry.org
>> List Archive: http:

Re: [sipx-dev] SipXbridge MOH

[Tony Graziano]

Since sipxbridge handles refer internally and there is a reinvite, the UA
is still going to play MOH. Perhaps you want to test this with a phone that
doesnt have MOH configurd (or one that doesnt support it).

With polycom you can remove the MOH uri in sipxconfig and send the profile
to the phone to remove it and test.

On Fri, Nov 16, 2012 at 5:05 AM, Kumaran <
thiru.venkateshwa...@ttplservices.com> wrote:

>  I just checking sipxbridge so I unchecked MOH checkbox(in SBC) and
> verifying MOH is playing or not if calls via ITSP..
>
> Regards,
> Kumaran T
>
>
> On 11/16/2012 3:29 PM, Tony Graziano wrote:
>
> If the phone (UA) itself has MOH it is customary to still hear MOH in that
> use case, yes.
>
>  MOH can happen at several layers:
>
>  At the ITSP if it is provided (or enabled)
> At sipxbridge for trunking
> At the UA (i.e. Polycom)
>
>  Can you explain what it is you are trying to prove/unprove or test?
>
> On Fri, Nov 16, 2012 at 4:49 AM, Kumaran <
> thiru.venkateshwa...@ttplservices.com> wrote:
>
>> Hi Joegen,
>> Just now tried following scenario(Edit SBC->MOH->disabled)
>>  1.user 200 calls mobile no 00919986252763 via ITSP
>>  2.00919986252763 answers the call
>>  3.2way-call is established
>>  4.user 200 consult transfer to 203
>>  5.MOH will be heard to 00919986252763(So you mean this behavior
>> is not excepted)
>>
>> Regards,
>> Kumaran T
>>
>> On 11/16/2012 2:51 PM, Joegen Baclor wrote:
>> > Disabling MoH in sipX bridge means it wont initiate MoH for transfer.
>> > That wont stop phones from initiating it themselves.  This is expected.
>> >
>> > On 11/16/2012 02:39 PM, Kumaran wrote:
>> >> Hi All,
>> >>  I have disabled MOH support in sipXbridge but  external users
>> hears
>> >> MOH when user put on hold while call established via ITSP..
>> >>
>> >> Regards,
>> >> Kumaran T
>> >>
>> >> ___
>> >> sipx-dev mailing list
>> >> sipx-dev@list.sipfoundry.org
>> >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>> >>
>>
>>
>> _______
>> sipx-dev mailing list
>> sipx-dev@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>>
>
>
>
>  --
> ~~
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: tgrazi...@voice.myitdepartment.net
> Fax: 434.465.6833
> ~~
> Linked-In Profile:
> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
> Ask about our Internet Fax services!
> ~~
>
>  Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
> 2013!
>  <http://sipxcolab2013.eventbrite.com/?discount=tony2013>
>
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: helpdesk@voice.myitdepartment.**net
>
>  Helpdesk Customers: 
> http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
> Blog: http://blog.myitdepartment.net
>
>
> ___
> sipx-dev mailing listsipx-...@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>
>
>


-- 
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~

Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013!
<http://sipxcolab2013.eventbrite.com/?discount=tony2013>

-- 
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
___
sipx-dev mailing list
sipx-dev@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-dev/

Re: [sipx-dev] SipXbridge MOH

[Tony Graziano]

If the phone (UA) itself has MOH it is customary to still hear MOH in that
use case, yes.

MOH can happen at several layers:

At the ITSP if it is provided (or enabled)
At sipxbridge for trunking
At the UA (i.e. Polycom)

Can you explain what it is you are trying to prove/unprove or test?

On Fri, Nov 16, 2012 at 4:49 AM, Kumaran <
thiru.venkateshwa...@ttplservices.com> wrote:

> Hi Joegen,
> Just now tried following scenario(Edit SBC->MOH->disabled)
>  1.user 200 calls mobile no 00919986252763 via ITSP
>  2.00919986252763 answers the call
>  3.2way-call is established
>  4.user 200 consult transfer to 203
>  5.MOH will be heard to 00919986252763(So you mean this behavior
> is not excepted)
>
> Regards,
> Kumaran T
>
> On 11/16/2012 2:51 PM, Joegen Baclor wrote:
> > Disabling MoH in sipX bridge means it wont initiate MoH for transfer.
> > That wont stop phones from initiating it themselves.  This is expected.
> >
> > On 11/16/2012 02:39 PM, Kumaran wrote:
> >> Hi All,
> >>  I have disabled MOH support in sipXbridge but  external users hears
> >> MOH when user put on hold while call established via ITSP..
> >>
> >> Regards,
> >> Kumaran T
> >>
> >> ___
> >> sipx-dev mailing list
> >> sipx-dev@list.sipfoundry.org
> >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
> >>
>
>
> ___
> sipx-dev mailing list
> sipx-dev@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>



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Re: [sipx-dev] Query on Call Status for a call from PSTN phone to Mobile number

[Tony Graziano]

Why in the world are you testing with a Polycom phone using firmware 3.3.1
instead of 3.2.6 or 3.2.7 at this time?

