Analog extension call transfer to SipX via Patton

Discussion:

Philippe Laurent

2012-07-19 19:37:14 UTC

Yeah, I know, this is a SipX list, not a Patton list. I'm working with
their tech support, but we're not getting very far, and since the Patton
4112 is connected to a SipX box, and there a quite a few Patton fans out
there working with SipX, I wanted to make sure I wasn't missing something
obvious in the link.

Patton 4112: SIP registered to SipX extension, FXS port connected to
pharmacy IVR

Here goes:
1. Caller dials 8 from the auto attendant, transfers to the pharmacy IVR on
the Patton FXS port. Success.
2. If the caller decides they want to speak to someone while sitting on the
IVR on the Patton, they dial 2.
3. The FXS IVR then dials the 600 extension to ring the hunt group on SipX.
The sequence used by the FXS IVR to transfer the call is Hook-flash, then
dial 600, then terminate. This fails to transfer the call (no ringing), and
the call terminates.

Clearly something isn't done right, or I've missed a concept. Should the
FXS IVR should be dialing a different string set to transfer the call? The
vendor says they can define the transfer on their side to match
requirements.

Many thanks in advance for your time.

Tony Graziano

2012-07-19 20:06:36 UTC

Permalink

Can you shoot me a sip debug of the transaction?

Since it is a sipx ivr (version 4.4/latest, right?), both transactions are
"attended transfers" from the IVR (I am assuming).

If it were me, I would try to create a phantom user with no VM permissions
and have the forwarding on "all the time" to the intended recipient, then
add that phantom user as an option on the IVR and see if that works. I have
found that hunt groups have been geeting frustratingly difficult lately.
Post back the results.

Post by Philippe Laurent
Yeah, I know, this is a SipX list, not a Patton list. I'm working with
their tech support, but we're not getting very far, and since the Patton
4112 is connected to a SipX box, and there a quite a few Patton fans out
there working with SipX, I wanted to make sure I wasn't missing something
obvious in the link.
Patton 4112: SIP registered to SipX extension, FXS port connected to
pharmacy IVR
1. Caller dials 8 from the auto attendant, transfers to the pharmacy IVR
on the Patton FXS port. Success.
2. If the caller decides they want to speak to someone while sitting on
the IVR on the Patton, they dial 2.
3. The FXS IVR then dials the 600 extension to ring the hunt group on
SipX. The sequence used by the FXS IVR to transfer the call is Hook-flash,
then dial 600, then terminate. This fails to transfer the call (no
ringing), and the call terminates.
Clearly something isn't done right, or I've missed a concept. Should the
FXS IVR should be dialing a different string set to transfer the call? The
vendor says they can define the transfer on their side to match
requirements.
Many thanks in advance for your time.
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
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Telephone: 434.984.8430
sip: ***@voice.myitdepartment.net
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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~~~~~~~~~~~~~~~~~~

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Philippe Laurent

2012-07-19 20:18:53 UTC

Permalink

Just heard from Patton, who indicated that there was a trans-coding issue,
with one side speaking ulaw and the other alaw during the transfer. Dunno
why this is, since all devices (Patton, PBX, phones) are set to prioritize
ulaw over alaw. The Patton 4112 apparently does not support trans-coding.

So, I set the Patton to talk only ulaw, saved and reloaded (you know, for
good luck), and boom, it works. Not great, but it works. During the
transfer, I get music for about 2 seconds, and then silence. No ringing, no
anything, until a phone in the hunt group picks up. Ringing of continued
tunes would ease the mind of the person on the other end.

Patton debug is on the way, but won't have access to the client's site
until this evening to do any testing.

Philippe

Post by Tony Graziano
Can you shoot me a sip debug of the transaction?
Since it is a sipx ivr (version 4.4/latest, right?), both transactions are
"attended transfers" from the IVR (I am assuming).
If it were me, I would try to create a phantom user with no VM permissions
and have the forwarding on "all the time" to the intended recipient, then
add that phantom user as an option on the IVR and see if that works. I have
found that hunt groups have been geeting frustratingly difficult lately.
Post back the results.

--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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>
Telephone: 434.984.8426
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Tony Graziano

2012-07-19 20:28:37 UTC

Permalink

FYI - I don't think I should see packet loss.There is packet loss during
the DTMF. I am going out on a lomb here, but would suggest it is a call the
patton is discarding because it does not know how to re-route it. I have
seen this with call park return.

