Re: [Freeswitch-users] Minimum audio length for uuid_record

[Dan Le]

Thanks, I think I found the thread you were referring to
("[Freeswitch-users] session record does not for very short calls"), which
doesn't seem to be a solution for my situation. However, I did find that
using session:recordFile() didn't delete the file if it was really short.
And following that thread lead me to an interesting channel variable that
could be useful to us, record_ms, but having trouble getting it to reflect
the audio length. I can see the variable when printing data from the info
application, but it's always 0. My snippet of code is very simple:
if session:ready() then
   session:recordFile("C:/Temp/recording.wav", 30, 600, 6);

   local record_length = session:getVariable("record_ms");
   freeswitch.consoleLog("INFO", "Recorded a " .. record_length .. " ms
file.\n");
end

Another side question, the silence secs parameter (in this example, 6), is
that 6s silence hits during the entire recording session or 6s of
consecutive silence? From a few tests, it seems to be the former, but just
wanted to verify, something that would make a good addition to the wiki.

Dan


On Wed, Sep 30, 2009 at 11:31 PM, Jason White  wrote:

> Dan Le  wrote:
> > We're running into a problem with the minimum file size when recording
> using
> > uuid_record. It seems if the audio is too short it deletes the audio
> file.
> > Is there a way to override that?
>
> Yes. It was discussed on the list recently. I suggest searching the list
> archives. Someone may have documented it on the wiki by now also.
>
>
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Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones

[João Mesquita]

Piece of advice, don't ask, just do it. ;)

jmesquita

On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards
wrote:

>  If you have time to take a look, I could put a trace in the pastebin?
>
> Jerry
>
>  --
> *From:* Jerry Richards [mailto:jerry.richa...@teotech.com]
> *Sent:* Thursday, October 01, 2009 10:29 AM
> *To:* 'freeswitch-users@lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
> ToSubscribing Phones
>
>  I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
>
> Thanks,
> Jerry
>
>  --
> *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com]
> *Sent:* Thursday, October 01, 2009 9:36 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
> ToSubscribing Phones
>
> which phone is it,
> we tested it with eyebeam and it appears to work for us.
>
>
> On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards  > wrote:
>
>>
>> By the way, I see the following lines at the FS console, which might be a
>> clue as to why this is happening.  Could someone point me toward what
>> might
>> cause this?  I set the "manage-presence" parameter to "true" in each XML
>> file where I saw it defined.
>>
>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>   ...
>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>   ...
>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>   ...
>> [WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping
>>
>>
>> Best Regards,
>> Jerry
>>
>>
>> -Original Message-
>> From: Jerry Richards [mailto:jerry.richa...@teotech.com]
>> Sent: Wednesday, September 30, 2009 9:12 AM
>> To: 'freeswitch-users@lists.freeswitch.org'
>> Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones
>>
>> I have two phones configured to subscribe to each other's presence status.
>> When I change the presence status in one phone, I see the SIP PUBLISH
>> message going to FS, but I don't see FS relaying that presence status to
>> the
>> subscribing phone.  Does anyone know why?
>>
>> Best Regards,
>> Jerry
>>
>>
>> ___
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>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_miness...@hotmail.com 
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
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>
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>
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Re: [Freeswitch-users] bgapi jobid to uuid

[Matt Riddell]

On 2/10/09 11:36 AM, Anthony Minessale wrote:
> if you make your own uuids you could set them in the originate string with
> {origination_uuuid=foo}
>
> where foo is a real uuid.
> if you have no other way to make them you can ask FS for one with the
> create_uuid api call.

Awesome, thanks - will give it a whirl over the weekend.

-- 
Cheers,

Matt Riddell
Director
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Re: [Freeswitch-users] bgapi jobid to uuid

[Michael Collins]

On Thu, Oct 1, 2009 at 2:36 PM, Matt Riddell  wrote:

> Hi,
>
> I decided to go with a linked list for current channels and maintain
> that through state changes.
>
> So, basically it works like this:
>
> 1. Originate a call using bgapi (we get a jobid in response)
> 2. Receive an event with the jobid and a uuid
> 3. Lookup the linked list for the jobid, set the uuid
> 4. Receive hangup etc (with uuid) remove from linked list
>
> The problem is that sometimes I receive the hangup with the uuid before
> I'm told what the match between the jobid and the uuid are.
>
> Any ideas?
>

If I may ask, what's the application? Are you working on Vicidial-ish stuff
for FreeSWITCH? Also, you might want to call the FreeSWITCH conference
tomorrow. We have some Q&A time with the FS devs and this kind of thing
might work better in realtime rather than email threads.
-MC
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Re: [Freeswitch-users] Connecting FS to Hicom 300

[Michael Collins]

On Thu, Oct 1, 2009 at 2:19 PM,  wrote:

> Anthony Minessale  said:
>
> > 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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Re: [Freeswitch-users] bgapi jobid to uuid

[Anthony Minessale]

if you make your own uuids you could set them in the originate string with
{origination_uuuid=foo}

where foo is a real uuid.
if you have no other way to make them you can ask FS for one with the
create_uuid api call.