On Fri, Nov 16, 2012 at 12:11 AM, sangeetha.prem <
sangeetha.p...@ttplservices.com> wrote:

> **
> Hi,
>
> Please find the pcap file attached.
>
> Thanks
> Sangeetha
>
> --
>
>
>  Original Message   Subject: Re: [sipx-dev] [Fwd: Re:
> Query on Call Status for a call from PSTN phone to Mobile number]  Date: Thu,
> 15 Nov 2012 19:21:39 +0800  From: Joegen Baclor 
>   Reply-To:
> sipXecs developer discussions 
>   To:
> sipXecs developer discussions 
>   References:
> <50a4ca75.5010...@qantom.com> <50a4ca75.5010...@qantom.com>
> 
>
> There is a chance that this is a race condition on call cancellation where
> CANCEL is being sent while 200 OK is already traversing the wire.  In this
> case, the call-termination mechanism will be a time-out in ACK.  However,
> CDR will show sucess because it base the connected flag on receipt of 200
> OK.  If this is the case, then it is not a bug.  Too many assumptions
> here.  a packet capture will shed light.
>
> On 11/15/2012 07:11 PM, Tony Graziano wrote:
>
> correct
>
> On Thu, Nov 15, 2012 at 5:56 AM, sangeetha.prem  > wrote:
>
>> Hi All,
>>
>> Thanks for the reply, so can i raise an issue since the call status is
>> displayed as "Completed" instead of "Abondoned" when the subscriber hung up
>> before there was an answer.
>>
>> Thanks
>> Sangeetha
>>
>>
>>
>> Tony Graziano wrote:
>>
>> if there was no answer
>>
>>  Subscriber --> PSTN
>>
>>  and the subscriber hung up before there was an answer, then the cdr
>> should say "abandoned".
>>
>> On Thu, Nov 15, 2012 at 5:30 AM, sangeetha.prem <
>> sangeetha.p...@qantom.com> wrote:
>>
>>> Hi All,
>>>
>>>Make a call from Phone 1 to Mobile number and do not answer the call
>>>  on Mobile or reject the call from mobile, In CDR historic page call
>>> status is displayed as "Completed" instead of "Abondoned" . What is the
>>> expected Call Status?
>>>
>>> Thanks
>>> Sangeetha
>>> ___
>>> sipx-dev mailing list
>>> sipx-dev@list.sipfoundry.org
>>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>>>
>>
>>
>>
>>  --
>> ~~
>> Tony Graziano, Manager
>> Telephone: 434.984.8430
>> sip: tgrazi...@voice.myitdepartment.net
>> Fax: 434.465.6833
>> ~~
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>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>> Ask about our Internet Fax services!
>> ~~
>>
>>  Using or developing for sipXecs from SIPFoundry? Ask me about
>> sipX-CoLab 2013!
>>  <http://sipxcolab2013.eventbrite.com/?discount=tony2013>
>>
>>
>> LAN/Telephony/Security and Control Systems Helpdesk:
>> Telephone: 434.984.8426
>> sip: helpd...@voice.myitdepartment.net
>>
>>  Helpdesk Customers: http://myhelp.myitdepartment.net
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>>
>> --
>>
>> ___
>> sipx-dev mailing listsipx-...@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>>
>>
>> ___
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>> sipx-dev@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>>
>
>
>
>  --
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> Telephone: 434.984.8430
> sip: tgrazi...@voice.myitdepartment.net
> Fax: 434.465.6833
> ~~
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>
>
&

Re: [sipx-users] NTP for phones

[Tony Graziano]

I don't see port 123 anywhere so I guess it is not being allowed.
On Nov 15, 2012 5:36 PM, "Kyle Haefner"  wrote:

