Re: [Freeswitch-users] sip message logging and analysis
I'm using VQManager (there is a 30 day trial) and it's useful for seeing who does what / when per call; it's very easy to install... From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Frank @ Impact Sent: Thursday, December 17, 2009 4:02 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] sip message logging and analysis I bit off topic but... Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier's first response is that we dropped the call. But this is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? ___ 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
[Freeswitch-users] Skype SIP Beta
Skype have opened their beta program up to all comers. http://www.skype.com/business/products/pbx-systems/sip/support-faqs/#paddedContent Three lines in a sip_profile make FreeSWITCH talk nicely; but using the PCMU codec. Any progress on SILK native support? Last I saw was discussion back in September with Brian lamenting that Skype was hard to work with on this. I know I could use mod_skypiax; but having a native solution would be one less IT headache. Thx, Chris. ___ 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
Re: [Freeswitch-users] NAT problems migrating from Version 1.0.trunk (13168M) to Version 1.0.trunk (15166)
I've tried all sorts of debug and parameter changes over the weekend,
but still can't figure out the correct solution.
If I disable timers on the sip profile then all works fine.
param name=enable-timer value=false/
But that seems like a hack; not a correct solution. With the build
13168M (which is pre the new NAT functionality) everything worked fine.
The SIP trace shows the phones and FreeSWITCH happily exchanging NOTIFY
and 200 OK messages. Audio's working - just calls timeout after 100
seconds with RECOVERY_ON_TIMER_EXPIRE. Is enforcement of this timer new
functionality - and really just exposing a problem I've always had
before?
The config is (50 Polycom Phones - NAT - Internet - Amazon EC2)
I would really appreciate some pointers on what to look for; additional
trace that might reveal something.
Thanks, Chris.
On Fri, 16 Oct 2009 20:06:52 -0700, Chris Fowler ch...@fowler.cc
said:
Hi,
We've been using 13168M in production for some time now (works great).
I want to get us onto the latest build but am having problems getting
NAT to work.
Phones can register; can dial test #, but after 100 seconds the
call is disconnected with error:
2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
sofia/internal/1...@myhost.mydomain.com [CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]
I took the standard internal.xml and vars.xml files from the new build
and made the following modifications - which worked previously:
modify conf/vars.xml and update
X-PRE-PROCESS cmd=set data=domain=myhost.mydomain.com/
X-PRE-PROCESS cmd=set data=bind_server_ip=1.2.3.4/
X-PRE-PROCESS cmd=set data=external_rtp_ip=1.2.3.4/
X-PRE-PROCESS cmd=set data=external_sip_ip=1.2.3.4/
Modify conf/sip_profiles/internal.xml
param name=aggressive-nat-detection value=true/ param
name=ext-rtp-ip value=$${external_rtp_ip}/ param
name=ext-sip-ip value=$${external_sip_ip}/ param
name=NDLB-received-in-nat-reg-contact value=true/ param
name=NDLB-force-rport value=true/ param
name=NDLB-broken-auth-hash value=true/
The big difference I note is that on PRODUCTION (which works) sofia
status profile internal yields:
URL sip:mod_so...@1.2.3.4:5060
BIND-URLsip:mod_so...@1.2.3.4:5060;maddr=10.250.35.224
But on Test I see:
URL sip:mod_so...@10.250.66.210:5060
BIND-URLsip:mod_so...@10.250.66.210:5060
Any ideas?
Thanks, Chris.
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[Freeswitch-users] NAT problems migrating from Version 1.0.trunk (13168M) to Version 1.0.trunk (15166)
Hi,
We've been using 13168M in production for some time now (works great).
I want to get us onto the latest build but am having problems getting
NAT to work.
Phones can register; can dial test #, but after 100 seconds the
call is disconnected with error:
2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
sofia/internal/1...@myhost.mydomain.com [CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]
I took the standard internal.xml and vars.xml files from the new build
and made the following modifications - which worked previously:
modify conf/vars.xml and update
X-PRE-PROCESS cmd=set data=domain=myhost.mydomain.com/
X-PRE-PROCESS cmd=set data=bind_server_ip=1.2.3.4/
X-PRE-PROCESS cmd=set data=external_rtp_ip=1.2.3.4/
X-PRE-PROCESS cmd=set data=external_sip_ip=1.2.3.4/
Modify conf/sip_profiles/internal.xml
param name=aggressive-nat-detection value=true/ param
name=ext-rtp-ip value=$${external_rtp_ip}/ param
name=ext-sip-ip value=$${external_sip_ip}/ param
name=NDLB-received-in-nat-reg-contact value=true/ param
name=NDLB-force-rport value=true/ param
name=NDLB-broken-auth-hash value=true/
The big difference I note is that on PRODUCTION (which works) sofia
status profile internal yields:
URL sip:mod_so...@1.2.3.4:5060
BIND-URLsip:mod_so...@1.2.3.4:5060;maddr=10.250.35.224
But on Test I see:
URL sip:mod_so...@10.250.66.210:5060
BIND-URLsip:mod_so...@10.250.66.210:5060
Any ideas?