On Thu, Oct 1, 2009 at 5:02 PM, Matt Riddell  wrote:

> On 2/10/09 10:50 AM, Anthony Minessale wrote:
> > if you pick your own job-uuid you can set it in the originate too
> >
> > send
> >
> > job-uuid: 1234
> >
> > in your bgapi event
> >
> > and
> >
> > {job_uuid=1234}sofia/internal/f...@bar.com 
> >
> > in your dial string.
> >
> > *shrug* i am not sure exactly what the goal is so maybe it's not a
> > useful suggestion...
>
> Does the job-uuid get used as the actual call uuid?  The problem is that
> the uuid for the call changes once it's answered (or hungup etc) and
> that the mapping doesn't come till after occasionally.
>
> I'm getting the job uuid no problem, but it's the replacement of the
> job-uuid with a call-uuid that is the issue.
>
> If this doesn't make too much sense I can post an example.
>
> --
> Cheers,
>
> Matt Riddell
> Director
> ___
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-- 
Anthony Minessale II

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Re: [Freeswitch-users] bgapi jobid to uuid

[Matt Riddell]

On 2/10/09 10:50 AM, Anthony Minessale wrote:
> if you pick your own job-uuid you can set it in the originate too
>
> send
>
> job-uuid: 1234
>
> in your bgapi event
>
> and
>
> {job_uuid=1234}sofia/internal/f...@bar.com 
>
> in your dial string.
>
> *shrug* i am not sure exactly what the goal is so maybe it's not a
> useful suggestion...

Does the job-uuid get used as the actual call uuid?  The problem is that 
the uuid for the call changes once it's answered (or hungup etc) and 
that the mapping doesn't come till after occasionally.

I'm getting the job uuid no problem, but it's the replacement of the 
job-uuid with a call-uuid that is the issue.

If this doesn't make too much sense I can post an example.

-- 
Cheers,

Matt Riddell
Director
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Re: [Freeswitch-users] Connecting FS to Hicom 300

[Anthony Minessale]

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,  wrote:

> Anthony Minessale  said:
>
> > 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!
>
>
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-- 
Anthony Minessale II

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Re: [Freeswitch-users] bgapi jobid to uuid

[Anthony Minessale]

if you pick your own job-uuid you can set it in the originate too

send

job-uuid: 1234

in your bgapi event

and

{job_uuid=1234}sofia/internal/f...@bar.com

in your dial string.

*shrug* i am not sure exactly what the goal is so maybe it's not a useful
suggestion...



On Thu, Oct 1, 2009 at 4:36 PM, Matt Riddell  wrote:

> Hi,
>
> I decided to go with a linked list for current channels and maintain
> that through state changes.
>
> So, basically it works like this:
>
> 1. Originate a call using bgapi (we get a jobid in response)
> 2. Receive an event with the jobid and a uuid
> 3. Lookup the linked list for the jobid, set the uuid
> 4. Receive hangup etc (with uuid) remove from linked list
>
> The problem is that sometimes I receive the hangup with the uuid before
> I'm told what the match between the jobid and the uuid are.
>
> Any ideas?
>
> --
> Cheers,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
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-- 
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Re: [Freeswitch-users] conference participant from behind NAT

[RobertT]

And here is a short piece of log from the server side:
...
nua(): refersh session after 62 seconds (in [55..65])...
send INVITE ...
rcv OK...
send ACK...
rcv BYE...

I see now that sdp for natted client has additional lines in OK response
compared to client with public ip.
Session-Expires: 120;refresher=uas
Min-SE: 120

How come that they differs? And how do I resolve this situation? Should
client handle these refresher messages normally?

Best regards, Robert.
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Re: [Freeswitch-users] Connecting FS to Hicom 300

[Russell.Mosemann]

Anthony Minessale  said:

> 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!


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[Freeswitch-users] bgapi jobid to uuid

[Matt Riddell]

Hi,

I decided to go with a linked list for current channels and maintain 
that through state changes.

So, basically it works like this:

1. Originate a call using bgapi (we get a jobid in response)
2. Receive an event with the jobid and a uuid
3. Lookup the linked list for the jobid, set the uuid
4. Receive hangup etc (with uuid) remove from linked list

The problem is that sometimes I receive the hangup with the uuid before 
I'm told what the match between the jobid and the uuid are.

Any ideas?

-- 
Cheers,

Matt Riddell
Director
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[Freeswitch-users] register problem

[Frank Carmickle]

Can someone point out what is wrong here.  Thanks.

Siptrace at http://carmickle.com/fs.txt


  











  

 
--FC

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Re: [Freeswitch-users] conference participant from behind NAT

[RobertT]

I am still experiencing problem with lost media in conference on a client
behind NAT.
This is what I've done - disabled VAD on a NATed client and asked my friend
to produce lots of animal sounds in order to keep channel busy. But at the
end of minute sounds of wild nature disapeared again. We reproduced that
without security with tcp SIP transport and got the same result.
Then I started to dig into SIP trace and this is what I found.
This client (behind NAT) recieve subsequent INVITE message from FS which
seem to destroy dialog and causes client app to close media stream after a
session being established normally. I performed the same call from box with
public ip and saw no subsequent INVITE's from FS. How come FS sends an
INVITE message to already connected client? Is it OK? Should client handle
this normally?

Below is client's SIP trace:

INVITE sip:1.conference...@74.208.167.44:5081;transport=TLS SIP/2.0
...
User-Agent: DoxWox SIP user agent
..

SIP/2.0 100 Trying
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 407 Proxy Authentication Required
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

ACK sip:1.conference...@74.208.167.44:5081;transport=TLS SIP/2.0
..

SIP/2.0 100 Trying
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 183 Session Progress
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 200 OK
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

ACK sip:1.conference...@74.208.167.44:5081;transport=tls SIP/2.0
..