> Hi All,
>
> Finally getting around to putting phones on my fresh install of openUC
> 4.6.  If I have the firewall disabled the phones get time from the sipx
> cluster.  If I have the firewall enabled then they do not.  I have tried
> setting the permit time synchronization and provide time settings under NTP
> to no avail.  It doesn't look like iptables is taking ntp into account, but
> maybe I'm missing something?
>
> Here is what I see fro iptables -L
>
> ACCEPT tcp  --  anywhere anywheretcp dpt:http
> state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp dpt:https
> state NEW,ESTABLISHED
> ACCEPT udp  --  anywhere anywhereudp
> dpt:domain state NEW,ESTABLISHED
> ACCEPT udp  --  anywhere anywhereudp
> dpts:irisa:12999 state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp dpt:ftp
> state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp
> dpt:ftp-data state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp
> dpts:5:50050 state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp
> dpt:xmpp-server state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp
> dpt:hpvirtgrp state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp
> dpt:xmpp-client state NEW,ESTABLISHED
> ACCEPT udp  --  anywhere anywhereudp
> dpts:3:31000 state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp
> dpts:fmtp:asterix state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp
> dpts:irdmi:xprint-server state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp dpt:8185
> state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp dpt:sip
> state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp
> dpt:sip-tls state NEW,ESTABLISHED
> ACCEPT udp  --  anywhere anywhereudp dpt:sip
> state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp
> dpt:ircu-2 state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp
> dpt:onscreen state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp
> dpt:sdl-ets state NEW,ESTABLISHED
> ACCEPT tcp  --  anywhere anywheretcp dpt:ssh
> state NEW,ESTABLISHED
> ACCEPT udp  --  anywhere anywhereudp dpt:tftp
> state NEW,ESTABLISHED
> ACCEPT all  --  anywhere anywherestate
> RELATED,ESTABLISHED
> ACCEPT icmp --  anywhere anywhere
> ACCEPT all  --  anywhere anywhere
>
>
> --
> Kyle Haefner, M.S.
> Communication Systems Programmer
> Colorado State University
> Fort Collins, CO
> Phone: 970-491-1012
> Email:  kyle.haef...@colostate.edu
>
> 01010010 01100101 0111 01101100 0010 01101101 01100101 01101110
> 0010 0111 01110010 0110 01100111 01110010 0111 01101101
> 0010 01101001 01101110 0010 01100010 01101001 01101110 0111
> 01110010 0001 00101110
>
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>

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Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

you really need to look at the mail log to see where the mail is coming
from regardless of your firewall settings. It can actually come from inside
you see.

On Thu, Nov 15, 2012 at 9:29 AM, Noah Mehl  wrote:

> I am seeing more spam in my mail queue.  I have iptables installed, and
> here are my rules:
>
> Chain INPUT (policy ACCEPT)
> target prot opt source   destination
> RH-Firewall-1-INPUT  all  --  anywhere anywhere
>
> Chain FORWARD (policy ACCEPT)
> target prot opt source   destination
> RH-Firewall-1-INPUT  all  --  anywhere anywhere
>
> Chain OUTPUT (policy ACCEPT)
> target prot opt source   destination
>
> Chain RH-Firewall-1-INPUT (2 references)
> target prot opt source   destination
> ACCEPT all  --  anywhere anywhere
> ACCEPT icmp --  anywhere anywhereicmp any
> ACCEPT esp  --  anywhere anywhere
> ACCEPT ah   --  anywhere anywhere
> ACCEPT udp  --  anywhere 224.0.0.251 udp dpt:mdns
> ACCEPT udp  --  anywhere anywhereudp dpt:ipp
> ACCEPT tcp  --  anywhere anywheretcp dpt:ipp
> ACCEPT all  --  anywhere anywherestate
> RELATED,ESTABLISHED
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:pcsync-https
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:http
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:xmpp-client
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:5223
> ACCEPT all  --  192.168.0.0/16   anywhere
> ACCEPT udp  --  anywhere anywherestate NEW udp
> dpt:sip
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:sip
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:sip-tls
> ACCEPT udp  --  sip02.gafachi.comanywherestate NEW
> udp dpts:sip:5080
> ACCEPT udp  --  204.11.192.0/22  anywherestate NEW
> udp dpts:sip:5080
> REJECT all  --  anywhere anywherereject-with
> icmp-host-prohibited
>
> As far as I can tell, no one should be able to use port 25 from the world.
>  Also, sendmail is only configured to allow relay from localhost:
>
> [root@sipx1 ~]# cat /etc/mail/access
> # Check the /usr/share/doc/sendmail/README.cf file for a description
> # of the format of this file. (search for access_db in that file)
> # The /usr/share/doc/sendmail/README.cf is part of the sendmail-doc
> # package.
> #
> # by default we allow relaying from localhost...
> Connect:localhost.localdomain   RELAY
> Connect:localhost   RELAY
> Connect:127.0.0.1   RELAY
>
> Can someone please help me figure out where this spam is coming from?
>  Thanks.
>
> ~Noah
>
> On Oct 13, 2012, at 10:17 AM, Noah Mehl  wrote:
>
> > I did not change the configuration of anything related to the PlcmSpIp
> user.  It does however make me feel better that it is related to the vsftpd
> service and the polycom phones.
> >
> >> From /etc/passwd:
> >
> >
> PlcmSpIp:x:500:500::/var/sipxdata/configserver/phone/profile/tftproot:/sbin/nologin
> >
> > So, that user cannot ssh to a shell. So I don't think it was that.
> >
> > ~Noah
> >
> > On Oct 12, 2012, at 9:05 AM, Tony Graziano 
> wrote:
> >
> >> ... more -- its a user that does not have login to the OS itself, just
> >> vsftpd, which is restricted to certain commands and must present a
> >> request for its mac address in order to get a configuration file. It
> >> is not logging into linux unless someone changed the rights of the
> >> user.
> >>
> >> On Fri, Oct 12, 2012 at 7:30 AM, George Niculae 
> wrote:
> >>> On Fri, Oct 12, 2012 at 2:26 PM, Tony Graziano
> >>>  wrote:
> >>>> this is not a valid system user unless you have manually added it to
> the
> >>>> system. I do think the logs would show more if access was granted.
> Why are
> >>>> you exposing sshd to the outside world with an acl or by protecting
> it at
> >>>> your firewall?
> >>>>
> >>>
> >>> PlcmSpIp is the user used by polycom phones for fetching config from
> server
> >>>
> >>> George
> >>> ___
> >>> sipx-users mailing list
> >>&g

