Hello,
The problem we are trying to solve here is handling a busy state according
to the user's prefference (some want a busy to be heard, some want the call
to go to voicemail, and some want to get the second call).
The first step is finding that an extension is busy. It would be nice in
2009/7/14 Michael Collins m...@freeswitch.org
On Mon, Jul 13, 2009 at 9:30 PM, Dome Charoenyost d...@tel.co.th wrote:
2009/7/14 Michael Collins m...@freeswitch.org:
What phone number do you call back? I mean, how do you know what the
customer's number is? Do you go by the caller id
Hello,
I am trying to integrate Polycom phones with a FrewSwitch server, and have
some problems with BLF and directed pickup.
I've defined a buddy list with BW (buddy watch) on. One of
the phone's line buttons (one fo the 3 ones on a Polycom-501 model) is
assigned to this buddy and indeed
and the destination is free - ring it.
I know roughly how to do the last two items, but how can I catch the
SUBSCRIBE, modify the destination number and then call the actual function?
Thanks! __Yehavi:
2009/7/21 Yehavi Bourvine yehavi.bourv...@gmail.com
Hello,
I am trying
Hello,
I have a problem when trying to put a call on hold: I get the above
message and the call is disconnected. Any idea where to look for the source
of the problem?
One thing I've tried is limiting all phones to use only one codec, but it
doesn't help...
the call is disconnected.
Thanks! __Yehavi:
2009/9/8 Jason White ja...@jasonjgw.net
Yehavi Bourvine yehavi.bourv...@gmail.com wrote:
I have a problem when trying to put a call on hold: I get the above
message and the call is disconnected. Any idea where to look
is this the only FreeSWITCH log
output you have in this transfer?
On Sep 15, 2009, at 1:55 AM, Yehavi Bourvine wrote:
Hello Jason,
Sorry for the delay in answering - I saw your reply only now as it
got burried with some other stuff...
Anyway, I attach bellow the relevant sip trace
I've solved the problem: I am running it on a Fedora-10 system. Once I've
installed a vanilla kernel (from kernel.org) the problem went away.
BTW, can someone shed the light on the kernel's bug which I see mentions
of it in this list?
Thanks! __Yehavi:
Hello,
I am trying to run FreeSwitch on a machine which has more than one
interface, all of them should be used for SIP. The FreeSwitch binds only to
the first one. I tried setting bind_server_ip to either auto or 0.0.0.0
but it doesn't help.
Any idea what to do?
Thanks!
__Yehavi:
2009/9/24 Seven Du dujinf...@gmail.com
It not possible to use 0.0.0.0 for on profile. however, you can create more
sip profiles for each of your interfaces. Search freeswitch-users archievs
then you will find similar topics.
2009/9/24 Yehavi Bourvine
Hello,
We would like to handle an incoming call to a busy phone according
to user's prefference: Some want waiting call, some want to just reject the
call, and others want to send the call to voicemail.
We have a small JavaScript which tests the status of the destination and
the user's will
Thanks! It works!
__Yehavi:
2009/11/1 Anthony Minessale anthony.miness...@gmail.com
try session.execute(hangup, user_busy);
On Sun, Nov 1, 2009 at 8:24 AM, Yehavi Bourvine
yehavi.bourv...@gmail.com wrote:
Hello,
We would like to handle an incoming call
Hello,
While trying to display the *called party *name on SNOM phones I've found
that the field sent to the phone needs to be changed slightly in order to
make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects
Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and
the sip_cid_type variable?
http://wiki.freeswitch.org/wiki/Variable_sip_cid_type
On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine yehavi.bourv...@gmail.com
wrote:
Hello,
While trying to display the called party name on SNOM phones I've
found
that the field sent to the phone needs to be changed
I am using Polycoms (430 and 501) with FreeSwitch. How do you provision
them? Via WEB or config files?
If you use config files than I can send you some sample files.
Regards, __Yehavi:
On Nov 12, 2009, at 11:41 AM, Adam Ford wrote:
Has anyone used a Polycom
Hello Brian,
the situation is as follows: Our PBX machine has more than one interface,
each one has a profile. Some phones are registered via one interface and tje
others on the other. The call should be sent usinbg the profile of the
destination as if not, the IP address of the server in the
Thanks Mike! However, this doesn't fully solve my problem. When using
sofia_contact() indeed it works ok with finding the destination's profile.
However, it breaks the BLFs...
When calling *sofia/sip_profile/local-user%local-do**main* the BLF works ok.
