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Russell.Mosemann at cu...
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PostPosted: Thu Oct 01, 2009 12:00 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

We have connected FS to a Siemens Hicomm 300. As you might guess, it's
not working right. Here is what we are working with.

Dell 1750 (dual socket, dual core Xeon 2.8GHz)
Debian 5
FS (15029), OpenZAP (without libpri)
TE110P T1 card (Zaptel driver)
Handles 71xx extensions

Siemens Hicom 300
TMDN64P T1 card
Handles 74xx extensions

We are pretty much using the stock FS configuration, yet, because we're
trying to get this to work. I have configured OpenZAP and the associated
files like the examples on the wiki (see below) to work with a PRI T1.
There are 23 B channels and 1 D channel. The Zaptel side looks fine.
OpenZAP is able to open the channels when FS boots. So far, so good.

When a call is made from 74xx (Rolmphone 624) to 71xx (X-Lite 4.0 beta
from CounterPath on an office PC), X-Lite rings. The call can be
answered, and the conversation sounds fine. That means the routing,
registration and authorization are working on the network between X-Lite
and FS. It also means that FS is able to communicate with the Hicom over
the T1. Great.

When the caller presses the transfer button on the 74xx phone, the Hicom
sends a message over the D channel, and the call is disconnected
(watching with fs_cli). As best I can interpret the bytes in the message,
the Hicom sends a disconnect message when 74xx presses the transfer key.

In order to call 74xx, I created dialplan/default/02_hicom.xml. The
contents are

<include>
<extension name="hicom">
<condition field="destination_number" expression="^(74\d{2})$">
<action application="bridge" data="openzap/1/a/$1"/>
</condition>
</extension>
</include>

If a call is made from 71xx to 74xx, the Hicom forwards the call to the
switchboard with "7100->7445 connection not possible" (or whatever
extensions) in the switchboard display.

1. Are these issues related to the way I have configured FS?

The Hicom is maintained by the local phone company. I do not have access
to view or configure the T1 card on the Hicom. According to the phone
guy, there isn't anything else that needs to be configured on the Hicom.
He believes that if 74xx can call 71xx, then 71xx should be able to call
74xx.

I suspect that something more needs to be done on the Hicom in order to
accept calls from FS and bridge/transfer them to a local extension on the
Hicom. It's as if the Hicom doesn't know how or is not permitted to route
incoming calls on the T1 to local extensions. I have no way to know,
though. I'm hoping someone else has connected FS to a Hicom 300 and can
provide configuration details. If I could tell the phone guy something
like, "You need to look at <this>," that would help him out.

2. Should I receive CID/ANI from the Hicom?

X-Lite displays "OpenZAP" as the call and "1" as Other when the call
comes in, which is the information for the endpoint. Is there something I
need to do in the FS configuration to capture CID/ANI information from
the Hicom and make it available (or is it not being provided by the Hicom)?

3. When dialing from the Rolmphone is there a way for FS to send the
called name back to the Hicom for it to appear in the display?

When dialing 74xx to 74xx, of course, it shows the called number and name
in the display. We also have a HiPath 4000 connected to the Hicom 300.
When dialing an extension on the HiPath from the Hicom, the HiPath ships
the called name back to the Hicom for display on the phone. It would be
nice to do that from FS.

Let me know if you need additional information. Thanks for any pointers
or insight as to how things work.

--
Russell Mosemann


openzap.conf
[span zt PRI_1]
name => OpenZAP
number => 1
trunk_type => t1
b-channel => 1-23
d-channel => 24

zt.conf
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 64
rxgain => 0.0
txgain => 0.0

openzap.conf
<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="0"/>
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<!--<param name="enable-analog-option" value="call-swap"/>-->
<!--<param name="enable-analog-option" value="3-way"/>-->
</settings>
<pri_spans>
<span name="PRI_1">
<param name="q921loglevel" value="alert"/>
<param name="q931loglevel" value="alert"/>
<param name="mode" value="user"/>
<param name="dialect" value="national"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
</span>
</pri_spans>
</configuration>

zaptel.conf
# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER)
span=1,1,0,esf,b8zs
# termtype: te
bchan=1-23
dchan=24

# Global data
loadzone = us
defaultzone = us


________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


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anthony.minessale at g...
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PostPosted: Thu Oct 01, 2009 2:26 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

You might want to try the ozmod_pri instead of ozmod_isdn until the new revision of ozmod_isdn is published into the source tree.


On Thu, Oct 1, 2009 at 9:37 AM, <Russell.Mosemann@cune.org (Russell.Mosemann@cune.org)> wrote:
Quote:
We have connected FS to a Siemens Hicomm 300. As you might guess, it's
not working right. Here is what we are working with.