Re: [sipx-dev] [Fwd: Re: Query on Call Status for a call from PSTN phone to Mobile number]

[Tony Graziano]

correct

On Thu, Nov 15, 2012 at 5:56 AM, sangeetha.prem
wrote:

> Hi All,
>
> Thanks for the reply, so can i raise an issue since the call status is
> displayed as "Completed" instead of "Abondoned" when the subscriber hung up
> before there was an answer.
>
> Thanks
> Sangeetha
>
> **
>
>
> Tony Graziano wrote:
>
> if there was no answer
>
>  Subscriber --> PSTN
>
>  and the subscriber hung up before there was an answer, then the cdr
> should say "abandoned".
>
> On Thu, Nov 15, 2012 at 5:30 AM, sangeetha.prem  > wrote:
>
>> Hi All,
>>
>>Make a call from Phone 1 to Mobile number and do not answer the call
>>  on Mobile or reject the call from mobile, In CDR historic page call
>> status is displayed as "Completed" instead of "Abondoned" . What is the
>> expected Call Status?
>>
>> Thanks
>> Sangeetha
>> ___
>> sipx-dev mailing list
>> sipx-dev@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>>
>
>
>
>  --
> ~~
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: tgrazi...@voice.myitdepartment.net
> Fax: 434.465.6833
> ~~
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> Ask about our Internet Fax services!
> ~~
>
>  Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
> 2013!
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>
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: helpdesk@voice.myitdepartment.**net
>
>  Helpdesk Customers: 
> http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
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>
> --
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>
>
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>



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Re: [sipx-dev] Query on Call Status for a call from PSTN phone to Mobile number

[Tony Graziano]

if there was no answer

Subscriber --> PSTN

and the subscriber hung up before there was an answer, then the cdr should
say "abandoned".

On Thu, Nov 15, 2012 at 5:30 AM, sangeetha.prem
wrote:

> Hi All,
>
>Make a call from Phone 1 to Mobile number and do not answer the call
> on Mobile or reject the call from mobile, In CDR historic page call
> status is displayed as "Completed" instead of "Abondoned" . What is the
> expected Call Status?
>
> Thanks
> Sangeetha
> ___
> sipx-dev mailing list
> sipx-dev@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>



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Re: [sipx-users] Subscribe forwarding 4.4.0-update #22

[Tony Graziano]

I also think subscribe should not be forwarded to gateways, I see this with
users who subscribe to presence in speed dial when the speeddial is a PSTN
number. I somehow think this is part of the same overall issue with
subscribe in this instance. I also think sipxconfig could remedy this by
qualifying the existence of the user on the system when presence is checked
though...