Thanks, Chris.
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Re: [Freeswitch-users] Freeswitch memory usage is too high
Hi Ray, This was a problem some time ago (couple of months ago). Are you running the latest build? Chris. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Raymond Chandler Sent: Wednesday, July 01, 2009 6:11 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch memory usage is too high On 07/01/2009 08:29 AM, Muhammad Danish Moosa wrote: Hi Freeswitch is being used in a scenario where two endpoints are running traffic with bypass media mode. Performance is good and all things are smooth. But as the time goes after starting freeswitch, it starts consuming almost whole of memory. How much is the whole? You should see the memory usage level off, it won't keep growing forever. -Ray ___ 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 ___ 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
[Freeswitch-users] Audio delay when conferencing
I'm using FreeSWITCH (Build 13168M) and we're having intermittent multi-second
delays on conference bridges with more than three participants (this is not a
new issue - just bubbled to the top of the stack to address).
The server is running on Amazon's AWS c1.medium instance, CentOS 5.0 with
Kernel 2.6.18 32-bit i386.
I recorded a conference which shows the problem nicely:
http://cfowl.postinbox.com/c.wav
Callers are coming on via the internal sofia profile from various physical
locations. I'm not sure how to proceed with debugging this issue - advice
welcome.
profiles
profile name=default
param name=domain value=$${domain}/
param name=rate value=32000/
param name=interval value=10/
param name=energy-level value=100/
param name=member-flags value=waste/
param name=caller-controls value=rightscale/
param name=sound-prefix value=$${base_dir}/sounds/en/us/callie/
param name=muted-sound value=conference/conf-muted.wav/
param name=unmuted-sound value=conference/conf-unmuted.wav/
param name=alone-sound value=conference/conf-alone.wav/
param name=moh-sound value=$${hold_music}/
param name=enter-sound value=tone_stream://%(200,0,500,600,700)/
param name=exit-sound
value=tone_stream://%(500,0,300,200,100,50,25)/
param name=kicked-sound value=conference/conf-kicked.wav/
param name=locked-sound value=conference/conf-locked.wav/
param name=is-locked-sound value=conference/conf-is-locked.wav/
param name=is-unlocked-sound value=conference/conf-is-unlocked.wav/
param name=pin-sound value=conference/conf-pin.wav/
param name=bad-pin-sound value=conference/conf-bad-pin.wav/
param name=comfort-noise-level value=1400/
param name=caller-id-name value=$${outbound_caller_name}/
param name=caller-id-number value=$${outbound_caller_id}/
param name=comfort-noise value=true/
/profile
Thanks, Chris.
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Re: [Freeswitch-users] Using Variables in Dialplans
I added a Wiki example on this. Hopefully it will help the next FreeSWITCHer... http://wiki.freeswitch.org/wiki/Time_of_Day_Routing From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, April 23, 2009 08:29 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Using Variables in Dialplans On Wed, Apr 22, 2009 at 5:20 PM, Chris Fowler ch...@fowler.cc wrote: Brian, Michael, Thanks for the help - I had read that but not fully comprehended it until you spun it the way you did. Here's what I ended up with - if there's optimization that could be done let me know. Happy to update the wiki if this is a common request. That looks pretty good. It is clean and readable and most importantly it sounds like it does what you need! :) -MC ___ 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
[Freeswitch-users] Using Variables in Dialplans
I have the following defined:
!-- Billing Open? --
extension name=billing_open continue=true
!-- man strftime - M-F, 9AM to 5PM --
condition field=${strftime(%w)} expression=^([1-5])$/
condition field=${strftime(%H%M)}
expression=^((09|1[0-6])[0-5][0-9]|1700)$
action application=set data=billingopen=true/
/condition
/extension
!-- billing line --
extension name=billing
condition field=destination_number expression=^billing$|^2001$/
condition field=billingopen expression=^true$
...