*Finally, this message cause media stream closing*
INVITE sip:1...@87.184.52.45:64183;transport=tls SIP/2.0
Via: SIP/2.0/TLS 74.208.167.44:5081;branch=z9hG4bK8269NDyXQNjyH
Max-Forwards: 70
From: 
>;tag=vQH234QtN2U8Q

To: >;tag=3a231ba86c894ceca81d5021b68d3b6c

Call-ID: 37edc38329f64fe98c36cc0a6ddcbd9d
CSeq: 121093810 INVITE
Contact: 
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Session-Expires: 120;refresher=uac
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 340
v=0
o=FreeSWITCH 1254396503 1254396504 IN IP4 74.208.167.44
s=FreeSWITCH
c=IN IP4 74.208.167.44
t=0 0
m=audio 27726 RTP/SAVP 103 101
a=rtpmap:103 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:J6x2UgRVUY8GfbwjCuyyttrtnXnwwuWA9Pt+o3VW
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Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

[Peder]

Looking thru the example, it looks like each box has a real address of 21,
22 or 23 and they all have a loopback of .17, right?  So even though they
connections are being load balanced, each box really thinks it is .17 and
each client that connects thinks it is connecting to .17, right?  If that’s
the case, how does a client on one box call a client on the other box?
Since every box thinks it is .17 how would you bridge to another user on
another box that also thinks it is .17?  Or am I totally missing how it
works?

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Thursday, October 01, 2009 2:14 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

 

can we do it without advertising to use ubuntu =D
We don't like encouraging our users to use bleeding edge OS for our own
sanity with debugging.
Not to say you are not allowed to I just don't want to encourage it =p




On Thu, Oct 1, 2009 at 1:12 PM, Even André Fiskvik  wrote:

That's very cool Mike!

 

I'm going to try to configure four boxes with this as well (Btw, did you use
physical hardware or virtualization?)

and see how it goes. I followed Daniel Aliaman's blog as well, but I can try
it again with the tips

you provided on FreeSWITCH config to see if I can get it working properly
this time.

We did the setup on CentOS, but I wouldn't think that would be any issue.

 

Perhaps you or we could write up a complete guide about this on the wiki
since this is an scenario

commonly used? Also it would be great if we could outline possible issues
(and even better solutions)

to this kind of setup with regards to stuff like conferencing, bridging
between registered users and presence.

 

 

Best regards,

Even André

 

 

On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:

 

Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
heartbeat and ldirectord for load-balancing, fail-over and high
availability! I'm probably not the first one to do it, but as near as Google
and I can tell, I'm the first one to write about it.

 

Here's how you can duplicate my setup:

 

1. Install Ubuntu Server 8 on four machines, either real or VM.

2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
following these instructions:
http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start

3. Configure both FreeSWITCH boxes, and make sure they are both working.

4. Follow (most of) these instructions from Daniel Aliaman's blog. They were
written for Asterisk, but since a SIP connection is a SIP connection, most
of the document applies to FreeSWITCH:

http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf

 

The one problem I ran into was the IP address and port to which FreeSWITCH
was bound. The default is to use the primary address, which works great
out-of-the-box for everything else. When a client tried to register, all it
got back was an ICMP error -- Destination Unreachable, Port Unreachable.
That error is returned when no sockets are listening for UDP packets. To get
FreeSWITCH to listen for your Virtual IP, you need to set it in two places:

 

5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".

6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".

 

That should do it! If you have any success, please report to this list.

 

Keep in mind that if you want to do something like conferencing between two
registered clients, then you have to deal with the fact that the clients may
or may not be on the same box.

 

Mike van Lammeren

 

 

 

On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren 
wrote:

 

On Mon, Sep 28, 2009 at 9:05 PM, "Even André Fiskvik" 
wrote:

From: "Even André Fiskvik" 
To: freeswitch-users@lists.freeswitch.org
Date: Mon, 28 Sep 2009 22:52:13 +0200
Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

I have been working with a similar setup myself, but for some reason I ended
up ditching the

UltraMonkey setup because I just couldn't get it to work right.

 

It's been quite a while since my effort, so I don't remember what the exact
issue was.

I got registrations to work, but had some other sip-dialog issues.

 

We have since then changed over to running OpenSIPs as a loadbalancer in
front of 

multiple FreeSWITCH instances. This setup is still in testing, but seemlingy
works fine

(and if it doesn't, it's my own fault for writing a bad opensips config).

 

After we have done some more testing I can create a wiki-page with config
details.

 

 

Best regards,

Even André

 

 

Thanks, Even, that would be great! I might have to give up on the
ultramonkey solution, since I can't find anyone who has made it work. It's
too bad, because it would fit well with the rest of our architecture.

 

Mike van Lammeren

 

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Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

[Hadley Rich]

On Thu, 2009-10-01 at 14:14 -0500, Anthony Minessale wrote:
> can we do it without advertising to use ubuntu =D
> We don't like encouraging our users to use bleeding edge OS for our
> own sanity with debugging. 

I understand your stance, though if we're talking about Ubuntu 8.04 LTS
(Long Term Support - 5 years) it's not really bleeding edge anymore. 18
months ago when it was released it may have been a little, but the LTS
releases still aren't as bleeding edge as the standard support in
between releases.

hads
-- 
http://nicegear.co.nz
New Zealand's Open Source Hardware Supplier


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Re: [Freeswitch-users] Problem with gateway registration

[Nicolas Brenner]

Any ideas about this?

The SIP provider is offering H323, but I'm not quite sure about that, is
mod_opal working right?

Thanks!

