gmane.comp.telephony.freeswitch.user

Re: DTMF outbound call

2013-05-23T13:27:11

It's not often time but now i see again message error
2013-05-23 15:24:44.452821 [WARNING] mod_sofia.c:1363 Pass 2833 mode 
may not work on a transcoded call.
2013-05-23 15:24:44.452821 [DEBUG] switch_rtp.c:2534 Correct ip/port 
confirmed.
2013-05-23 15:24:44.372152 [WARNING] mod_sofia.c:1363 Pass 2833 mode 
may not work on a transcoded call.

here my config file external.xml and internal.xml

<profile name="external">
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<!--aliases are other names that will work as a valid profile name for 
this profile-->
<aliases>
<!--
<alias name="outbound"/>
<alias name="nat"/>
-->
</aliases>

<!-- Outbound Registrations -->
<gateways>
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
</gateways>

<domains>
<domain name="all" alias="false" parse="true"/>
</domains>

<settings>
<param name="inbound-late-negotiation" value="true"/>
<param name="debug" value="7"/>
<param name="pass-rfc2833" value="true"/>
<param name="sip-trace" value="true"/>
<param name="sip-capture" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="$${external_sip_port}"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="zrtp-passthru" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="local-network-acl" value="localnet.auto"/>
<param name="manage-presence" value="false"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<param name="tls" value="$${external_ssl_enable}"/>
<param name="tls-only" value="false"/>
<param name="tls-bind-params" value="transport=tls"/>
<param name="tls-sip-port" value="$${external_tls_port}"/>
<param name="tls-cert-dir" value="$${external_ssl_dir}"/>
<param name="tls-passphrase" value=""/>
<param name="tls-verify-date" value="true"/>
<param name="tls-verify-policy" value="none"/>
<param name="tls-verify-depth" value="2"/>
<param name="tls-verify-in-subjects" value=""/>
<param name="tls-version" value="$${sip_tls_version}"/>

</settings>
</profile>


<profile name="internal">
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<!--aliases are other names that will work as a valid profile name for 
this profile-->
<aliases>
<!--
<alias name="default"/>
-->
</aliases>

<!-- Outbound Registrations -->
<gateways>
<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
</gateways>

<domains>
<!-- indicator to parse the directory for domains with parse="true" 
to get gateways-->
<!--<domain name="$${domain}" parse="true"/>-->
<!-- indicator to parse the directory for domains with parse="true" 
to get gateways and alias every domain to this profile -->
<!--<domain name="all" alias="true" parse="true"/>-->
<domain name="all" alias="true" parse="false"/>
</domains>

<settings>
<param name="inbound-late-negotiation" value="true"/>
<param name="debug" value="7"/>
<param name="sip-trace" value="true"/>
<param name="sip-capture" value="true"/>
<param name="watchdog-enabled" value="no"/>
<param name="watchdog-step-timeout" value="30000"/>
<param name="watchdog-event-timeout" value="30000"/>
<param name="log-auth-failures" value="true"/>
<param name="forward-unsolicited-mwi-notify" value="false"/>
<param name="context" value="public"/>
<param name="rfc2833-pt" value="101"/>
<param name="pass-rfc2833" value="true"/>
<param name="sip-port" value="$${internal_sip_port}"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="rtp-timer-name" value="soft"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="apply-nat-acl" value="nat.auto"/>
<!--<param name="apply-inbound-acl" value="domains"/>-->
<param name="local-network-acl" value="localnet.auto"/>
<param name="record-path" value="$${recordings_dir}"/>
<param name="record-template" 
value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<param name="manage-presence" value="true"/>
<param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
<param name="presence-privacy" value="$${presence_privacy}"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="tls" value="$${internal_ssl_enable}"/>
<param name="tls-only" value="false"/>
<param name="tls-bind-params" value="transport=tls"/>
<param name="tls-sip-port" value="$${internal_tls_port}"/>
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
<param name="tls-passphrase" value=""/>
<param name="tls-verify-date" value="true"/>
<param name="tls-verify-policy" value="none"/>
<param name="tls-verify-depth" value="2"/>
<param name="tls-verify-in-subjects" value=""/>
<param name="tls-version" value="$${sip_tls_version}"/>
<param name="nonce-ttl" value="60"/>
<!--<param name="auth-calls" value="$${internal_auth_calls}"/>-->
<param name="inbound-reg-force-matching-username" value="true"/>
<param name="auth-all-packets" value="false"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<param name="force-register-domain" value="$${domain}"/>
<param name="force-subscription-domain" value="$${domain}"/>
<param name="force-register-db-domain" value="$${domain}"/>
<param name="challenge-realm" value="auto_from"/>
<param name="pass-callee-id" value="true"/>
</settings>
</profile>


