Hi Todd,
I've done that originally, that is, use the sipxecs template (without making
the changes suggested in the wiki) for the SP492 to configure them. I only did
the changes suggested in the wiki because I encountered the mentioned problem.
I think there is a good possibility that Primus does not support INVITE without
SDP, and if so my setup will never work as Tony and Joegen stated. I'll ask
Primus that question.
In the mean time, I'll try to learn more on setting up pfSense, including
producing a pcap trace and producing a siptrace on sipxecs to aid me
identifying the root of the problem.
Thanks a bunch to all,
Henry Kwan
________________________________
From: Todd Hodgen <[email protected]>
To: 'Henry Kwan' <[email protected]>; 'Discussion list for users of sipXecs
software' <[email protected]>
Sent: Saturday, October 20, 2012 1:10:52 AM
Subject: RE: [sipx-users] External calls cannot be transferred to voice mail
(sipXecs 4.4.0)
Henry, Try allowing the existing sipxecs template configure the phone, without
making the changes in the wiki to the profiles.
I have a system with approximately 20 of those phones working perfectly with
the system managing the templates for the phones completely.
From:[email protected]
[mailto:[email protected]] On Behalf Of Henry Kwan
Sent: Friday, October 19, 2012 4:40 PM
To: Tony Graziano; Joegen Baclor
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] External calls cannot be transferred to voice mail
(sipXecs 4.4.0)
Hi Tony,
I really appreciate that you took the time to elaborate in detail below. I
shall follow-up and perform your suggestions when time permits. Please also
see my response below.
Best regards,
Henry Kwan
________________________________
From:Tony Graziano <[email protected]>
To: Joegen Baclor <[email protected]>
Cc: Discussion list for users of sipXecs software
<[email protected]>; Henry Kwan <[email protected]>
Sent: Friday, October 19, 2012 2:36:25 AM
Subject: Re: [sipx-users] External calls cannot be transferred to voice mail
(sipXecs 4.4.0)
Primus is also LINGO. Primus is a large aggregator and also runs a
residential service (Lingo). The Lingo service does not support invite
without sdp, unless the reinvite is to one of their services and
typically only from one of their ATA's.
>> OK, I'll ask Primus about this.
I think you would do well to ask them if they support this AND it is
very important to make sure the invite for the incoming call comes to
your server on port 5080.
>> I've confirmed with Primus that they could accept signalling on 5060 on
>> their side and sent signalling to us on 5080.
I don't think your issue is unusual and usually stems from one of 3
core misconfiguration types:
1. Incompatible ITSP - Does a transfer from the AA to a user work?
Does a call from a user to another user work? (both as inbound calls
via the trunk). Is the original invite coming on port 5080.
>> Did not test a transfer from the AA to a user, will try that.
>> Call from a user (internal phone) to another user through local dialing plan
>> (i.e. 9-...) worked.
>> I think the original invite must come on port 5080 as that was the port that
>> was forwarded. 5060 was not forwarded.
2. Does the phone ring? If so, how was it configured (manually of by
sipx)? Please tell me you didn't register the line manually using the
sipx ip address. DNS is VERY important for the refer to voicemail. IF
you registered by IP, make sure you add the IP as a domain alias, but
really you should NEVER register by IP and expect all things to work
well.
>> Yes, the phone rang. The phone, Linksys SPA942, was configured via the sipX
>> web pages. I then reset the phone and have the configuration downloaded to
>> the phone via TFTP (I think this is the mechanism).
>> I re-installed sipXecs 4.4 a number of times. Sometimes IP as a domain
>> alias would appear automatically. I've also manually done that. In any
>> event, that did not help.
>> I also run the tests on the configuration test page and everything passed,
>> including DNS checks. I've also downloaded Flight Test (I think that was
>> the name) and everything passed.
3. Firewall - Unless your firewall is doing 1:1 NAT, you need to make
sure it can do Manual AON (static port nat). With pfsense this is
easy, but YOU CANNOT create port forward rules until you do this for
SIPX becuase they will follow the original NAT type. I sent you a link
of how to do this earlier, its pretty straightforward. You should be
able to use the pcap tool in pfsense and have it listen on WAN port
5080 and do a capture and see if the ITSP sends the call in on the
right port. If not, it will never work right (no matter what version)
and you need to ask them if they support this.