Nicolas

On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner wrote:

> Anthony, thanks. Below are my config files for the two gateways from the
> sip trace. Both files are located in conf/directory/default.
>
> -
>
> redvoiss.xml (the one that works)
>
> 
>   
> 
>   
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
>   
> 
> 
>   
> 
>   
> 
>
> -
>
> orange.xml (the one that doesn't work)
>
> 
>   
> 
>   
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
>   
> 
> 
>   
> 
>   
> 
>
> -
>
> If I remove the register=true param for the non-working gateway, I don't
> get the registration error on the cli, but then all call attempts get
> rejected with a 401 Unauthorized, and I get a hangup cause of
> NORMAL_UNSPECIFIED.
>
>
> Best,
>
> Nicolas
>
>
>
> On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale <
> anthony.miness...@gmail.com> wrote:
>
>> 900 level errors are sofia internal errors so probably something is wrong
>> with your gateway config xml.
>> if you want to send it with any critical info replaced with XXX maybe we
>> can see the issue for you.
>>
>>
>>
>> On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner 
>> wrote:
>>
>>> Hello everyone,
>>>
>>> I am trying to add a gateway, but after configuring it just like the
>>> others gateways I have, it is failing to register with a message like this:
>>>
>>> 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration
>>> Failed with status Operation has no matching challenge  [904]. failure #1
>>> 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed
>>> Registration, setting retry to 10 seconds.
>>>
>>> I captured the sip traffic and noticed that when trying to register with
>>> one gateway (the one that works), I get a "Trying" reply immediately
>>> followed by a "401 Unauthorized" which contains a "WWW-Authenticate: digest"
>>> with a "qop=auth" parameter. Then Freeswitch replies with a second REGISTER
>>> including a large "Authorization: digest" section with cnonce and
>>> nc=0001 parameters.
>>>
>>> The gateway which doesn't register, doesn't send the "qop=auth" parameter
>>> together with the "401 Unauthorized", and then Freeswitch sends a
>>> "Authorization: digest" section on the second REGISTER with no cnonce or nc
>>> parameters.
>>>
>>> I know very little abouth SIP, so I'm wondering what this "qop=auth"
>>> parameter means and how does it affect the registration process. Is there
>>> any way to do without the qop=auth parameter?
>>>
>>> Also, I tried registering with X-Lite directly to the gateway, and it
>>> worked, so it appears to be a problem in the Freeswitch/gateway combination.
>>> (Note: X-Lite sends an "Authorization: digest" section on the _first_
>>> REGISTER, apparently this makes a difference)
>>>
>>> Attached is a sip trace for the registration traffic when doing "sofia
>>> profile external restart reloadxml" on the cli, captured with "tshark -i
>>> eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/capture.pcap -b filesize:51200 -b
>>> files:100 -R 'sip or rtp or icmp or dns or rtcp or t38'"
>>>
>>> Thanks!
>>>
>>> Nicolas
>>>
>>>
>>>
>>>
>>> ___
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>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_miness...@hotmail.com 
>> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:8...@conference.freeswitch.org 
>> iax:gu...@conference.freeswitch.org/888
>> googletalk:conf+...@conference.freeswitch.org
>> pstn:213-799-1400
>>
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>
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Re: [Freeswitch-users] Connecting FS to Hicom 300

[Anthony Minessale]

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,  wrote:

> 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
>
> 
>  
>
>  
>
>  
> 
>
> 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 ," 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
> 
>  
>
>
>
>
>  
>   
> 
>   
>   
>   
>   
>   
>   
> 
>   
> 
>
> 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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>



-- 
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ClueCon http://www.cluecon.com/
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Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

[Anthony Minessale]

can we do it without advertising to use ubuntu =D
We don't like encouraging our users to use bleeding edge OS for our own
sanity with debugging.
Not to say you are not allowed to I just don't want to encourage it =p



On Thu, Oct 1, 2009 at 1:12 PM, Even André Fiskvik  wrote:

> That's very cool Mike!
> I'm going to try to configure four boxes with this as well (Btw, did you
> use physical hardware or virtualization?)
> and see how it goes. I followed Daniel Aliaman's blog as well, but I can
> try it again with the tips
> you provided on FreeSWITCH config to see if I can get it working properly
> this time.
> We did the setup on CentOS, but I wouldn't think that would be any issue.
>
> Perhaps you or we could write up a complete guide about this on the wiki
> since this is an scenario
> commonly used? Also it would be great if we could outline possible issues
> (and even better solutions)
> to this kind of setup with regards to stuff like conferencing, bridging
> between registered users and presence.
>
>
> Best regards,
> Even André
>
>
> On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:
>
> Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
> heartbeat and ldirectord for load-balancing, fail-over and high
> availability! I'm probably not the first one to do it, but as near as Google
> and I can tell, I'm the first one to write about it.
> Here's how you can duplicate my setup:
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
> 2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
> following these instructions:
> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
>  3. Configure both
> FreeSWITCH boxes, and make sure they are both working.
> 4. Follow (most of) these instructions from Daniel Aliaman's blog. They
> were written for Asterisk, but since a SIP connection is a SIP connection,
> most of the document applies to FreeSWITCH:
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
> The one
> problem I ran into was the IP address and port to which FreeSWITCH was
> bound. The default is to use the primary address, which works great
> out-of-the-box for everything else. When a client tried to register, all it
> got back was an ICMP error -- Destination Unreachable, Port Unreachable.
> That error is returned when no sockets are listening for UDP packets. To get
> FreeSWITCH to listen for your Virtual IP, you need to set it in two places:
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
> That should do it! If you have any success, please report to this list.
>
> Keep in mind that if you want to do something like conferencing between two
> registered clients, then you have to deal with the fact that the clients may
> or may not be on the same box.
>
> Mike van Lammeren
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren  > wrote:
>
>>
>> On Mon, Sep 28, 2009 at 9:05 PM, "Even André Fiskvik" > > wrote:
>>
>>> From: "Even André Fiskvik" 
>>> To: freeswitch-users@lists.freeswitch.org
>>> Date: Mon, 28 Sep 2009 22:52:13 +0200
>>> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>>> I have been working with a similar setup myself, but for some reason I
>>> ended up ditching theUltraMonkey setup because I just couldn't get it to
>>> work right.
>>>
>>> It's been quite a while since my effort, so I don't remember what the
>>> exact issue was.
>>> I got registrations to work, but had some other sip-dialog issues.
>>>
>>> We have since then changed over to running OpenSIPs as a loadbalancer in
>>> front of
>>> multiple FreeSWITCH instances. This setup is still in testing, but
>>> seemlingy works fine
>>> (and if it doesn't, it's my own fault for writing a bad opensips config).
>>>
>>> After we have done some more testing I can create a wiki-page with config
>>> details.
>>>
>>>
>>> Best regards,
>>> Even André
>>>
>>>
>> Thanks, Even, that would be great! I might have to give up on the
>> ultramonkey solution, since I can't find anyone who has made it work. It's
>> too bad, because it would fit well with the rest of our architecture.
>>
>> Mike van Lammeren
>>
>
> ___
> 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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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
Cl

Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones

[Jerry Richards]

If you have time to take a look, I could put a trace in the pastebin?
 