On Mon, Nov 12, 2012 at 11:19 AM, Kyle Haefner
wrote:

> Hi All,
>
> Did the above fix work for you?  I'm having a similar problem, but with
> NOTIFY's. SIPX is forwarding these on and it might be what is contributing
> to our gateways failing.
>
> At any rate, SUBSCRIBE and NOTIFY should not follow forwarding rules.
>
> Kyle
>
>
> On Fri, Nov 9, 2012 at 5:52 AM, George Niculae  wrote:
>
>>  On Fri, Nov 9, 2012 at 2:43 PM, Melcon Moraes  wrote:
>>
>>> Hi George,
>>>
>>>  I might be terribly wrong here but how this issue can be related to
>>> the itsp? I believe the scenario Elwin described has no itsp at all.
>>>
>>
>>  Well not quite, I might be terribly wrong :) Thanks for pointing this
>> out, what I suggested could be just a part of the big fix
>>
>>
>>>
>>>  It seems that this config change would allow only INVITE to be
>>> forwarded to itsp accounts, but wouldn't prevent SUBSCRIBE to be forwarded
>>> to all of userforward targets.
>>>
>>>  I'll try it and get back to you with the results.
>>>
>>>
>>  Yes, please, if you have an itsp :)
>>
>>  Thanks
>> George
>>
>
>
>
> --
> Kyle Haefner, M.S.
> Communication Systems Programmer
> Colorado State University
> Fort Collins, CO
> Phone: 970-491-1012
> Email:  kyle.haef...@colostate.edu
>
> 01010010 01100101 0111 01101100 0010 01101101 01100101 01101110
> 0010 0111 01110010 0110 01100111 01110010 0111 01101101
> 0010 01101001 01101110 0010 01100010 01101001 01101110 0111
> 01110010 0001 00101110
>
> ___
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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Re: [sipx-dev] [sipx-users] something weird with proxy(sipXtackLib) [high cpu]

[Tony Graziano]

does it make sense for us to try to build the proxy up to fix UA's that
might be considered a little more than misguided in the way they handle
transactions? I don't disagree with the concept of trying to fix it, I just
wonder if we head down a path of no-return by having to deal with poorly
written ua's...

On Tue, Nov 13, 2012 at 10:48 AM, Domenico Chierico <
domenico.chier...@sip2ser.it> wrote:

> Hi Joegen
>
> In more genereical way I've found that we have a problem with
> uncorrectly closed socket from UA, this can be seen with an unfinished
> sip stack that ends prematurely and with some softphone that crash or
> (like linphone) allow to change the transport protocol on fly.
>
> Using many different softphone make our server behave as I described,
> with this patch seems that things go better.
>
> I'm still testing so this aren't final results, what I really like to
> know is your opinion about the validity of the approach, basically I
> think that check if socket is broken before read or write on it seems
> to be more safe way of manage.
> Do you agree ?
>
>
> On Tue, Nov 13, 2012 at 4:09 PM, Joegen Baclor  wrote:
> > Domenico,
> >
> > Thanks for the patch.  Just clarifying, this patch is for the behavior
> you
> > specified in the August 3 post?  If I'm correct, All I need to do to
> > reproduce is send an INVITE using TCP, on receipt of 183, close the
> socket.
> >
> > -j
> >
> >
> > On 11/13/2012 10:53 PM, Domenico Chierico wrote:
> >
> > Just to simplify tests here is the patch
> >
> > On Tue, Nov 13, 2012 at 3:14 PM, Domenico Chierico
> >  wrote:
> >
> > Hi
> > We have 1 sipxecs 4.4 with 50 users installed on kvm based virtual
> machine.
> > We had the proxy that ran over 290% of cpu with an average cpu load
> > close to 95%. Applying the review #22, the stuff start goes better and
> > we are now close to 40% of cpu load.
> >
> > Some of this load come from the known SUBSCRIBE issue, but some others
> > come from a strange behaviour of the tcp part of the sip stack that we
> > found:
> >
> > - linphone client increases the load on sipXproxy, with his own
> > strange keepalive method ("Jak" msg to the proxy) and switching the
> > transport from tcp to udp.
> >
> > - Some other evidences come from my personal tests as I notify on 3 of
> > August on dev-ml.
> >
> > Now I'm testing a solution that seems to work, but I wish to know your
> > opinion. I've change the order of "if" statements into SipClient::run
> > and I moved the branch about POLLERR and POLLHUP as first.
> >
> > On Fri, Aug 3, 2012 at 11:43 AM, Domenico Chierico
> >  wrote:
> >
> > I'm just playing around with go(lang), and this days I was starting
> > with sip stack implementation, just when messages starts float around
> > I'd realize that I've written a DOS for proxy ..
> > I just send INVITE to the proxy than reads for 100 and 180 and so I
> > close the socket, at this point I got this into the logs forever:
> >
> > "2012-08-03T09:31:03.817653Z":43810:SIP:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"SipClient[SipClientTcp-30]::run
> > resPoll= 1 revents: fd[0]= 0 fd[1]= 1d"
> > "2012-08-03T09:31:03.817668Z":43811:KERNEL:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"OsSocket::isReadyToWrite
> > poll returned 1 in socket: 21 0x7f5eec002070"
> > "2012-08-03T09:31:03.817683Z":43812:SIP:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"SipClient[SipClientTcp-30]::run
> > resPoll= 1 revents: fd[0]= 0 fd[1]= 1d"
> > "2012-08-03T09:31:03.817698Z":43813:KERNEL:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"OsSocket::isReadyToWrite
> > poll returned 1 in socket: 21 0x7f5eec002070"
> > "2012-08-03T09:31:03.817714Z":43814:SIP:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"SipClient[SipClientTcp-30]::run
> > resPoll= 1 revents: fd[0]= 0 fd[1]= 1d"
> > "2012-08-03T09:31:03.817728Z":43815:KERNEL:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"OsSocket::isReadyToWrite
> > poll returned 1 in socket: 21 0x7f5eec002070"
> >
> > I hope this helps..
> >
> > bye
> > Domenico Chierico
> >
> >
> >
> > ___
> > sipx-users mailing list
> > sipx-us...@list.sipfoundry.org
> > List Archive: http