>> I did not do 1:1 NAT as I was not sure how to do that properly. I've read
>> up on it now and will try that out in the future. For port forwarding, I
>> did do Manual AON with static port checked on pfSense. I also needed to
>> create rules to pass this traffic. Same thing was done for port range 30000
>> to 31000. With this setup on pfSense, I could call in from an external
>> phone but still could not transfer to voice mail when no one answered. It
>> behaved exactly the same as using other routers - fast busy when the attempt
>> of transfer was made.
>> I have not had time to follow the link that you sent me earlier but will
>> definitely read up on it.
Good luck!
On Fri, Oct 19, 2012 at 1:50 AM, Joegen Baclor <[email protected]> wrote:
> Transferring ITSP originated calls requires that your ITSP supports INVITE
> without SDP. Before barking on something on the system, check first if your
> ITSP supports this. If not, there is no way your ITSP will work with sipx
> initiated transfers.
>
>
>
> On 10/19/2012 01:19 PM, Tony Graziano wrote:
>>
>> On Fri, Oct 19, 2012 at 1:07 AM, Henry Kwan <[email protected]> wrote:
>>>
>>> My installation was right from the 4.4 ISO. I did try without updating
>>> at
>>> all but to no avail.
>>>
>>> My ITSP is Primus Canada.
>>>
>>> Well I have to admit that I am not knowledgeable in setting up pfSense.
>>> In
>>> fact I am not knowledgeable on how to produce a pcap or produce a
>>> siptrace
>>> as Tony suggested. Having said that, I'll continue to play with 4.4 and
>>> look into how to perform the tasks suggested when time permits.
>>>
>> Pfsense
>> http://blog.myitdepartment.net/?p=297
>>>
>>> In the mean time, 4.2.1 will have to suffice until I can figure out what
>>> I
>>> did wrong.
>>>
>>> By the way, my observation regarding the inconsistent behaviour on
>>> restarts
>>> for both 4.2.1 and 4.4.0 is not an attack on sipXecs. It is just that -
>>> an
>>> observation. Maybe someone can comment if this observation is also only
>>> experienced by me. If that's the case, I must be a jinx or have a unique
>>> ability to bring out the worst in sipXecs.
>>>
>> I can set up a new system each day and don't experience this behavior.
>> It's really important to observe how much RAM you have installed (I
>> prefer 8GB minimum but for very small installs 4GB on 4.4 is enough,
>> though 8GB should be the minimum for 4.6).
>>>
>>> Best regards to all,
>>>
>>> Henry Kwan
>>>
>>> ________________________________
>>> From: George Niculae <[email protected]>
>>>
>>> To: Discussion list for users of sipXecs software
>>> <[email protected]>
>>> Cc: Henry Kwan <[email protected]>
>>> Sent: Thursday, October 18, 2012 6:29:01 PM
>>>
>>> Subject: Re: [sipx-users] External calls cannot be transferred to voice
>>> mail
>>> (sipXecs 4.4.0)
>>>
>>> On Fri, Oct 19, 2012 at 3:26 AM, Tony Graziano
>>> <[email protected]> wrote:
>>>>
>>>> Rather than use an old unsupportable version, produce a pcap from your
>>>> firewall or produce a siptrace from sipx itself.
>>>>
>>>> I don't think your off the cuff observation is exactly right on targetm
>>>> .
>>>> Version 4.2 used its own media server while 4.4 uses FreeSWITCH and
>>>> there
>>>> are significant close changes.
>>>>
>>>> You could also indicate whether or not you followed a tutorial on how to
>>>> properly configure pfsense and who the itsp is.
>>>>
>>> Additionally, if you could try scenario with 4.4 built from ISO,
>>> without yum updating to latest, and report back, will help identifying
>>> if issue in latest patches
>>>
>>> Thanks
>>> George
>>>
>>>
>>>
>>> _______________________________________________
>>> sipx-users mailing list
>>> [email protected]
>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>>
>>
>
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Telephone: 434.984.8430
sip: [email protected]
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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