It will depend entirely upon the itsp or telco provider.
Some itsp's do not actually support hair pinned calls.
I ran into an instance recently where the outbound call (forward) was a
local call but we had to use a 10 digit number instead of 7 only for the
forward.
A siptrace would be helpful.
On Wed, Sep 19, 2012 at 5:17 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Life is good. A 64 kbps MP3 @ 16 kHz and a 32 kbps MP3 @ 8 kHz are a
quarter the size of their WAV equivalents with a quality high enough most
folks can't hear any compression artifacts at all. Works for me.
Glad it
Andrewe any news on the patch i sent you offlist?
On 09/13/2012 03:19 AM, andrewpit...@comcast.net wrote:
Joegen,
We recently upgraded a bunch of our servers to 4.4.0 update #18, and
while this issue doesn't occur as frequently, we are still
experiencing it.
I was going to try running a
Please disregard this. The behavior is correct (4.4 sends PDF by default,
UNLESS the call is not completed, in which case the attachment is a TIFF
file, which is inconsequential since the testing was incorrectly reviewed).
On Tue, Sep 18, 2012 at 1:54 PM, Tony Graziano
On Wed, Sep 19, 2012 at 3:04 AM, George Niculae geo...@ezuce.com wrote:
Glad it worked. If you feel we should expose these as settings in UI
please raise a JIRA and will schedule for next releases
Done. http://track.sipfoundry.org/browse/XTRN-1073
- Jeff
what kind of gateway/who is the telco?
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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Adtran TA908e. I manage it. CLEC (network) PRI behind it.
The user's extension is set to ring first for 4 seconds, then forward to a
10-digit PSTN number. The PRI on the gateway will accept 7 or 10-digit
local number and the gateway is set to pass through whatever it receives.
I can dial a 7
On Wed, Sep 19, 2012 at 4:43 PM, Jeff Pyle jp...@fidelityvoice.com wrote:
On Wed, Sep 19, 2012 at 3:04 AM, George Niculae geo...@ezuce.com wrote:
Glad it worked. If you feel we should expose these as settings in UI
please raise a JIRA and will schedule for next releases
Done.
Have you tried removing all the phones but one on that called user to ensure
its not a bad behaving endpoint?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jeff Pyle
Sent: Wednesday, September 19, 2012 7:16 AM
To: Discussion list for
Scenario is now one Polycom 550 registered to sipX, receiving a call from
the outside. No change in the gateway-to-sipX messaging (call still
answered by VM instead of forwarded). Gateway is .53.11, sipX is .45.46.
0.00 192.168.53.11 - 192.168.54.46 SIP/SDP Request: INVITE
On Wed, Sep 19, 2012 at 10:21 AM, Daniel Peinado Lopez
daniel.pein...@iant.de wrote:
Hello,
I want to try some Yealink Telephones in SipX 4.4. I have downloaded the
Plugin in https://github.com/siplabs/sipXyealink and I have it in my CentOS
system. The problem is that I don`t know exactly
It is more likely a provider issue. I have seen this also but in my case
the provided wanted 10 digits instead of 7 even though 7 was valid.
I'd call the telco and troubleshoot with them. They are getting the digits
but they need to tell you what is wrong with the digits. really. Call the
telco.
Hi Tony,
In the case of an internal caller, all is well. The called-user's call
forward definitions work as one would expect. SipX sends the 7 or 10 digit
INVITE to the gateway when it should, and the telco handles the call. I
happen to be using 10 at the moment.
In the case of an external
Hi Jeff,
Just to ensure I understand the scenario:
• Calling party originates via PRI
• Calling party is routed to called party.
• Called party is unavailable, rule is set to forward to
10 digit destination number via PRI
• Calling party actually arrives at Free Switch
Hi Mark,
Your understand is spot on. A4.11. One PRI to a Lucent 5ESS CO. It's a
lab setup so there is next to no traffic. Not that it's related to this
case but the 908e does handle REFERs correctly from sipX when necessary.
That got crossed off the list yesterday.
Back to the sipX box.
Hi Jeff,
Thanks for the reply.
My knee jerk reaction when I first saw your post is that you
may have had multiple PRIs across multiple TDM groups. I'm
glad you've dug into this a bit.
By default, inter B-channel transfer (hair pinning) is not
allowed via AOS switchboard. However, I was
On Thu, Sep 20, 2012 at 12:34 AM, Mark A. Smith masm...@bcslive.biz wrote:
Hi Jeff,
Thanks for the reply.
My knee jerk reaction when I first saw your post is that you
may have had multiple PRIs across multiple TDM groups. I'm
glad you've dug into this a bit.
By default, inter B-channel
On Thu, Sep 20, 2012 at 12:47 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Mark, thanks anyway. I appreciate the replies. We haven't gotten far
enough to try TBCT yet! I like you're thinking, though.
George, I think I'm onto something. I've recreated this scenario with two
local users,
This is a long standing issue and is all about 302 redirects not able to
grant permissions of the called number to the caller. On top of this,
branches will also be enforced if you have set one. The caller will not
inherent the branch where the callee is located.
On 09/20/2012 10:04 AM,
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