But despite the Billing Open ext firing correctly:
Action set(billingopen=true)
The Billing extenstion fails to get the data:
Regex (FAIL) [billing] billingopen() =~ /^true$/ break=on-false
Please could someone point me in the right direction for using variables
correctly in a dialplan? I must be missing something simple...
Thanks, Chris.
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Re: [Freeswitch-users] Using Variables in Dialplans
Brian, Michael,
Thanks for the help - I had read that but not fully comprehended it until you
spun it the way you did.
Here's what I ended up with - if there's optimization that could be done let me
know. Happy to update the wiki if this is a common request.
!-- Provide an internal ext # for Sales --
extension name=RS-Sales-x2002
condition field=destination_number expression=^2002$
action application=transfer data=RS-Sales/
/condition
/extension
!-- Setup variables prior to transfering to call handler for sales --
extension name=RS-Sales continue=true
!-- man strftime - M-F, 9AM to 5PM --
condition field=destination_number expression=^RS-Sales$/
condition field=${strftime(%w)} expression=^([1-5])$/
condition field=${strftime(%H%M)}
expression=^((09|1[0-6])[0-5][0-9]|1700)$
action application=set data=RS-Sales_open=true/
action application=transfer data=xfer-to-sales/
anti-action application=set data=RS-Sales_open=false/
anti-action application=transfer data=xfer-to-sales/
/condition
/extension
!-- Handle Sales Call --
!-- If Sales is open then route to extension first then vMail; else direct
to vMail --
extension name=xfer-to-sales
condition field=destination_number expression=^xfer-to-sales$/
condition field=${RS-Sales_open} expression=^true$
action application=bridge data=user/1...@${domain_name}/
action application=answer/
action application=sleep data=2000/
action application=voicemail data=default ${domain_name} 2001/
anti-action application=voicemail data=default ${domain_name}
2001/
/condition
/extension
Thx, Chris.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Wednesday, April 22, 2009 13:10
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Using Variables in Dialplans
On Wed, Apr 22, 2009 at 12:19 PM, Chris Fowler ch...@fowler.cc wrote:
I have the following defined:
!-- Billing Open? --
extension name=billing_open continue=true
!-- man strftime - M-F, 9AM to 5PM --
condition field=${strftime(%w)} expression=^([1-5])$/
condition field=${strftime(%H%M)}
expression=^((09|1[0-6])[0-5][0-9]|1700)$
action application=set data=billingopen=true/
/condition
/extension
!-- billing line --
extension name=billing
condition field=destination_number expression=^billing$|^2001$/
condition field=billingopen expression=^true$
...
But despite the Billing Open ext firing correctly:
Action set(billingopen=true)
The Billing extenstion fails to get the data:
Regex (FAIL) [billing] billingopen() =~ /^true$/ break=on-false
Please could someone point me in the right direction for using variables
correctly in a dialplan? I must be missing something simple...
Thanks, Chris.
This is a classic case of dialplan is parsed all at once. The reason that it
is failing is because ${billingopen} is not defined when the extension named
billing is parsed. You need another pass through the dialplan, for example by
adding a transfer app to your billing_open extension:
!-- Billing Open? --
extension name=billing_open continue=true
!-- man strftime - M-F, 9AM to 5PM --
condition field=${strftime(%w)} expression=^([1-5])$/
condition field=${strftime(%H%M)} expression=^((09|1[0-6])[0-5]
action application=set data=billingopen=true/
action application=transfer data=billing/
/condition
/extension
This will send the call back through the dialplan and into the billing
extension, this time with the value of ${billing_open} == true so you can now
work with it. The originally dialed number is now available in the ${rdnis}
channel variable in case you need it.
Have fun and let us know how it goes.