Tks adavnce for your help

Le 2013-05-23 14:58, Michael Jerris a écrit :

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Re: DTMF outbound call

Michael Jerris — 2013-05-23T13:24:14

How do you use what?

On May 23, 2013, at 9:07 AM, ehermouet-ogZ//5ZwM7ZGWvitb5QawA< at >public.gmane.org wrote:



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Re: DTMF outbound call

2013-05-23T13:07:40

how do you use it without interface ? it's server with only ssh access.
tks
Le 2013-05-23 14:58, Michael Jerris a écrit :


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Re: why it hangs up after originate

Michael Jerris — 2013-05-23T13:00:56

On May 22, 2013, at 10:12 PM, Vincent Xia <gmangudai-Re5JQEeQqe8AvxtiuMwx3w< at >public.gmane.org> wrote:


http://wiki.freeswitch.org/wiki/Mod_commands#originate

"&<application_name>"
you need & not < at >

Mike


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Re: DTMF outbound call

Michael Jerris — 2013-05-23T12:58:42

this log does not seem to have a complete call let alone any attempt at dtmf.  I don't see anything wrong from this log but as I said, its incomplete.  If you pcap the traffic, do you see 2833 dtmf flowing ?

Mike

On May 23, 2013, at 8:44 AM, ehermouet-ogZ//5ZwM7ZGWvitb5QawA< at >public.gmane.org wrote:



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Re: Conference questions

Michael Jerris — 2013-05-23T12:51:09

On May 23, 2013, at 2:00 AM, Matthew Cordes <cordes.matthew-Re5JQEeQqe8AvxtiuMwx3w< at >public.gmane.org> wrote:


You don't really. Why exactly would you want to do this?


https://wiki.freeswitch.org/wiki/Mod_conference

conference flags "wait-mod"


my recommendation : https://wiki.freeswitch.org/wiki/Mod_xml_curl


If your doing dialplan via mod_xml_curl you can do all that validation in your dialplan lookup.


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Re: DTMF outbound call

2013-05-23T12:44:22

Yes

http://pastebin.freeswitch.org/20947

Le 2013-05-23 14:28, Michael Jerris a écrit :

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Re: DTMF outbound call

Michael Jerris — 2013-05-23T12:28:35

Did you ever post a new log after you changed codec negotiation settings?

On May 23, 2013, at 1:27 AM, Hermouet Erwan <ehermouet-ogZ//5ZwM7ZGWvitb5QawA< at >public.gmane.org> wrote:


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Re: Max. Number of 8 span PRI cards

Michael Jerris — 2013-05-23T12:30:58

Talk to sangoma, they have tested at > 3 cards I know, but I don't know system specifics.

On May 23, 2013, at 7:27 AM, Ashish gautam <ashish-kMxHSFh002n/PtFMR13I2A< at >public.gmane.org> wrote:


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Conference questions

Matthew Cordes — 2013-05-23T06:00:17

Hi,

I'm new to FreeSwitch and starting to experiment with conferences. I've
browsed the wiki and I'm half way through the ebook and I'm running into a
few questions I'm hoping I might be able to get some help with.

1. How do I translate conference cli commands (e.g. "conference xxx mute
yyy") into a dialplan xml actions?

2. The default behavior of the conference is to disable hold music and
allow the participants to speak when there are two or more callers. How
might I  keep the conference on hold until a particular caller joins?
Additionally when this special user leaves I'd like to place the conference
back on hold.

3. Where might I find info regarding loading configuration information
(particularly the user directory) from an external source (probably a
database and probably via mod_python)?

4. I'd like to have some private conferences that only certain callers can
join. I'll know the callers' ids before hand. I'm assuming a reasonable way
to handle is with user groups. How would I test if a user is a member of a
particular group?