If this continues to be the case it would be an issue where the patton is
having a problem with the destination. "Why" we don't know. I would suspect
a dial plan entry in the SN to be able to convert or transform it in a way
that would alleviate that.

(i.e. why is the from "anonymous" where it should know where it is from).
The firmware on the patton is correct. Do you have a way to put a 3.2
firmware on the destination (v v x)?

Post by Philippe Laurent
Just heard from Patton, who indicated that there was a trans-coding issue,
with one side speaking ulaw and the other alaw during the transfer. Dunno
why this is, since all devices (Patton, PBX, phones) are set to prioritize
ulaw over alaw. The Patton 4112 apparently does not support trans-coding.
So, I set the Patton to talk only ulaw, saved and reloaded (you know, for
good luck), and boom, it works. Not great, but it works. During the
transfer, I get music for about 2 seconds, and then silence. No ringing, no
anything, until a phone in the hunt group picks up. Ringing of continued
tunes would ease the mind of the person on the other end.
Patton debug is on the way, but won't have access to the client's site
until this evening to do any testing.
Philippe
On Thu, Jul 19, 2012 at 4:06 PM, Tony Graziano <

Post by Tony Graziano
Can you shoot me a sip debug of the transaction?
Since it is a sipx ivr (version 4.4/latest, right?), both transactions
are "attended transfers" from the IVR (I am assuming).
If it were me, I would try to create a phantom user with no VM
permissions and have the forwarding on "all the time" to the intended
recipient, then add that phantom user as an option on the IVR and see if
that works. I have found that hunt groups have been geeting frustratingly
difficult lately. Post back the results.