Dell 1750 (dual socket, dual core Xeon 2.8GHz)
Debian 5
FS (15029), OpenZAP (without libpri)
TE110P T1 card (Zaptel driver)
Handles 71xx extensions

Siemens Hicom 300
TMDN64P T1 card
Handles 74xx extensions

We are pretty much using the stock FS configuration, yet, because we're
trying to get this to work. I have configured OpenZAP and the associated
files like the examples on the wiki (see below) to work with a PRI T1.
There are 23 B channels and 1 D channel. The Zaptel side looks fine.
OpenZAP is able to open the channels when FS boots. So far, so good.

When a call is made from 74xx (Rolmphone 624) to 71xx (X-Lite 4.0 beta
from CounterPath on an office PC), X-Lite rings. The call can be
answered, and the conversation sounds fine. That means the routing,
registration and authorization are working on the network between X-Lite
and FS. It also means that FS is able to communicate with the Hicom over
the T1. Great.

When the caller presses the transfer button on the 74xx phone, the Hicom
sends a message over the D channel, and the call is disconnected
(watching with fs_cli). As best I can interpret the bytes in the message,
the Hicom sends a disconnect message when 74xx presses the transfer key.

In order to call 74xx, I created dialplan/default/02_hicom.xml. The
contents are

<include>
 <extension name="hicom">
   <condition field="destination_number" expression="^(74\d{2})$">
     <action application="bridge" data="openzap/1/a/$1"/>
   </condition>
 </extension>
</include>

If a call is made from 71xx to 74xx, the Hicom forwards the call to the
switchboard with "7100->7445 connection not possible" (or whatever
extensions) in the switchboard display.

1. Are these issues related to the way I have configured FS?

The Hicom is maintained by the local phone company. I do not have access
to view or configure the T1 card on the Hicom. According to the phone
guy, there isn't anything else that needs to be configured on the Hicom.
He believes that if 74xx can call 71xx, then 71xx should be able to call
74xx.

I suspect that something more needs to be done on the Hicom in order to
accept calls from FS and bridge/transfer them to a local extension on the
Hicom. It's as if the Hicom doesn't know how or is not permitted to route
incoming calls on the T1 to local extensions. I have no way to know,
though. I'm hoping someone else has connected FS to a Hicom 300 and can
provide configuration details. If I could tell the phone guy something
like, "You need to look at <this>," that would help him out.

2. Should I receive CID/ANI from the Hicom?

X-Lite displays "OpenZAP" as the call and "1" as Other when the call
comes in, which is the information for the endpoint. Is there something I
need to do in the FS configuration to capture CID/ANI information from
the Hicom and make it available (or is it not being provided by the Hicom)?

3. When dialing from the Rolmphone is there a way for FS to send the
called name back to the Hicom for it to appear in the display?

When dialing 74xx to 74xx, of course, it shows the called number and name
in the display. We also have a HiPath 4000 connected to the Hicom 300.
When dialing an extension on the HiPath from the Hicom, the HiPath ships
the called name back to the Hicom for display on the phone. It would be
nice to do that from FS.

Let me know if you need additional information. Thanks for any pointers
or insight as to how things work.

--
Russell Mosemann


openzap.conf
[span zt PRI_1]
name => OpenZAP
number => 1
trunk_type => t1
b-channel => 1-23
d-channel => 24

zt.conf
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 64
rxgain => 0.0
txgain => 0.0

openzap.conf
<configuration name="openzap.conf" description="OpenZAP Configuration">
 <settings>
   <param name="debug" value="0"/>
   <!--<param name="hold-music" value="$${moh_uri}"/>-->
   <!--<param name="enable-analog-option" value="call-swap"/>-->
   <!--<param name="enable-analog-option" value="3-way"/>-->
 </settings>
  <pri_spans>
    <span name="PRI_1">
      <param name="q921loglevel" value="alert"/>
      <param name="q931loglevel" value="alert"/>
      <param name="mode" value="user"/>
      <param name="dialect" value="national"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="public"/>
    </span>
  </pri_spans>
</configuration>

zaptel.conf
# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER)
span=1,1,0,esf,b8zs
# termtype: te
bchan=1-23
dchan=24

# Global data
loadzone        = us
defaultzone     = us


________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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Russell.Mosemann at cu...
Guest





PostPosted: Thu Oct 01, 2009 4:48 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

Anthony Minessale <anthony.minessale@gmail.com> said:

Quote:
You might want to try the ozmod_pri instead of ozmod_isdn until the new
revision of ozmod_isdn is published into the source tree.

libpri took care of the problem with the transfer. Now, someone can call
into FS from the Hicomm and then transfer the call to another extension
on the Hicomm.