Thanks,
-Matt
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why it hangs up after originate

Vincent Xia — 2013-05-23T02:12:56

when im executing console commands like "originate user/1005
< at >eavesdrop(uuid)" to have 1005 listen to a call party, 1005 rings then i
click answer and find out it hangs up immediately, the freeswitch log says
"[NOTICE] switch_core_state_machine.c:262 sofia/internal/sip:1005< at >IPaddr:5088
has executed the last dialplan instruction, hanging up."

does that mean i need to add something the to the dialplan? is there a
simple solution by extending the originate command like "originate
user/1005 < at >eavesdrop(uuid) &wait_for_hangup"?
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Re: inband DTMF bleeding through

Trever L. Adams — 2013-05-23T12:28:02

I have not yet tried what was suggested. I believe it works, but I just
haven't got around to trying it yet as I have a lot on my plate and
FreeSWITCH isn't a top priority yet.

freetdm_disable_dtmf=true
freetdm_pre_buffer_size=N (size in bytes of a SLIN pre buffer so it can
cut detected dtmf fragments out)

I do not know if there is any documentation for those yet or not.

I hope it helps.

Trever

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Re: how to turn an ongoing call in to aconference

Vincent Xia — 2013-05-23T00:42:27

sorry for the late response, normally the caller C will make an inbound
call through a certain number or preferably a prefix+caller A or B's number
(e.g. **+1001), i also would like to know what should i do if caller A or B
would dial a prefix+caller C's number and pull him into a three-way call.

many thanks!



2013/5/21 Brian Foster <bdfoster-v1fKxmSWr8YAvxtiuMwx3w< at >public.gmane.org>

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SDP manipulation and sofia_glue.c (developerhelp)

Daniel Ivanov — 2013-05-23T12:01:54

I am having difficulties forcefully stripping the a:crypto lines from the
remote sdp string. At first i was doing <action
application="set"><![CDATA[switch_r_sdp=$sdp]]>
</action>
from the diaplan, but this breaks the proxy media easily, because it
doesn't patch the SDP later for glueing.
I commented the following lines where the patcher is checking if the sdp
string has been set before :
void sofia_glue_tech_patch_sdp(private_object_t *tech_pvt)
{
        switch_size_t len;
        char *p, *q, *pe, *qe;
        int has_video = 0, has_audio = 0, has_ip = 0;
        char port_buf[25] = "";
        char vport_buf[25] = "";
        char *new_sdp;
        int bad = 0;
*/**
*        if (zstr(tech_pvt->local_sdp_str)) {*
*                return;*
*        }*
**/*

And now i got the correct sdp sent to the b-leg(patched for proxy media and
nat), but no audio is going either way. Would there be a more clever way to
"rip" some lines from the SDP and not break the NAT/Proxy media processing
afterwards?

Help is much appreciated.
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Re: Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg

Adam Raszynski — 2013-05-23T11:47:54

Finally solved this problem by:
1. outbound-codec-prefs to PCMU< at >20i,PCMA< at >20i
2. setting inbound-codec-negotiation = scrooge

Spent 4 hours on fixing that, maybe my solution will help someone with same
problem in the future
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Re: Cannot ring extension from DID

Mike Hendrie — 2013-05-23T11:45:06

I made a change to 1000 this morning and I can now dial each others
extensions.

When I call the number I still go directly to the voicemail when extension
1000 should ring.


freeswitch< at >internal> sofia status profile internal reg


Registrations:

=================================================================================================

Call-ID:        519830183003-muKoWyl7ZPNp9BbEXvtYYg< at >public.gmane.org

User:           1001-fnel96BYnfCU7knFcz9Dig< at >public.gmane.org

Contact:        "user" <sip:1001-muKoWyl7ZPNp9BbEXvtYYg< at >public.gmane.org:45048
;transport=udp;fs_nat=yes;fs_path=sip%3A1001%4010.2.1.209%3A45048%3Btransport%3Dudp>

Agent:          Sipdroid/3.0 beta/SCH-I605

Status:         Registered(UDP-NAT)(unknown) EXP(2013-05-22 23:44:29)
EXPSECS(3501)