_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: ***@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!
~~~~~~~~~~~~~~~~~~

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Philippe Laurent

2012-07-19 20:47:37 UTC

Permalink

The packet loss did seem awfully odd and repeatable.

I won't be on site at the client location until at least next week, so I
hesitate to make any firmware changes when I don't have the capacity to
reach out and hard reset, especially with the relatively new VVX500.

Don't see any 3.2 downgrades for the VVX. Looks like the 4.0x series is all
there is to choose. Next time I'm by that location, I could pop in an IP650
and see how it behaves.

Post by Tony Graziano
FYI - I don't think I should see packet loss.There is packet loss during
the DTMF. I am going out on a lomb here, but would suggest it is a call the
patton is discarding because it does not know how to re-route it. I have
seen this with call park return.
If this continues to be the case it would be an issue where the patton is
having a problem with the destination. "Why" we don't know. I would suspect
a dial plan entry in the SN to be able to convert or transform it in a way
that would alleviate that.
(i.e. why is the from "anonymous" where it should know where it is from).
The firmware on the patton is correct. Do you have a way to put a 3.2
firmware on the destination (v v x)?

Post by Philippe Laurent
Just heard from Patton, who indicated that there was a trans-coding
issue, with one side speaking ulaw and the other alaw during the transfer.
Dunno why this is, since all devices (Patton, PBX, phones) are set to
prioritize ulaw over alaw. The Patton 4112 apparently does not support
trans-coding.
So, I set the Patton to talk only ulaw, saved and reloaded (you know, for
good luck), and boom, it works. Not great, but it works. During the
transfer, I get music for about 2 seconds, and then silence. No ringing, no
anything, until a phone in the hunt group picks up. Ringing of continued
tunes would ease the mind of the person on the other end.
Patton debug is on the way, but won't have access to the client's site
until this evening to do any testing.
Philippe
On Thu, Jul 19, 2012 at 4:06 PM, Tony Graziano <

Post by Tony Graziano
Can you shoot me a sip debug of the transaction?
Since it is a sipx ivr (version 4.4/latest, right?), both transactions
are "attended transfers" from the IVR (I am assuming).
If it were me, I would try to create a phantom user with no VM
permissions and have the forwarding on "all the time" to the intended
recipient, then add that phantom user as an option on the IVR and see if
that works. I have found that hunt groups have been geeting frustratingly
difficult lately. Post back the results.

Post by Philippe Laurent
Yeah, I know, this is a SipX list, not a Patton list. I'm working with
their tech support, but we're not getting very far, and since the Patton
4112 is connected to a SipX box, and there a quite a few Patton fans out
there working with SipX, I wanted to make sure I wasn't missing something
obvious in the link.
Patton 4112: SIP registered to SipX extension, FXS port connected to
pharmacy IVR
1. Caller dials 8 from the auto attendant, transfers to the pharmacy
IVR on the Patton FXS port. Success.
2. If the caller decides they want to speak to someone while sitting on
the IVR on the Patton, they dial 2.
3. The FXS IVR then dials the 600 extension to ring the hunt group on
SipX. The sequence used by the FXS IVR to transfer the call is Hook-flash,
then dial 600, then terminate. This fails to transfer the call (no
ringing), and the call terminates.
Clearly something isn't done right, or I've missed a concept. Should
the FXS IVR should be dialing a different string set to transfer the call?
The vendor says they can define the transfer on their side to match
requirements.
Many thanks in advance for your time.
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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>
Telephone: 434.984.8426
Helpdesk Customers: http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
Blog: http://blog.myitdepartment.net
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Tony Graziano

2012-07-19 22:04:51 UTC

Permalink

Ah. Vvx500. I was think vvx1500

Post by Philippe Laurent
The packet loss did seem awfully odd and repeatable.
I won't be on site at the client location until at least next week, so I
hesitate to make any firmware changes when I don't have the capacity to
reach out and hard reset, especially with the relatively new VVX500.
Don't see any 3.2 downgrades for the VVX. Looks like the 4.0x series is
all there is to choose. Next time I'm by that location, I could pop in an
IP650 and see how it behaves.
On Thu, Jul 19, 2012 at 4:28 PM, Tony Graziano <

Post by Philippe Laurent
Just heard from Patton, who indicated that there was a trans-coding
issue, with one side speaking ulaw and the other alaw during the transfer.
Dunno why this is, since all devices (Patton, PBX, phones) are set to
prioritize ulaw over alaw. The Patton 4112 apparently does not support
trans-coding.
So, I set the Patton to talk only ulaw, saved and reloaded (you know,
for good luck), and boom, it works. Not great, but it works. During the
transfer, I get music for about 2 seconds, and then silence. No ringing, no
anything, until a phone in the hunt group picks up. Ringing of continued
tunes would ease the mind of the person on the other end.
Patton debug is on the way, but won't have access to the client's site
until this evening to do any testing.
Philippe
On Thu, Jul 19, 2012 at 4:06 PM, Tony Graziano <

Post by Tony Graziano
Can you shoot me a sip debug of the transaction?
Since it is a sipx ivr (version 4.4/latest, right?), both transactions
are "attended transfers" from the IVR (I am assuming).
If it were me, I would try to create a phantom user with no VM
permissions and have the forwarding on "all the time" to the intended
recipient, then add that phantom user as an option on the IVR and see if
that works. I have found that hunt groups have been geeting frustratingly
difficult lately. Post back the results.

Post by Philippe Laurent
Yeah, I know, this is a SipX list, not a Patton list. I'm working
with their tech support, but we're not getting very far, and since the
Patton 4112 is connected to a SipX box, and there a quite a few Patton fans
out there working with SipX, I wanted to make sure I wasn't missing
something obvious in the link.