Re: [sipx-users] something weird with proxy(sipXtackLib) [high cpu]

[Tony Graziano]

does it make sense for us to try to build the proxy up to fix UA's that
might be considered a little more than misguided in the way they handle
transactions? I don't disagree with the concept of trying to fix it, I just
wonder if we head down a path of no-return by having to deal with poorly
written ua's...

On Tue, Nov 13, 2012 at 10:48 AM, Domenico Chierico <
domenico.chier...@sip2ser.it> wrote:

> Hi Joegen
>
> In more genereical way I've found that we have a problem with
> uncorrectly closed socket from UA, this can be seen with an unfinished
> sip stack that ends prematurely and with some softphone that crash or
> (like linphone) allow to change the transport protocol on fly.
>
> Using many different softphone make our server behave as I described,
> with this patch seems that things go better.
>
> I'm still testing so this aren't final results, what I really like to
> know is your opinion about the validity of the approach, basically I
> think that check if socket is broken before read or write on it seems
> to be more safe way of manage.
> Do you agree ?
>
>
> On Tue, Nov 13, 2012 at 4:09 PM, Joegen Baclor  wrote:
> > Domenico,
> >
> > Thanks for the patch.  Just clarifying, this patch is for the behavior
> you
> > specified in the August 3 post?  If I'm correct, All I need to do to
> > reproduce is send an INVITE using TCP, on receipt of 183, close the
> socket.
> >
> > -j
> >
> >
> > On 11/13/2012 10:53 PM, Domenico Chierico wrote:
> >
> > Just to simplify tests here is the patch
> >
> > On Tue, Nov 13, 2012 at 3:14 PM, Domenico Chierico
> >  wrote:
> >
> > Hi
> > We have 1 sipxecs 4.4 with 50 users installed on kvm based virtual
> machine.
> > We had the proxy that ran over 290% of cpu with an average cpu load
> > close to 95%. Applying the review #22, the stuff start goes better and
> > we are now close to 40% of cpu load.
> >
> > Some of this load come from the known SUBSCRIBE issue, but some others
> > come from a strange behaviour of the tcp part of the sip stack that we
> > found:
> >
> > - linphone client increases the load on sipXproxy, with his own
> > strange keepalive method ("Jak" msg to the proxy) and switching the
> > transport from tcp to udp.
> >
> > - Some other evidences come from my personal tests as I notify on 3 of
> > August on dev-ml.
> >
> > Now I'm testing a solution that seems to work, but I wish to know your
> > opinion. I've change the order of "if" statements into SipClient::run
> > and I moved the branch about POLLERR and POLLHUP as first.
> >
> > On Fri, Aug 3, 2012 at 11:43 AM, Domenico Chierico
> >  wrote:
> >
> > I'm just playing around with go(lang), and this days I was starting
> > with sip stack implementation, just when messages starts float around
> > I'd realize that I've written a DOS for proxy ..
> > I just send INVITE to the proxy than reads for 100 and 180 and so I
> > close the socket, at this point I got this into the logs forever:
> >
> > "2012-08-03T09:31:03.817653Z":43810:SIP:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"SipClient[SipClientTcp-30]::run
> > resPoll= 1 revents: fd[0]= 0 fd[1]= 1d"
> > "2012-08-03T09:31:03.817668Z":43811:KERNEL:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"OsSocket::isReadyToWrite
> > poll returned 1 in socket: 21 0x7f5eec002070"
> > "2012-08-03T09:31:03.817683Z":43812:SIP:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"SipClient[SipClientTcp-30]::run
> > resPoll= 1 revents: fd[0]= 0 fd[1]= 1d"
> > "2012-08-03T09:31:03.817698Z":43813:KERNEL:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"OsSocket::isReadyToWrite
> > poll returned 1 in socket: 21 0x7f5eec002070"
> > "2012-08-03T09:31:03.817714Z":43814:SIP:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"SipClient[SipClientTcp-30]::run
> > resPoll= 1 revents: fd[0]= 0 fd[1]= 1d"
> > "2012-08-03T09:31:03.817728Z":43815:KERNEL:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"OsSocket::isReadyToWrite
> > poll returned 1 in socket: 21 0x7f5eec002070"
> >
> > I hope this helps..
> >
> > bye
> > Domenico Chierico
> >
> >
> >
> > ___
> > sipx-users mailing list
> > sipx-users@list.sipfoundry.org
> > List Archive: http