-MC
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Re: [Freeswitch-users] Configure FS using Flowroute.com
You've called your gateway flowroute vs. mine which is sip.flowroute.com;
you need to amend the dialplan to reference flowroute; then you should be set.
flowroute gateway sip:...@sip.flowroute.com
action application=bridge
data=sofia/gateway/flowroute/${default_provider_username}#$1/
C.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Alex Harris
Sent: Monday, April 20, 2009 20:22
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com
The vars.xml has been modified with the following:
X-PRE-PROCESS cmd=set data=default_provider=sip.flowroute.com/
X-PRE-PROCESS cmd=set data=default_provider_username=***/
X-PRE-PROCESS cmd=set data=default_provider_password=/
X-PRE-PROCESS cmd=set
data=default_provider_from_domain=sip.flowroute.com/
!-- true or false --
X-PRE-PROCESS cmd=set data=default_provider_register=true/
X-PRE-PROCESS cmd=set data=default_provider_contact=5000/
the 01_example.xml has been modified with the code you provided
The Status output is:
Name Type Data
State
===
internal profile sip:mod_so...@**.***.219.221:5060
RUNNING (0)
external profile sip:mod_so...@**.***.219.221:5080
RUNNING (0)
flowroute gateway sip:...@sip.flowroute.com
REGED
**.***.219.221 alias internal
ALIASED
internal-ipv6 profile
sip:mod_so...@[2002:63be:dbdd::63be:dbdd]:5060RUNNING (0)
default alias internal
ALIASED
nat alias external
ALIASED
outbound alias external
ALIASED
==
When I try to dial out this is what I am getting:
freeswi...@s4bs 2009-04-20 20:19:45 [NOTICE] switch_channel.c:592
switch_channel_set_name() New Channel sofia/internal/1...@**.***.219.221
[7bbb17eb-e953-1743-84b5-d4b0ae651332] 2009-04-20 20:19:45 [INFO]
mod_dialplan_xml.c:252 dialplan_hunt() Processing Alex-15184282539 in
context default 2009-04-20 20:19:45 [ERR] mod_sofia.c:2411
sofia_outgoing_channel() Invalid Gateway 2009-04-20 20:19:45 [NOTICE]
mod_sofia.c:2624 sofia_outgoing_channel() Close Cha
nnel N/A [CS_NEW]
2009-04-20 20:19:45 [ERR] switch_ivr_originate.c:1459 switch_ivr_originate()
Can
not create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2009-04-20 20:19:45 [INFO] mod_dptools.c:2036 audio_bridge_function()
Originate
Failed. Cause: INVALID_NUMBER_FORMAT
2009-04-20 20:19:45 [NOTICE] mod_dptools.c:2068 audio_bridge_function()
Hangup s
ofia/internal/1...@**.***.219.221 [CS_EXECUTE] [INVALID_NUMBER_FORMAT]
2009-04-20 20:19:45 [NOTICE] switch_core_session.c:1018
switch_core_session_thre
ad() Session 3 (sofia/internal/1...@**.***.219.221) Ended
2009-04-20 20:19:45 [NOTICE] switch_core_session.c:1020
switch_core_session_thre
ad() Close Channel sofia/internal/1...@**.***.219.221 [CS_DONE]
Thank you for the help.
~Alex
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris
Fowler
Sent: Monday, April 20, 2009 7:09 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com
Are you either restarting FS or issuing the reloadxml command (press F6 on
the console) after making these changes?
Did you modify vars.xml per my last note?
Also, check out http://wiki.freeswitch.org/wiki/Getting_Started_Guide - it's
worth investing the time to understand how FS parses the various config
files. Look in /usr/local/freeswitch/log at freeswitch.xml.fsxml - this
file contains the entire free switch configuration since last as parsed by
the application.
Seems you're not hitting the issue I was with FlowRoute - but as the error
indicates you're trying to route a call out from a gateway that does not
exist.
What's the output of sofia status (F5 on the console)? It should show:
sip.flowroute.com gateway sip:...@sip.flowroute.com
REGED
Chris.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Alex
Harris
Sent: Monday, April 20, 2009 17:05
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com
I modified the 01_example.com.xml with the code below and I am still getting
Invalid Gateway. I tried creating a file 01_flowroute.com.xml and placed
Re: [Freeswitch-users] Configure FS using Flowroute.com
About two weeks ago FlowRoute stopped working with FreeSwitch. Looking at the
SIP trace there is chatter about a Proxy Auth required. I ran out of time to
debug as this was on a production system.
The work around was:
1) You need to prefix the dialed number with the FlowRoute Account Number in
the dialplan:
action application=bridge
data=sofia/gateway/sip.flowroute.com/${default_provider_username}#$1/
2) Specifiy FreeSWITCH's Public IP in the FlowRoute Interconnection page in the
Outbound Allowed IPs
I've updated the wiki with the work around. This needs more debug as it used
to work with registration only - which is preferable if your public IP is
changing. (FWIW - Inbound calls have always worked).