When calling
Hello,
We have a mix of phones that support RTP encryption and those that do not.
I have to support both types in the meanwhile, and would like to have
encryption enabled on the relevant leg, even if the other leg does not
support it (why? one of our ATAs either must have it unencrypted or have
Hello Anthony,
Indeed I see the reference to this channel variable in the code, but when
trying to access it from the dial plan it is empty... I try to get the value
of ${sip_profile_name} and it is empty.
Thanks! __Yehavi:
2009/11/23 Anthony Minessale
Hello,
I am trying to set a Polycom 501 phone to do conferencing via the
conference room on Freeswitch rather than on the phone (as on the phone it
is limited to 3 participants only). Anyone had success with it?
I have on the Freeswitch an extension named Conf.* which activates the
conference
Hello,
I would like Freeswitch to pass the Remote-Party-ID field of the called
party (sent in the Ringing OK when answering the call) back to the
originator's phone. How can I do that?
The drive for this is: Our Freeswitch is connected via a Cisco gateway and
PRI to the university's phone
the variables effective_callee_id_name and
effective_callee_id_number in your dp before you answer the call
On Dec 1, 2009 12:08 AM, Yehavi Bourvine yehavi.bourv...@gmail.com
wrote:
Hello,
I would like Freeswitch to pass the Remote-Party-ID field of the called
party (sent in the Ringing OK
Are you on SVN trunk? As far as I recall the callee_id_number/name stuff
isnt in 1.0.4.
No, because the SVN has problems with Emailing the voicemail...
We use 1.0.4 and set sip_callee_id_number/name which works. I would like to
not set it and get it from the other side...
It is MODAPP-373.
Thanks, __yehavi:
2009/12/1 Michael Jerris m...@jerris.com
What is the jira bug number on this voicemail email issue? I don't
recall seeing it.
Mike
On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine
yehavi.bourv...@gmail.com wrote:
Are you on SVN trunk
Hello,
We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On the PRI
there is a Nortel with Q.Sig. After a lot of configuration trials I've
managed to set it to send back the connected name over the SIP (i.e. when a
call goes from SIP to PRI, the PRI sends back the connected name
Hello,
I have Polycom phones which send only RFC-2833 (or inband which I dislike)
and they should go out to the PSTN via a Cisco gateway. The Cisco gateway
has some bug and accepts only INFO.
I did a few tests:
- Some of the phones are on different profile than the Cisco. On their
BTW, I forgot to update: I changed the bridge parameters to use
sofia_contact() and it solved the problem. I also fixed the presence problem
I had before with sofia_contact() (added presence_id to the bridge command).
Regards, __Yehavi:
2009/11/24 Yehavi Bourvine
-number
(config-if)#isdn outgoing ie called-number
-metik
Yehavi Bourvine wrote:
Hello,
We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On
the PRI there is a Nortel with Q.Sig. After a lot of configuration
trials I've managed to set it to send back the connected name over
Hello,
I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they
should support TLS also (will try it next week; up to now I preffered to not
use TLS so I can sniff the traffic and debug things).
Regards, __Yehavi:
2009/12/4 Mark Campbell-Smith
I am taking my words back... The Cisco sends back what I want.
I got confused because the Nortel sends the name only for the connected PBX
and not for the othes ones (although it gets this infomation from them).
Thanks, __Yehavi:
2009/12/3 Yehavi Bourvine yehavi.bourv
!
On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine
yehavi.bourv...@gmail.com wrote:
Hello,
I have AudioCodes MP and Vega ATA adapters. They both support SRTP;
they
should support TLS also (will try it next week; up to now I preffered to
not
use TLS so I can sniff the traffic and debug
...
Regards, __Yehavi:
2009/12/3 Ognjen Seslija osesl...@gmail.com
Bear in mind that FS will accept both 2833 and INFO in any profile on an
inbound call. Param dtmf-type is valid only for outbound calls from the
profile.
Ognjen
On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine
Hello Metik,
2009/12/6 Metik freeswitch-users-l...@metik.com
You previously stated that your Cisco gateway has some bug that
prevents you from using RFC2833, did you enable dtmf-relay rtp-nte on
the voip dial-peer that the call is using?
It is a PSTN dialpeer here, and it cannot be
Hello,
I have a few questions about Ploycom's usage and provisioning for which I
found no answers neither at the docs nor on the WEB:
- I would like to enable SIP/TLS. for this I have to import the root
certificate. How can I do it via the XML config files? the only method I
found is
server, I have seen
some issues when multple dtmf relay types are left enabled on a voip
dial peer. Also, there are some (older) IOS versions that have issues
with DTMF duration which cause digits to be misinterpreted by the
far-end (PSTN/POTS) but not ignored altogether.