Jerry


  _  

From: Jerry Richards [mailto:jerry.richa...@teotech.com] 
Sent: Thursday, October 01, 2009 10:29 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _  

From: Anthony Minessale [mailto:anthony.miness...@gmail.com] 
Sent: Thursday, October 01, 2009 9:36 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards 
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com
 
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
 
IRC: irc.freenode.net #freeswitch

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Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

[Even André Fiskvik]

That's very cool Mike!

I'm going to try to configure four boxes with this as well (Btw, did  
you use physical hardware or virtualization?)
and see how it goes. I followed Daniel Aliaman's blog as well, but I  
can try it again with the tips
you provided on FreeSWITCH config to see if I can get it working  
properly this time.
We did the setup on CentOS, but I wouldn't think that would be any  
issue.


Perhaps you or we could write up a complete guide about this on the  
wiki since this is an scenario
commonly used? Also it would be great if we could outline possible  
issues (and even better solutions)
to this kind of setup with regards to stuff like conferencing,  
bridging between registered users and presence.



Best regards,
Even André


On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:

Guess what? I have two FreeSWITCH servers working behind  
UltraMonkey, using heartbeat and ldirectord for load-balancing, fail- 
over and high availability! I'm probably not the first one to do it,  
but as near as Google and I can tell, I'm the first one to write  
about it.


Here's how you can duplicate my setup:

1. Install Ubuntu Server 8 on four machines, either real or VM.
2. Compile and install FreeSWITCH v1.0.4 from source on two  
machines, following these instructions: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
3. Configure both FreeSWITCH boxes, and make sure they are both  
working.
4. Follow (most of) these instructions from Daniel Aliaman's blog.  
They were written for Asterisk, but since a SIP connection is a SIP  
connection, most of the document applies to FreeSWITCH:

http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf

The one problem I ran into was the IP address and port to which  
FreeSWITCH was bound. The default is to use the primary address,  
which works great out-of-the-box for everything else. When a client  
tried to register, all it got back was an ICMP error -- Destination  
Unreachable, Port Unreachable. That error is returned when no  
sockets are listening for UDP packets. To get FreeSWITCH to listen  
for your Virtual IP, you need to set it in two places:


5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".

That should do it! If you have any success, please report to this  
list.


Keep in mind that if you want to do something like conferencing  
between two registered clients, then you have to deal with the fact  
that the clients may or may not be on the same box.


Mike van Lammeren



On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren > wrote:


On Mon, Sep 28, 2009 at 9:05 PM, "Even André Fiskvik"  
 wrote:

From: "Even André Fiskvik" 
To: freeswitch-users@lists.freeswitch.org
Date: Mon, 28 Sep 2009 22:52:13 +0200
Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with  
Ultramonkey
I have been working with a similar setup myself, but for some reason  
I ended up ditching the

UltraMonkey setup because I just couldn't get it to work right.

It's been quite a while since my effort, so I don't remember what  
the exact issue was.

I got registrations to work, but had some other sip-dialog issues.

We have since then changed over to running OpenSIPs as a  
loadbalancer in front of
multiple FreeSWITCH instances. This setup is still in testing, but  
seemlingy works fine
(and if it doesn't, it's my own fault for writing a bad opensips  
config).


After we have done some more testing I can create a wiki-page with  
config details.



Best regards,
Even André


Thanks, Even, that would be great! I might have to give up on the  
ultramonkey solution, since I can't find anyone who has made it  
work. It's too bad, because it would fit well with the rest of our  
architecture.


Mike van Lammeren

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Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

[Raffaele P. Guidi]

And, should someone succed replicating this setup, consider writing about it
on the wiki :)

On Thu, Oct 1, 2009 at 18:45, Mike van Lammeren wrote:

> Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
> heartbeat and ldirectord for load-balancing, fail-over and high
> availability! I'm probably not the first one to do it, but as near as Google
> and I can tell, I'm the first one to write about it.
> Here's how you can duplicate my setup:
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
> 2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
> following these instructions:
> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
>  3. Configure both
> FreeSWITCH boxes, and make sure they are both working.
> 4. Follow (most of) these instructions from Daniel Aliaman's blog. They
> were written for Asterisk, but since a SIP connection is a SIP connection,
> most of the document applies to FreeSWITCH:
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
> The one
> problem I ran into was the IP address and port to which FreeSWITCH was
> bound. The default is to use the primary address, which works great
> out-of-the-box for everything else. When a client tried to register, all it
> got back was an ICMP error -- Destination Unreachable, Port Unreachable.
> That error is returned when no sockets are listening for UDP packets. To get
> FreeSWITCH to listen for your Virtual IP, you need to set it in two places:
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
> That should do it! If you have any success, please report to this list.
>
> Keep in mind that if you want to do something like conferencing between two
> registered clients, then you have to deal with the fact that the clients may
> or may not be on the same box.
>
> Mike van Lammeren
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren  > wrote:
>
>>
>> On Mon, Sep 28, 2009 at 9:05 PM, "Even André Fiskvik" > > wrote:
>>
>>> From: "Even André Fiskvik" 
>>> To: freeswitch-users@lists.freeswitch.org
>>> Date: Mon, 28 Sep 2009 22:52:13 +0200
>>> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>>> I have been working with a similar setup myself, but for some reason I
>>> ended up ditching theUltraMonkey setup because I just couldn't get it to
>>> work right.
>>>
>>> It's been quite a while since my effort, so I don't remember what the
>>> exact issue was.
>>> I got registrations to work, but had some other sip-dialog issues.
>>>
>>> We have since then changed over to running OpenSIPs as a loadbalancer in
>>> front of
>>> multiple FreeSWITCH instances. This setup is still in testing, but
>>> seemlingy works fine
>>> (and if it doesn't, it's my own fault for writing a bad opensips config).
>>>
>>> After we have done some more testing I can create a wiki-page with config
>>> details.
>>>
>>>
>>> Best regards,
>>> Even André
>>>
>>>
>> Thanks, Even, that would be great! I might have to give up on the
>> ultramonkey solution, since I can't find anyone who has made it work. It's
>> too bad, because it would fit well with the rest of our architecture.
>>
>> Mike van Lammeren
>>
>
>
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Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones

[Jerry Richards]

I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _  

From: Anthony Minessale [mailto:anthony.miness...@gmail.com] 
Sent: Thursday, October 01, 2009 9:36 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards 
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com
 
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[Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

[Dmitry Bely]

My SIP provider allows only one call (incoming or outgoing) via one
SIP account. For FreeSWITCH I have configured it as public DID
extension and outgoing gateway. Now I would like to transfer to
another gw (or generate "limit exceded") when one tries to place an
outgoing call while incoming call is in progress. How tho do that?
Limiting the number of outgoing calls is easy (mod_limit), but how to
take into account incoming one?

- Dmitry Bely

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[Freeswitch-users] Connecting FS to Hicom 300

[Russell.Mosemann]

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


  

  

  


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 ," 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

  




  
   
 
   
   
   
   
   
   
 
   


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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Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

[Mike van Lammeren]

Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
heartbeat and ldirectord for load-balancing, fail-over and high
availability! I'm probably not the first one to do it, but as near as Google
and I can tell, I'm the first one to write about it.
Here's how you can duplicate my setup:

1. Install Ubuntu Server 8 on four machines, either real or VM.
2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
following these instructions:
http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
3. Configure both
FreeSWITCH boxes, and make sure they are both working.
4. Follow (most of) these instructions from Daniel Aliaman's blog. They were
written for Asterisk, but since a SIP connection is a SIP connection, most
of the document applies to FreeSWITCH:
http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf

The one
problem I ran into was the IP address and port to which FreeSWITCH was
bound. The default is to use the primary address, which works great
out-of-the-box for everything else. When a client tried to register, all it
got back was an ICMP error -- Destination Unreachable, Port Unreachable.
That error is returned when no sockets are listening for UDP packets. To get
FreeSWITCH to listen for your Virtual IP, you need to set it in two places:

5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".

That should do it! If you have any success, please report to this list.

Keep in mind that if you want to do something like conferencing between two
registered clients, then you have to deal with the fact that the clients may
or may not be on the same box.

Mike van Lammeren



On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren
wrote:

>
> On Mon, Sep 28, 2009 at 9:05 PM, "Even André Fiskvik"  > wrote:
>
>> From: "Even André Fiskvik" 
>> To: freeswitch-users@lists.freeswitch.org
>> Date: Mon, 28 Sep 2009 22:52:13 +0200
>> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>> I have been working with a similar setup myself, but for some reason I
>> ended up ditching theUltraMonkey setup because I just couldn't get it to
>> work right.
>>
>> It's been quite a while since my effort, so I don't remember what the
>> exact issue was.
>> I got registrations to work, but had some other sip-dialog issues.
>>
>> We have since then changed over to running OpenSIPs as a loadbalancer in
>> front of
>> multiple FreeSWITCH instances. This setup is still in testing, but
>> seemlingy works fine
>> (and if it doesn't, it's my own fault for writing a bad opensips config).
>>
>> After we have done some more testing I can create a wiki-page with config
>> details.
>>
>>
>> Best regards,
>> Even André
>>
>>
> Thanks, Even, that would be great! I might have to give up on the
> ultramonkey solution, since I can't find anyone who has made it work. It's
> too bad, because it would fit well with the rest of our architecture.
>
> Mike van Lammeren
>
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Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH To Subscribing Phones

[Anthony Minessale]

which phone is it,
we tested it with eyebeam and it appears to work for us.


On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards
wrote:

>
> By the way, I see the following lines at the FS console, which might be a
> clue as to why this is happening.  Could someone point me toward what might
> cause this?  I set the "manage-presence" parameter to "true" in each XML
> file where I saw it defined.
>
> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>   ...
> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>   ...
> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>   ...
> [WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping
>
>
> Best Regards,
> Jerry
>
>
> -Original Message-
> From: Jerry Richards [mailto:jerry.richa...@teotech.com]
> Sent: Wednesday, September 30, 2009 9:12 AM
> To: 'freeswitch-users@lists.freeswitch.org'
> Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones
>
> I have two phones configured to subscribe to each other's presence status.
> When I change the presence status in one phone, I see the SIP PUBLISH
> message going to FS, but I don't see FS relaying that presence status to
> the
> subscribing phone.  Does anyone know why?
>
> Best Regards,
> Jerry
>
>
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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com 
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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Re: [Freeswitch-users] Dialplan Issue

[Shelby Ramsey]

Just to confirm ... works like a champ.

Thanks again!!!
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Re: [Freeswitch-users] Dialplan Issue

[Brian West]

I wouldn't go that far! :P  You might be able to get away with it on  
the patch tony wrote but not sure.