Alex: To make flowroute your default provider add to
/usr/local/freeswitch/conf/vars.xml
X-PRE-PROCESS cmd=set data=default_provider=sip.flowroute.com/
X-PRE-PROCESS cmd=set data=default_provider_username=/
X-PRE-PROCESS cmd=set data=default_provider_password=secret/
X-PRE-PROCESS cmd=set
data=default_provider_from_domain=sip.flowroute.com/
Chris.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Alex Harris
Sent: Monday, April 20, 2009 16:04
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Configure FS using Flowroute.com
I placed the xml found at
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Flowroute.com into the
file \conf\directory\default\flowroute.com.xml
Should I have placed it in another location?
Thank you,
Alex
You'll have to do this
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Flowroute.com
/b
On Apr 20, 2009, at 5:15 PM, Alex Harris wrote:
I am new to VoIP and FS and I was hoping that someone could help me
configure my FS server.
I have add flowroute.com.xml to C:\Program Files (x86)\FreeSWITCH
\conf\directory\default using the example on the wiki.
I have updated vars.xml with
X-PRE-PROCESS cmd=set data=default_provider=Flowroute.com/
X-PRE-PROCESS cmd=set
data=default_provider_username=*/
X-PRE-PROCESS cmd=set data=default_provider_password=***/
X-PRE-PROCESS cmd=set
data=default_provider_from_domain=Flowroute.com/
!-- true or false --
X-PRE-PROCESS cmd=set data=default_provider_register=true/
X-PRE-PROCESS cmd=set data=default_provider_contact=5000/
Right now everytime I try to dial out I get a Gateway Invalid error
and I try to dial the DID and only get a busy tone. Any help would
be much appriciate.
Thank you,
Alex
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br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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End of Freeswitch-users Digest, Vol 34, Issue 111
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Re: [Freeswitch-users] Configure FS using Flowroute.com
Are you either restarting FS or issuing the reloadxml command (press F6 on the console) after making these changes? Did you modify vars.xml per my last note? Also, check out http://wiki.freeswitch.org/wiki/Getting_Started_Guide - it's worth investing the time to understand how FS parses the various config files. Look in /usr/local/freeswitch/log at freeswitch.xml.fsxml - this file contains the entire free switch configuration since last as parsed by the application. Seems you're not hitting the issue I was with FlowRoute - but as the error indicates you're trying to route a call out from a gateway that does not exist. What's the output of sofia status (F5 on the console)? It should show: sip.flowroute.com gateway sip:...@sip.flowroute.com REGED Chris. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Alex Harris Sent: Monday, April 20, 2009 17:05 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com I modified the 01_example.com.xml with the code below and I am still getting Invalid Gateway. I tried creating a file 01_flowroute.com.xml and placed the code below and had the same unfortunate result. On flowroute site I have placed my IP in the outbound allowed IPs and also inbound routes. Is there something else that I should do to try to get things to work? Sorry, for asking for so much help. ~Alex ___ 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
Re: [Freeswitch-users] Polycom register problem in private address
I'm running FS on Amazons' EC2 compute cloud (AWS) and have 30 Polycom phones working happily in this config. I modified the Internal profile in /usr/local/freeswitch/conf/sip_profiles/internal.xml to include: param name=aggressive-nat-detection value=true/ param name=NDLB-force-rport value=true/ The phones connect on port 5060 - nothing specical to config in the mac-phone.cfg file for the phone; just host, port, user/pass. Chris. ___ 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
Re: [Freeswitch-users] Polycom register problem in private address
Brian: Did you request public IP's for your EC2 instance? Yes; there is an Elastic IP (EIP) associated with the instance. Also specify the EIP in vars.xml X-PRE-PROCESS cmd=set data=bind_server_ip=insert EIP here/ X-PRE-PROCESS cmd=set data=external_rtp_ip=insert EIP here/ X-PRE-PROCESS cmd=set data=external_sip_ip=insert EIP here/ Re: Wiki Yup I need to get on this. FWIW - I work for RightScale; our computer room is empty except for routers and switches. *Everything* else lives in the Cloud :-) Cheers, Chris. ___ 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
Re: [Freeswitch-users] DTMF Missing Digits
Sent: Wednesday, March 25, 2009 12:43 btw you'll have to reinstall your phrase macros make vm-sync I think should do it if it doesn't let me know... we removed the 250ms sleeps and that was the problem which we fixed. I re-did the macros; the only change I could detect was the elimination of the 250ms sleeps; and the change to: macro name=welcome pause=250 I'm running build 12782; should this have fixed it? If so, I will follow the bug reporting instructions you sent earlier. Thanks, Chris. Here's the errors caught today on my production system. 2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '1101' 2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '55' 2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '015' 2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '' 2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' ___ 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
Re: [Freeswitch-users] DTMF Missing Digits
Did you provide the menu you are using and what you expect to happen?