A call from FS to the Hicomm still transfers to the switchboard. I'm not
seeing any CID/ANI on the X-Lite. It shows up as "OpenZAP" and "1". Is
there something like ngrep for the D channel of a PRI? It would be nice
to see what data is being sent between FS and the Hicom.

--
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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anthony.minessale at g...
Guest





PostPosted: Thu Oct 01, 2009 5:00 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

there was a feature to generate a pcap from the debug logs but i forgot who posted it.


On Thu, Oct 1, 2009 at 4:19 PM, <Russell.Mosemann@cune.org (Russell.Mosemann@cune.org)> wrote:
Quote:
Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> said:

Quote:
You might want to try the ozmod_pri instead of ozmod_isdn until the new
revision of ozmod_isdn is published into the source tree.


libpri took care of the problem with the transfer. Now, someone can call
into FS from the Hicomm and then transfer the call to another extension
on the Hicomm.

A call from FS to the Hicomm still transfers to the switchboard. I'm not
seeing any CID/ANI on the X-Lite. It shows up as "OpenZAP" and "1". Is
there something like ngrep for the D channel of a PRI? It would be nice
to see what data is being sent between FS and the Hicom.

--
Russell Mosemann




________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
Back to top
msc at freeswitch.org
Guest





PostPosted: Thu Oct 01, 2009 6:45 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

On Thu, Oct 1, 2009 at 2:19 PM, <Russell.Mosemann@cune.org (Russell.Mosemann@cune.org)> wrote:
Quote:
Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> said:

Quote:
You might want to try the ozmod_pri instead of ozmod_isdn until the new
revision of ozmod_isdn is published into the source tree.


libpri took care of the problem with the transfer. Now, someone can call
into FS from the Hicomm and then transfer the call to another extension
on the Hicomm.

A call from FS to the Hicomm still transfers to the switchboard. I'm not
seeing any CID/ANI on the X-Lite. It shows up as "OpenZAP" and "1". Is
there something like ngrep for the D channel of a PRI? It would be nice
to see what data is being sent between FS and the Hicom.

I believe the "OpenZAP" and "1" are coming from your conf file:
openzap.conf
[span zt PRI_1]
name => OpenZAP
number => 1



As far as debugging with ozmod_libpri I believe the syntax is:
oz libpri debug 1 all

It will do a traditional libpri-style debug, just like "pri debug span 1" in Asterisk.
-MC
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Russell.Mosemann at cu...
Guest





PostPosted: Fri Oct 02, 2009 6:58 am    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

Michael Collins <msc@freeswitch.org> said:

Quote:
Quote:
I believe the "OpenZAP" and "1" are coming from your conf file:
openzap.conf
[span zt PRI_1]
name => OpenZAP
number => 1

That is correct. If that information is removed, then X-Lite displays

FreeSWITCH
[Other: 0000000000]

Are there any variables to set to get CID, or is OpenZap supposed to be
filling that in?

Quote:
As far as debugging with ozmod_libpri I believe the syntax is:
oz libpri debug 1 all

That works. Now, I have to figure out what some of these abbreviations mean.

--
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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msc at freeswitch.org
Guest





PostPosted: Fri Oct 02, 2009 11:20 am    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

On Fri, Oct 2, 2009 at 4:48 AM, <Russell.Mosemann@cune.org (Russell.Mosemann@cune.org)> wrote:
Quote:
Michael Collins <msc@freeswitch.org (msc@freeswitch.org)> said:

Quote:
Quote:
I believe the "OpenZAP" and "1" are coming from your conf file:
openzap.conf
[span zt PRI_1]
name => OpenZAP
number => 1


That is correct. If that information is removed, then X-Lite displays

FreeSWITCH
[Other: 0000000000]

do something like:
name => XYZ Corp
number => 8005551212

Quote:

Are there any variables to set to get CID, or is OpenZap supposed to be
filling that in?

Quote:
As far as debugging with ozmod_libpri I believe the syntax is:
oz libpri debug 1 all


That works. Now, I have to figure out what some of these abbreviations mean.

Welcome to the wacky world of Q931. The wiki has info:
http://wiki.freeswitch.org/wiki/ISDN:_Integrated_Services_Digital_Network



-MC
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Russell.Mosemann at cu...
Guest





PostPosted: Fri Oct 02, 2009 12:34 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

Michael Collins <msc@freeswitch.org> said:

Quote:
do something like:
name => XYZ Corp
number => 8005551212

I was expecting that information to be filled with the caller name and
number. It doesn't really help if someone calls from outside the
business, and it looks like my business is calling me. Doesn't OpenZAP
extract caller information from a PRI T1?