/b

On Oct 1, 2009, at 10:59 AM, Shelby Ramsey wrote:

> SIP is fantastic :)


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Re: [Freeswitch-users] Dialplan Issue

[Shelby Ramsey]

Brian,

Thanks for the info.  I guess I'll go read section 19.1 of RFC3261 again.  I
do think the above has a valid host portion (I don't think the port is
required).

I'm not so sure that putting params in the user portion of the uri is valid
(from the RFC it states sip:user:passw...@host:port;uri-parameters?headers).

The issue is that in the real world this is done all the time  SIP is
fantastic :)

Shelby


On Thu, Oct 1, 2009 at 9:42 AM, Brian West  wrote:

>
> On Oct 1, 2009, at 9:27 AM, Shelby Ramsey wrote:
>
> 
>
>
> This will produce an INVALID sip uri... You can not feed this to sofia
> it'll get PISSED.
>
> Its missing the host portion.
>
>
> So the issue is the placement of the user params  if they are before
> the @ FS will send a 500 internal server error ... if they are after the @
> FS will send a 302.  Unfortunately placing the user params after the @
> doesn't quite conform to the way other devices expect to receive the 302 for
> this application.
>
> Any help would be greatly appreciated.
>
> Shelby
>
> PS ... hats off to the author of mod_memcache ... that is extremely useful!
>
>
>
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Re: [Freeswitch-users] Dialplan Issue

[Shelby Ramsey]

Tony,

Once again ... you are the man!

I'll try this right now.

SDR
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Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH To Subscribing Phones

[Jerry Richards]

By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping

 
Best Regards,
Jerry
 

-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com] 
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


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Re: [Freeswitch-users] Dialplan Issue

[Anthony Minessale]

you left too fast.

1) I told you if you put <> around the sip uri it will work.
2) I told you I added a patch in tree to add one for you if it's not
supplied so update to trunk.


On Thu, Oct 1, 2009 at 9:27 AM, Shelby Ramsey  wrote:

> Hello:
>
> I asked this on IRC yesterday and I think I confused everyone involved.  So
> I apologize in advance here for reposting the question and if I wasted
> anyone's time.
>
> So here is the issue I'm having.  I'm trying to use FS as a redirect server
> (specifically to serve up LNP queries via 302 redirects).  But I'm having an
> issue where based on the string in the dialplan FS will respond with a 500
> internal error message instead of a 300 redirect.
>
> The call flow should be this:
>-- remote party sends an Invite to my FS instance
>-- FS should respond with a 302
>
> The following works as expected (FS will send a 302 when it receives an
> Invite):
>
> 
>
> However if I do this (which is the way the response should look) FS will
> respond with a 500 internal server error:
>
>  data="sip:${destination_number};rn=${rn};npdi=...@${network_addr}"/>
>
> So the issue is the placement of the user params  if they are before
> the @ FS will send a 500 internal server error ... if they are after the @
> FS will send a 302.  Unfortunately placing the user params after the @
> doesn't quite conform to the way other devices expect to receive the 302 for
> this application.
>
> Any help would be greatly appreciated.
>
> Shelby
>
> PS ... hats off to the author of mod_memcache ... that is extremely useful!
>
>
>
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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com 
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org 
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] REGISTER fails with 407 after minutes of success register

[Brian West]

Thanks for posting the logs... But I'm not going to spend the time to  
download it.. unzip it and look at it... I would rather just click a  
link with the logs in plain text and read them in my browser.


I'll do it now but next time lets not add steps to the process that  
are not needed.  This goes for Jira too don't upload zip files of text  
logs that just makes it harder for us to quickly help you.


This isn't going to help me much know why Sofia/FreeSWITCH isn't  
working.


sofia profile xxx siptrace on
press F8
sofia loglevel all 9

Then post that please.

/b

On Oct 1, 2009, at 9:35 AM, Fernando Testa wrote:

In the link below you have the entire SIP trace from system startup  
until start receiving this annoying 407 Proxy Auth Required,  
preventing FS to register successfully on the Ericsson Pabx.
You can notice multiple registrations from named ericsson_1050 to  
ericsson_1064 that starts failing after ~50 minutes after the boot.  
Issuing a 'sofia external profile restart' solves the registration  
problems.

Brian, thanks for reply, but I really didn't get your point.
Thank you, I apreciate any help.

http://dl.getdropbox.com/u/410277/sip.log.gz


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Re: [Freeswitch-users] Dialplan Issue

[Brian West]

On Oct 1, 2009, at 9:27 AM, Shelby Ramsey wrote:




This will produce an INVALID sip uri... You can not feed this to sofia  
it'll get PISSED.


Its missing the host portion.



So the issue is the placement of the user params  if they are  
before the @ FS will send a 500 internal server error ... if they  
are after the @ FS will send a 302.  Unfortunately placing the user  
params after the @ doesn't quite conform to the way other devices  
expect to receive the 302 for this application.


Any help would be greatly appreciated.

Shelby

PS ... hats off to the author of mod_memcache ... that is extremely  
useful!