Here's the setup;
Caller - FlowRoute - FreeSwitch
menu name=main_ivr
greet-long=phrase:welcome
greet-short=phrase:top-menu
invalid-sound=ivr/ivr-that_was_an_invalid_entry.wav
exit-sound=ivr/ivr-operator.wav
timeout =1
inter-digit-timeout=1500
max-failures=2
max-timeouts=7
digit-len=4
entry action=menu-exec-app digits=/^(10[0-2][0-9])$/
param=transfer $1 XML public/
entry action=menu-exec-app digits=/^(30\d{2})$/
param=transfer $1 XML default/
entry action=menu-exec-app digits=0 param=transfer 1000 XML
public/ !-- Send to the operator extension --
entry action=menu-exec-app digits=# param=transfer 6000 XML
default/
/menu
macro name=welcome pause=250
input pattern=(.*)
match
action function=play-file
data=/usr/local/freeswitch/sounds/fr1.wav/
action function=play-file
data=/usr/local/freeswitch/sounds/fr2.wav/
action function=play-file
data=/usr/local/freeswitch/sounds/if-u-know-ext-dial.wav/
action function=play-file
data=/usr/local/freeswitch/sounds/fr3.wav/
/match
/input
/macro
macro name=top-menu pause=250
input pattern=(.*)
match
action function=play-file
data=/usr/local/freeswitch/sounds/if-u-know-ext-dial.wav/
action function=play-file
data=/usr/local/freeswitch/sounds/fr3.wav/
/match
/input
/macro
B: Right and that is the fix for this. If you have the sleep's in your
phrase macro's remove them and use the pause= param... you shouldn't have
any problems.
Still seeing multiple issues logged during ivr process for
mis-interpreted DTMF. Here's today's list from our production server.
2009-03-27 06:38:59 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input
'1100'
2009-03-27 07:20:33 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input
'1101'
2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '55'
2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '015'
2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input
''
2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 11:58:35 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '028'
2009-03-27 11:59:27 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '050'
2009-03-27 12:01:52 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 12:02:01 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 12:02:41 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 12:02:53 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
Any other debug I can capture to assist?
Thanks, Chris.
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[Freeswitch-users] DTMF Missing Digits
Any thoughts on why FS saw all digits 1029 but only reports '029'?
2009-03-25 10:48:45 [DEBUG] switch_ivr_menu.c:364 play_and_collect()
digits '029'
Config:
menu name=main_ivr
greet-long=phrase:welcome
greet-short=phrase:top-menu
invalid-sound=ivr/ivr-that_was_an_invalid_entry.wav
exit-sound=voicemail/vm-goodbye.wav
timeout =1
inter-digit-timeout=1500
max-failures=3
max-timeouts=7
digit-len=4
entry action=menu-exec-app digits=/^(10[0-2][0-9])$/
param=transfer $1 XML public/
entry action=menu-exec-app digits=/^(30\d{2})$/
param=transfer $1 XML default/
entry action=menu-exec-app digits=0 param=transfer 1000 XML
public/
entry action=menu-exec-app digits=# param=transfer 6000 XML
default/
entry action=menu-top digits=9/
Trace:
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf()
RTP RECV DTMF 1:2000
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1301 do_2833() Send start
packet for [1] ts=1129880426 dur=160/160/2000 seq=2804
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=320/320/2000 seq=2805
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=480/480/2000 seq=2806
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=640/640/2000 seq=2807
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=800/800/2000 seq=2808
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=960/960/2000 seq=2809
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1120/1120/2000 seq=2810
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1280/1280/2000 seq=2811
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1440/1440/2000 seq=2812
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1600/1600/2000 seq=2813
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1760/1760/2000 seq=2814
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1920/1920/2000 seq=2815
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [1] ts=1129880426 dur=2080/2080/2000 seq=2816
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [1] ts=1129880426 dur=2080/2080/2000 seq=2817
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [1] ts=1129880426 dur=2080/2080/2000 seq=2818
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf()
RTP RECV DTMF 0:2160
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1301 do_2833() Send start
packet for [0] ts=1129884426 dur=160/160/2160 seq=2819