Re: [sipx-dev] something weird with proxy(sipXtackLib) [high cpu]

[Tony Graziano]

I had a similar issue and found it was a UA misconfigured after a rebuild
with the wrong password.
On Nov 13, 2012 9:14 AM, "Domenico Chierico" 
wrote:

> Hi
> We have 1 sipxecs 4.4 with 50 users installed on kvm based virtual machine.
> We had the proxy that ran over 290% of cpu with an average cpu load
> close to 95%. Applying the review #22, the stuff start goes better and
> we are now close to 40% of cpu load.
>
> Some of this load come from the known SUBSCRIBE issue, but some others
> come from a strange behaviour of the tcp part of the sip stack that we
> found:
>
> - linphone client increases the load on sipXproxy, with his own
> strange keepalive method ("Jak" msg to the proxy) and switching the
> transport from tcp to udp.
>
> - Some other evidences come from my personal tests as I notify on 3 of
> August on dev-ml.
>
> Now I'm testing a solution that seems to work, but I wish to know your
> opinion. I've change the order of "if" statements into SipClient::run
> and I moved the branch about POLLERR and POLLHUP as first.
>
> On Fri, Aug 3, 2012 at 11:43 AM, Domenico Chierico
>  wrote:
> > I'm just playing around with go(lang), and this days I was starting
> > with sip stack implementation, just when messages starts float around
> > I'd realize that I've written a DOS for proxy ..
> > I just send INVITE to the proxy than reads for 100 and 180 and so I
> > close the socket, at this point I got this into the logs forever:
> >
> > "2012-08-03T09:31:03.817653Z":43810:SIP:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"SipClient[SipClientTcp-30]::run
> > resPoll= 1 revents: fd[0]= 0 fd[1]= 1d"
> > "2012-08-03T09:31:03.817668Z":43811:KERNEL:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"OsSocket::isReadyToWrite
> > poll returned 1 in socket: 21 0x7f5eec002070"
> > "2012-08-03T09:31:03.817683Z":43812:SIP:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"SipClient[SipClientTcp-30]::run
> > resPoll= 1 revents: fd[0]= 0 fd[1]= 1d"
> > "2012-08-03T09:31:03.817698Z":43813:KERNEL:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"OsSocket::isReadyToWrite
> > poll returned 1 in socket: 21 0x7f5eec002070"
> > "2012-08-03T09:31:03.817714Z":43814:SIP:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"SipClient[SipClientTcp-30]::run
> > resPoll= 1 revents: fd[0]= 0 fd[1]= 1d"
> > "2012-08-03T09:31:03.817728Z":43815:KERNEL:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"OsSocket::isReadyToWrite
> > poll returned 1 in socket: 21 0x7f5eec002070"
> >
> > I hope this helps..
> >
> > bye
> > Domenico Chierico
> ___
> sipx-dev mailing list
> sipx-dev@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>

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LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net

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Re: [sipx-users] Karoo Configs

[Tony Graziano]

It needs two interfaces. It can sit behind a firewall. Both interfaces can
be numbered differently (same man).
On Nov 12, 2012 5:56 PM, "Chris Rawlings"  wrote:

> i have been reading through the Karoo bridge setup cookbook.. wow
>
> so i was wondering if anyone had any sample configs that would work in
> this scenario
>
> IP Based Authentication SIP Trunk ITSP
> Unmanaged GW for SipX / eZuce
> allow for inbound and outbound dialing/routing of SIP URI's
> passthrough for phone registrations
>
>
> my other question about Karoo is does it need 2 Interfaces... one for WAN
> and one for LAN or can i do everything from one interface on port 5060
> behind a NAT firewall.
>
>
>
> --
>
> Thank You,
>
> Chris Rawlings
>
> BlueCloud Consultants – CEO
>
> Phone. 484-335-1444 x201
>
> SIP URI. sip:chris.rawli...@bluecloudconsultants.com
>
> XMPP / Jabber / Google Talk – chris.rawli...@bluecloudconsultants.com
>
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>

-- 
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
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Re: [sipx-users] FreeSWITCH for SIP Trunking

[Tony Graziano]

wheter it is valet or not is not very concerning to me, since FS will not
have to handle the assignments (if sipx can configure it via its gui).