--
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


_______________________________________________
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FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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msc at freeswitch.org
Guest





PostPosted: Fri Oct 02, 2009 1:30 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

On Fri, Oct 2, 2009 at 10:24 AM, <Russell.Mosemann@cune.org (Russell.Mosemann@cune.org)> wrote:
Quote:
Michael Collins <msc@freeswitch.org (msc@freeswitch.org)> said:


Quote:
do something like:
name => XYZ Corp
number => 8005551212


I was expecting that information to be filled with the caller name and
number. It doesn't really help if someone calls from outside the
business, and it looks like my business is calling me. Doesn't OpenZAP
extract caller information from a PRI T1?

Can you pastebin a dialplan snippet (or put it here) so I can see what you're doing?
-MC
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Russell.Mosemann at cu...
Guest





PostPosted: Fri Oct 02, 2009 2:03 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

Quote:
Can you pastebin a dialplan snippet (or put it here) so I can see what
you're doing?
-MC

It is the stock FS configuration with a small change. We're still testing things, getting them to work. This is from public.xml. It detects calls to internal 71xx extensions and transfers them. The transfer works. Do some additional variables need to be set here?

<extension name="public_extensions">
<condition field="destination_number" expression="^(10[01][0-9]|71\d{2})$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>

--
Russell Mosemann


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PostPosted: Fri Oct 02, 2009 2:45 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

On Fri, Oct 2, 2009 at 11:53 AM, Russell Mosemann <Russell.Mosemann@cune.org (Russell.Mosemann@cune.org)> wrote:
Quote:
> Can you pastebin a dialplan snippet (or put it here) so I can see what
Quote:
you're doing?
-MC


It is the stock FS configuration with a small change. We're still testing things, getting them to work. This is from public.xml. It detects calls to internal 71xx extensions and transfers them. The transfer works. Do some additional variables need to be set here?

<extension name="public_extensions">
 <condition field="destination_number" expression="^(10[01][0-9]|71\d{2})$">
   <action application="transfer" data="$1 XML default"/>
 </condition>
</extension>

cool. can you pastebin a debug log on an incoming call?
-MC
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Russell.Mosemann at cu...
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PostPosted: Fri Oct 02, 2009 5:10 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

Quote:
cool. can you pastebin a debug log on an incoming call?
-MC

Here you go.

http://pastebin.freeswitch.org/10570

One thing I notice is that in the second line, the caller number is missing.

2009-10-02 16:42:39.539736 [NOTICE] ozmod_libpri.c:772 -- Ring on channel 1:1 (from to 7100)

If libpri doesn't know the number, then it's probably not being sent by the Hicomm.

--
Russell Mosemann


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PostPosted: Fri Oct 02, 2009 7:10 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

On Fri, Oct 2, 2009 at 2:53 PM, Russell Mosemann <Russell.Mosemann@cune.org (Russell.Mosemann@cune.org)> wrote:
Quote:
> cool. can you pastebin a debug log on an incoming call?
Quote:
-MC


Here you go.

http://pastebin.freeswitch.org/10570

One thing I notice is that in the second line, the caller number is missing.

2009-10-02 16:42:39.539736 [NOTICE] ozmod_libpri.c:772 -- Ring on channel 1:1 (from  to 7100)

If libpri doesn't know the number, then it's probably not being sent by the Hicomm.

Exactly. Turn on q931 debugging and try again:


Quote:
oz libpri debug 1 all
PB the results again and we'll check it out.
-MC
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PostPosted: Fri Oct 02, 2009 8:04 pm    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

Quote:
Exactly. Turn on q931 debugging and try again:

oz libpri debug 1 all
PB the results again and we'll check it out.
-MC

Here's the next one. I'm not sure what to look for, but nothing pops out right away.

http://pastebin.freeswitch.org/10571

--
Russell Mosemann


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msc at freeswitch.org
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PostPosted: Mon Oct 05, 2009 11:22 am    Post subject: [Freeswitch-users] Connecting FS to Hicom 300 Reply with quote

On Fri, Oct 2, 2009 at 5:54 PM, Russell Mosemann <Russell.Mosemann@cune.org (Russell.Mosemann@cune.org)> wrote:
Quote:
> Exactly. Turn on q931 debugging and try again:
Quote:

oz libpri debug 1 all
PB the results again and we'll check it out.
-MC


Here's the next one. I'm not sure what to look for, but nothing pops out right away.

http://pastebin.freeswitch.org/10571


Confirmed: the Hicomm isn't sending anything at all in the SETUP message except the usual stuff: dialed number, channel number, etc. Does the Hicomm have any config parameters, like Caller ID presentation?

-MC
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