-metik
Yehavi
Hello,
I have some black hole understading how to debug Freeswitch. In fs_cli I
do sofia debug all 7 and indeed get a lot of debugging messages on the
console; however, the logfiles get only Critical messages. Where do I define
which messages go to the logfile?
And in a related topic: I've
it.
*VegaStream Europa-50*: SRTP works. Waiting for Vega for instructions how to
enable TLS from the WEB interface.
Regards, __Yehavi:
2009/12/4 Yehavi Bourvine yehavi.bourv...@gmail.com
I'll report when I am done.
So far I've enabled only SRTP and both support
Hello,
In the WIKI page that talks about Freeswitch performance there is a
sentence:
*libsofia only handles 1 thread per profile, so if that is your bottle neck
use more profiles*
How can I enable more than one profile on the same interface? Won't they
colide when using the same IP and port?
I would like all phones have the same general configuration... If no other
way, then I'll do that.
Thanks, __Yehavi:
2009/12/13 Seven Du dujinf...@gmail.com
you can use the same ip with different port
2009/12/13, Yehavi Bourvine yehavi.bourv...@gmail.com:
Hello
We are still on a small proof of concept system, but I am looking at the
future...
Thanks, __Yehavi:
2009/12/13 Frank Carmickle fr...@carmickle.com
On Sun, Dec 13, Yehavi Bourvine wrote:
I would like all phones have the same general configuration... If no
other
way
Hello,
I am trying to debug a TLS handshake error between FreeSwitch and some
ATA. When setting the loglevel to 9 I get only a message that TLS handshake
failed. Is there some other debug command to show what happens during the
TLS handshake process?
Thanks!
/12/10 Brian West br...@freeswitch.org
I have confirmed it works with Polycom, Snom and a few others
polycom is the hardest to set due to having to put the ca cert into
the phone... but other than that its good.
/b
On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote:
An intermediate
:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Yehavi
Bourvine
*Sent:* Sunday, November 29, 2009 8:48 AM
*To:* freeswitch-users
*Subject:* [Freeswitch-users] Polycom 501 conferencing with FreeSwitch
Hello,
I am
, Yehavi Bourvine wrote:
After some discussions with Polycom support it seems that their
conferencing support is based on draft-ietf-sipping-cc-conferencing-03
(which is not the latest and is not compatible with the latest one).
Any idea whether it is possible to program Freeswitch to support
I am trying Audiocodes and Vegastream ATAs, and work with either the
manufacturer or the local representative here.
On SNOM I managed to make it work, and will try Polycom soon (once I manage
to grab one unit from our users...).
Thanks, __yehavi:
2009/12/17 Brian West
Try the following:
action application=hangup data=USER_BUSY/
I don't know whether it will work in your case, but here we use it to reject
a call while we want to signal that the remote party is busy.
Regards, __Yehavi:
2009/12/18 bcxml
:
2009/12/17 Yehavi Bourvine yehavi.bourv...@gmail.com
I am trying Audiocodes and Vegastream ATAs, and work with either the
manufacturer or the local representative here.
On SNOM I managed to make it work, and will try Polycom soon (once I manage
to grab one unit from our users
It is usually CODEC related. probably the SIP messages has the cause inside.
__Yehavi:
2009/12/22 Fred-145 codecompl...@free.fr
I found the cause for #2: The GS phone was still configured to use NAT,
even
though both XLite and GS are located in the same, private LAN.
after I uploaded the
ca cert and marked it as trusted/used on the phone.
/b
On Dec 20, 2009, at 8:26 AM, Yehavi Bourvine wrote:
I am trying now to set a Polycom to work with FreeSwitch and TLS. I have
a Polycom-501 which does not have an internal certificate, thus only one-way
certificate
Hello,
Is there anyone who is using SNOM with TLS encryption and shared lines and
it works?
We have 1.0.5pre9 connected to SNOM-820 with shared lines between 2-3
SNOM phones. The TLS is defined by adding transport=tls to the registrar
field (proxy is left blank). We noticed the following
waiting a response from
'engineering' to tell us if they plan to implement standard SRTP
support in the Linksys ATA's.
TLS is working fine.
On Thu, Dec 17, 2009 at 4:39 PM, Yehavi Bourvine
yehavi.bourv...@gmail.com wrote:
An interim update:
Audiocodes: No success yet. I am working
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