Patton 4112: SIP registered to SipX extension, FXS port connected to
pharmacy IVR
1. Caller dials 8 from the auto attendant, transfers to the pharmacy
IVR on the Patton FXS port. Success.
2. If the caller decides they want to speak to someone while sitting
on the IVR on the Patton, they dial 2.
3. The FXS IVR then dials the 600 extension to ring the hunt group on
SipX. The sequence used by the FXS IVR to transfer the call is Hook-flash,
then dial 600, then terminate. This fails to transfer the call (no
ringing), and the call terminates.
Clearly something isn't done right, or I've missed a concept. Should
the FXS IVR should be dialing a different string set to transfer the call?
The vendor says they can define the transfer on their side to match
requirements.
Many thanks in advance for your time.
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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>
Telephone: 434.984.8426
Helpdesk Customers: http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
Blog: http://blog.myitdepartment.net
_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/

_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
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Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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>
Telephone: 434.984.8426
Helpdesk Customers: http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
Blog: http://blog.myitdepartment.net
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Todd Hodgen

2012-07-20 04:53:54 UTC

Permalink

Ensure you have 4.0.2x in the VVX500. I identified a bug in 4.0.1 that was
fixed in the 4.0.2 release. It was specific to Music on Hold.

From: sipx-users-***@list.sipfoundry.org
[mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Philippe
Laurent
Sent: Thursday, July 19, 2012 1:48 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Analog extension call transfer to SipX via Patton

The packet loss did seem awfully odd and repeatable.

I won't be on site at the client location until at least next week, so I
hesitate to make any firmware changes when I don't have the capacity to
reach out and hard reset, especially with the relatively new VVX500.

Don't see any 3.2 downgrades for the VVX. Looks like the 4.0x series is all
there is to choose. Next time I'm by that location, I could pop in an IP650
and see how it behaves.

On Thu, Jul 19, 2012 at 4:28 PM, Tony Graziano
<***@myitdepartment.net> wrote:

FYI - I don't think I should see packet loss.There is packet loss during the
DTMF. I am going out on a lomb here, but would suggest it is a call the
patton is discarding because it does not know how to re-route it. I have
seen this with call park return.

If this continues to be the case it would be an issue where the patton is
having a problem with the destination. "Why" we don't know. I would suspect
a dial plan entry in the SN to be able to convert or transform it in a way
that would alleviate that.

(i.e. why is the from "anonymous" where it should know where it is from).
The firmware on the patton is correct. Do you have a way to put a 3.2
firmware on the destination (v v x)?

On Thu, Jul 19, 2012 at 4:18 PM, Philippe Laurent <***@ideos.com> wrote:

Just heard from Patton, who indicated that there was a trans-coding issue,
with one side speaking ulaw and the other alaw during the transfer. Dunno
why this is, since all devices (Patton, PBX, phones) are set to prioritize
ulaw over alaw. The Patton 4112 apparently does not support trans-coding.

So, I set the Patton to talk only ulaw, saved and reloaded (you know, for
good luck), and boom, it works. Not great, but it works. During the
transfer, I get music for about 2 seconds, and then silence. No ringing, no
anything, until a phone in the hunt group picks up. Ringing of continued
tunes would ease the mind of the person on the other end.

Patton debug is on the way, but won't have access to the client's site until
this evening to do any testing.

Philippe

On Thu, Jul 19, 2012 at 4:06 PM, Tony Graziano
<***@myitdepartment.net> wrote:

Can you shoot me a sip debug of the transaction?

Since it is a sipx ivr (version 4.4/latest, right?), both transactions are
"attended transfers" from the IVR (I am assuming).

If it were me, I would try to create a phantom user with no VM permissions
and have the forwarding on "all the time" to the intended recipient, then
add that phantom user as an option on the IVR and see if that works. I have
found that hunt groups have been geeting frustratingly difficult lately.
Post back the results.

On Thu, Jul 19, 2012 at 3:37 PM, Philippe Laurent <***@ideos.com> wrote:

Yeah, I know, this is a SipX list, not a Patton list. I'm working with their
tech support, but we're not getting very far, and since the Patton 4112 is
connected to a SipX box, and there a quite a few Patton fans out there
working with SipX, I wanted to make sure I wasn't missing something obvious
in the link.

Patton 4112: SIP registered to SipX extension, FXS port connected to
pharmacy IVR

Here goes:

1. Caller dials 8 from the auto attendant, transfers to the pharmacy IVR on
the Patton FXS port. Success.

2. If the caller decides they want to speak to someone while sitting on the
IVR on the Patton, they dial 2.

3. The FXS IVR then dials the 600 extension to ring the hunt group on SipX.
The sequence used by the FXS IVR to transfer the call is Hook-flash, then
dial 600, then terminate. This fails to transfer the call (no ringing), and
the call terminates.

Clearly something isn't done right, or I've missed a concept. Should the FXS
IVR should be dialing a different string set to transfer the call? The
vendor says they can define the transfer on their side to match
requirements.

Many thanks in advance for your time.

_______________________________________________
sipx-users mailing list
sipx-***@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: ***@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
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<http://sipxcolab2013.eventbrite.com/?discount=tony2013>

<http://sipxcolab2013.eventbrite.com/?discount=tony2013>

<http://sipxcolab2013.eventbrite.com/?discount=tony2013>
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sip: ***@voice.myitdepartment.net
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<http://sipxcolab2013.eventbrite.com/?discount=tony2013>
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