My only point is that I did have a long discussion with the FS developers
regarding presence and that you could add it using the mod_fifo as an
example, it would have to be written and to keep it intact should be
contributed upstream to FS. While it all has to be invented, most of the
groundwork and examples are already there.

On Mon, Nov 12, 2012 at 11:44 AM, Chris Rawlings wrote:

> if we are to change the parking lot setup.. please keep the idea of how it
> works now similar
>
> park to an extension ... pickup from extension
>
> i hate the idea of changing evertyhign to the idea of ... park the call...
> then it announces what parking lot it is on... that does not work for alot
> of my current customers...
>
> currently we use EFK to park and unpark calls and it is very fluid and
> easy for customers to understand the current way
>
>
> On Mon, Nov 12, 2012 at 8:36 AM, Michael Picher  wrote:
>
>> If somebody can point us to how to make presence work with a park orbit
>> we could probably move to a FS based park orbit...
>>
>> I'd love nothing better...  :-)
>>
>>
>>
>> On Mon, Nov 12, 2012 at 1:06 AM, Tony Graziano <
>> tgrazi...@myitdepartment.net> wrote:
>>
>>> http://track.sipfoundry.org/browse/XX-8473
>>>
>>> Is a good description of what needs to happen on the parking aspect. At
>>> some point perhaps it can be replaced using a FS service or B2BUA.
>>> On Nov 11, 2012 5:45 PM, "Chris Rawlings"  wrote:
>>>
>>>> has anyone tried this and then tried to place a call on park more than
>>>> once...
>>>>
>>>> i am having the issue as follows
>>>>
>>>> i place a call put it on park .. pick the call up.. put it on park
>>>> again... call gets stuck
>>>>
>>>> i may have just not have set this up correctly per the wiki but i doubt
>>>> it as the wiki was very easy to follow... could someone please try and
>>>> reproduce this issue..
>>>>
>>>> i also was working on FreeSWITCH awhile back to do this exact same
>>>> thing and had the freeswitch consultants looking at this and this is
>>>> exactly the same issue i ran into with them and there was no fix.
>>>>
>>>> On Mon, Nov 5, 2012 at 12:31 PM, Josh Patten  wrote:
>>>>
>>>>> I ran into a dependency issue with js
>>>>>
>>>>> spidermonkey required a different version of js than what we provide.
>>>>>
>>>>>
>>>>> On Mon, Nov 5, 2012 at 10:33 AM, Douglas Hubler wrote:
>>>>>
>>>>>> Why do you say it's unlikely Michael and Josh?
>>>>>>
>>>>>> I went out of my way to integrate with standard fs package.  We do
>>>>>> ignore all the configuration files that get installed with fs.
>>>>>> On Nov 5, 2012 10:50 AM, "Michael Picher"  wrote:
>>>>>>
>>>>>>> Not likely.
>>>>>>>
>>>>>>> At some point we'll probably look to use RPMs now that they are
>>>>>>> available.
>>>>>>>
>>>>>>> You are of course free to try ;-)
>>>>>>>
>>>>>>> Mike
>>>>>>>
>>>>>>>
>>>>>>> On Mon, Nov 5, 2012 at 8:51 AM, Melcon Moraes wrote:
>>>>>>>
>>>>>>>> Can we simply drop in a new FS RPM package to replace the 1.0.7
>>>>>>>> version that comes with sipXecs and all the current working stuff will
>>>>>>>> continue to work?
>>>>>>>>
>>>>>>>>
>>>>>>>> On Mon, Nov 5, 2012 at 1:58 AM, Josh Patten wrote:
>>>>>>>>
>>>>>>>>> Yes, MoH, ring on transfer, etc. all works in my tests.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>  On Sun, Nov 4, 2012 at 6:00 PM, Tony Graziano <
>>>>>>>>> tgrazi...@myitdepartment.net> wrote:
>>>>>>>>>
>>>>>>>>>> Karoo Bridge also uses the FS libraries. What FS lacks is an
>>>>>>>>>> admin GUI, just like karoo bridge. Does MOH work in your tests too?
>