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Re: [Freeswitch-users] REGISTER fails with 407 after minutes of success register

[Fernando Testa]

In the link below you have the entire SIP trace from system startup until
start receiving this annoying 407 Proxy Auth Required, preventing FS to
register successfully on the Ericsson Pabx.You can notice multiple
registrations from named ericsson_1050 to ericsson_1064 that starts failing
after ~50 minutes after the boot. Issuing a 'sofia external profile restart'
solves the registration problems.
Brian, thanks for reply, but I really didn't get your point.
Thank you, I apreciate any help.

http://dl.getdropbox.com/u/410277/sip.log.gz

On Wed, Sep 30, 2009 at 10:33 AM, Brian West  wrote:

> I don't see a challenge in your 407 so how can we answer properly
> against the far end if they don't challenge us?
>
> /b
>
> On Sep 30, 2009, at 7:06 AM, Fernando Testa wrote:
>
> > Brian,
> >
> > Thanks for the reply. The SIP trace is mixed with the log (+ sofia
> > loglevel all 9) on the pastebin I mention on the previous email
> > (http://pastebin.freeswitch.org/10517 ). That log is from FS 1.0.4.
> >
> >
> > On Tue, Sep 29, 2009 at 10:13 PM, Brian West 
> > wrote:
> >> I need the sip trace.
> >>
> >> /b
> >>
> >> On Sep 29, 2009, at 7:28 PM, Fernando Testa wrote:
> >>
> >>> Hi all,
> >>>
> >>> I have a FS that registers on an Ericsson pabx as gateway under
> >>> sip_external.
> >>> This gateway start registering on the Ericsson ok, but after a
> >>> while,
> >>> around 50mins, it fails with the logs below.
> >>> If I hit *sofia profile external restart* on fs_cli then the gateway
> >>> returns to register with success (that means, we get 200 OK from
> >>> Ericsson).
> >>> This happens with FS 1.0.4 release tarball, and trunk r15011.
> >>> I found similar situations on these links, but not actually found a
> >>> solution.
> >>> Any help is very welcome.
> >>>
> >>> *OS*
> >>> CentOS 5.3 x86_64
> >>> 4Gb RAM
> >>>
> >>> log at http://pastebin.freeswitch.org/10517
> >>>
> >>> conf/sip_profiles/external/ericsson.xml:
> >>>  
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>  
> >>>
> >
> > ___
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-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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[Freeswitch-users] Dialplan Issue

[Shelby Ramsey]

Hello:

I asked this on IRC yesterday and I think I confused everyone involved.  So
I apologize in advance here for reposting the question and if I wasted
anyone's time.

So here is the issue I'm having.  I'm trying to use FS as a redirect server
(specifically to serve up LNP queries via 302 redirects).  But I'm having an
issue where based on the string in the dialplan FS will respond with a 500
internal error message instead of a 300 redirect.

The call flow should be this:
   -- remote party sends an Invite to my FS instance
   -- FS should respond with a 302

The following works as expected (FS will send a 302 when it receives an
Invite):



However if I do this (which is the way the response should look) FS will
respond with a 500 internal server error:



So the issue is the placement of the user params  if they are before the
@ FS will send a 500 internal server error ... if they are after the @ FS
will send a 302.  Unfortunately placing the user params after the @ doesn't
quite conform to the way other devices expect to receive the 302 for this
application.

Any help would be greatly appreciated.

Shelby

PS ... hats off to the author of mod_memcache ... that is extremely useful!
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Re: [Freeswitch-users] Listening to a connected call [barge in]

[Vinuth Madinur]

Hi,
Use the eavesdrop command.
Just supply it with the call UUID and the extension of B.
Wiki has more details.

Thanks,
Vinuth.


On Thu, Oct 1, 2009 at 7:20 PM, Nagalenoj  wrote:

>
> In ES outbound, I need to do the following,
> * A calls 2000(FS ES outbound extension)
> * In the script, It'll answer the call, play some files and get the reply
> from A(as voice).
> * Simultaneously(when doing the above), the script has to call B.
> * When B attends the call, B has to listen to the live conversation between
> 2000 & A.
>
> How should I do.?
>
> I've tried this with async mode and by listening to the events. But I
> couldn't do it. Help me to do this..
>
> Regards,
> Nagalenoj
> --
> View this message in context:
> http://www.nabble.com/Listening-to-a-connected-call--barge-in--tp25696889p25696889.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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[Freeswitch-users] Listening to a connected call [barge in]

[Nagalenoj]

In ES outbound, I need to do the following, 
* A calls 2000(FS ES outbound extension)
* In the script, It'll answer the call, play some files and get the reply
from A(as voice).
* Simultaneously(when doing the above), the script has to call B.
* When B attends the call, B has to listen to the live conversation between
2000 & A.

How should I do.?

I've tried this with async mode and by listening to the events. But I
couldn't do it. Help me to do this..

Regards,
Nagalenoj
-- 
View this message in context: 
http://www.nabble.com/Listening-to-a-connected-call--barge-in--tp25696889p25696889.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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[Freeswitch-users] Problem with subscription expire

[Helmut Kuper]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hello,

it seems exired subsciptions are never cleared in FS.

A look into sofia_presence.c confirms explains this

/* negative in exptime means keep bumping up sub time to avoid a snafu
where every device has it's own rules about subscriptions
   that somehow barely resemble the RFC not that
I blame them because the RFC MAY be amibiguous and SHOULD be deleted.
   So to avoid the problem we keep resetting the
expiration date of the subscription so it never expires.

   Eybeam completely ignores this option and
most other subscription-state: directives from rfc3265 and still expires.
   Polycom is happy to keep upping the
subscription expiry back to the original time on each new notify.
   The rest ... who knows...?

*/

For some reasons subscriptions created by Snom phones are filling up the
sip_subscriptions table over time. This leads to some kind of DOS by FS
against the subscribing phone ... The subscribtions are differentiate by
call-id. This can be explained by RFC 3842 chapter 3.6 where expired
subscriptions must be renewed with a NEW call-id. Because there is no
hint about unsubscribing the old subscription I guess the clean up
process has to be done by FS.

Any way to get FS to do this job? Since there is no creation date or
expire value which represents the expire as a timestamp I have no way to
clean up the table manually via sql and cronjob - except cleaning the
whole table ...


A further (but background) question is, why do the subscriptions expire
in snom phones at all ...



regards
helmut
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