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=320/320/2160 seq=2820
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=480/480/2160 seq=2821
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=640/640/2160 seq=2822
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=800/800/2160 seq=2823
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=960/960/2160 seq=2824
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1120/1120/2160 seq=2825
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1280/1280/2160 seq=2826
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1440/1440/2160 seq=2827
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1600/1600/2160 seq=2828
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1760/1760/2160 seq=2829
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1920/1920/2160 seq=2830
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=2080/2080/2160 seq=2831
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [0] ts=1129884426 dur=2240/2240/2160 seq=2832
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [0] ts=1129884426 dur=2240/2240/2160 seq=2833
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [0] ts=1129884426 dur=2240/2240/2160 seq=2834
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf()
Re: [Freeswitch-users] DTMF Missing Digits
First off what SVN rev? Remember when reporting issues try to include all the information you can! Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647) ___ 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
Re: [Freeswitch-users] Possible memory / cpu leak
Thanks for the tip Brian. Here's a link to the valgrind output : http://cfowl.postinbox.com/vg.log Chris. ___ 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
[Freeswitch-users] Possible memory / cpu leak
Hi, Ive been seeing an issue where FreeSWITCHs CPU and memory utilization climb over time; a restart of FS clears up the problem. See graphs for the past week. http://cfowl.postinbox.com/fs.jpg Observed on the Release Candidate, and then upgraded to the current trunk a couple of times. Currently running version FreeSWITCH Version 1.0.trunk (12604). This is seen both when FS is being used (~200 calls/day, and over the weekend when ~5 calls/day). How can I best debug this? Thanks, Chris. ___ 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
Re: [Freeswitch-users] Possible memory / cpu leak
Jay : what happens in your dialplan ? Nothing special; no external script execution just default pattern matching to route to extensions (per the stock config). Brian: Can you update to SVN trunk as of now? Yup, I will pull the trunk and report back in 24 hours. Chris. ___ 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
Re: [Freeswitch-users] Possible memory / cpu leak
Brian: Can you update to SVN trunk as of now? I updated - version reports: FreeSWITCH Version 1.0.trunk (12631) Only difference I note with this build is that upon shutdown FS now SegFaults. The mem/cpu usage continues to slowly climb. snip 2009-03-16 20:59:32 [CONSOLE] switch_loadable_module.c:1237 do_shutdown() Stopping: mod_spidermonkey Segmentation fault (core dumped) Anthony - nothing special is a bit vague. I modified the dial plan to accept extension in the 1000-1029 range Added DialByLast name using directory.lua (from the wiki) Modified the ivr config to play company specific greetings Voicemail is used a few times per day freeswi...@ip-10-250-155-18 sofia status API CALL [sofia(status)] output: Name Type Data State = internal profile sip:mod_so...@xxx.xxx.xxx.xx:5060 RUNNING (0) external profile sip:mod_so...@xxx.xxx.xxx.xx:5080 RUNNING (0) sip.flowroute.com gateway sip:xxx...@sip.flowroute.com REGED inphonex gateway sip:xx...@sip.inphonex.com REGED callwithus-did-xx gateway sip:...@east.callwithus.com REGED callwithus-did-xx gateway sip:...@east.callwithus.com REGED callwithus-did-xx gateway sip:...@east.callwithus.com REGED default alias internal ALIASED nat alias external ALIASED x..aaa alias internal ALIASED outbound alias external ALIASED = Anthony - valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg Nothing got logged, here's the output. Did I invoke valgrind incorrectly? ==32545== Memcheck, a memory error detector. ==32545== Copyright (C) 2002-2006, and GNU GPL'd, by Julian Seward et al. ==32545== Using LibVEX rev 1658, a library for dynamic binary translation. ==32545== Copyright (C) 2004-2006, and GNU GPL'd, by OpenWorks LLP. ==32545== Using valgrind-3.2.1, a dynamic binary instrumentation framework. ==32545== Copyright (C) 2000-2006, and GNU GPL'd, by Julian Seward et al. ==32545== For more details, rerun with: -v ==32545== ==32545== My PID = 32545, parent PID = 32511. Prog and args are: ==32545==/usr/local/freeswitch/bin/freeswitch ==32545==-vg ==32545== ___ 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