Host:           GothamCity-00

IP:             10.2.1.209

Port:           45048

Auth-User:      1001

Auth-Realm:     gothamcity.xom

MWI-Account:    1001-fnel96BYnfCU7knFcz9Dig< at >public.gmane.org


Call-ID:        7eb5078e-316ad43b-c274631c-ZxN56qRi8w00ZgchR83hnQ< at >public.gmane.org

User:           1000-fnel96BYnfCU7knFcz9Dig< at >public.gmane.org

Contact:        "user" <sip:1000-ZxN56qRi8w00ZgchR83hnQ< at >public.gmane.org:5060
;transport=tcp;fs_nat=yes;fs_path=sip%3A1000%4010.2.1.50%3A64461%3Btransport%3Dtcp>

Agent:          PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069

Status:         Registered(TCP-NAT)(unknown) EXP(2013-05-22 22:48:44)
EXPSECS(156)

Host:           GothamCity-00

IP:             10.2.1.50

Port:           64461

Auth-User:      1000

Auth-Realm:     gothamcity.xom

MWI-Account:    1000-fnel96BYnfCU7knFcz9Dig< at >public.gmane.org


Total items returned: 2

=================================================================================================


Mike Hendrie
T: 847.366.5881
E: mike-0JVlv975Uqdt1OO0OYaSVA< at >public.gmane.org
On May 23, 2013 6:16 AM, "Philippe Le Toquin" <philippe-9OQXqz8nkhQ< at >public.gmane.org> wrote:

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Lua creating multiple session's

Jon Schøpzinsky — 2013-05-23T11:36:46

Hello List,

Im trying to do dial multiple destination through lua, on a single
incoming call.

I do know that i could do a simple session:execute("bridge",
"dst1,dst2,dst3") but I need to do it in individual session, for
processing I need to do in a later version of the lua script.

I loop through the destinations that needs to be called, creating a new
session for each destination, and storing that in an array.
Firstly, it seems as freeswitch.Session doesn't reply right away, but
waits for early-media. Thats ok though, but makes dialling mobile devices
a rather long wait.

The problem is, that when I do the second freeswitch.Session, it seems to
hold up further lua processing, until the last created call is answered.
Is this how its supposed to work.

Heres my current code:

local legs = {}

for key, dev in pairs(dstDevices) do
      freeswitch.consoleLog("info", "Lets call " .. dev.username .. " with
tech " .. dev.devicetech .. "\n")
      if dev.devicetech == "1" then
        freeswitch.consoleLog("info", "calling " .. dev.username .. " on
uasbc\n")
        legs[key] = freeswitch.Session("sofia/gateway/uasbc01/" ..
dev.username);
        freeswitch.consoleLog("info", "called " .. dev.username .. " on
uasbc\n")
      elseif dev.devicetech == "2" then
        freeswitch.consoleLog("info", "calling " .. dev.username .. " on
ccsbc, mvno\n")
        legs[key] =
freeswitch.Session("{origination_caller_id_name=+xxxxxxxx,origination_calle
r_id_number=+xxxxxxxxx}sofia/gateway/ccsbc01/+xx" ..
string.match(dev.username, "^mvno_(.+)"))
        freeswitch.consoleLog("info", "called " .. dev.username .. " on
ccsbc, mvno\n")
      end
  End


freeswitch.consoleLog("info", "Enter loop now\n")

It doesn't reach the last consoleLog until the last call is answered.



Venlig hilsen/kind regards

Jon Leren Schøpzinsky


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Max. Number of 8 span PRI cards

Ashish gautam — 2013-05-23T11:27:39

Hi,

I want to ask that is it possible to use 3 eight span PRI cards on a single
FS server? I mean is there any limit up to which the system performs
optimally on this?

Please throw some light on this.

Thanks

Re: Cannot ring extension from DID

Philippe Le Toquin — 2013-05-23T11:12:57

so your extension are not registered

On my FS I can see my own extension.

Now this is where I am going to learn something because I have no idea 
why you can call from one extension to another if they are not registered.

On 13-05-22 10:48 PM, Mike Hendrie wrote:

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Re: Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg

Adam Raszynski — 2013-05-23T10:51:49

Changed also rtp-timer-name to none

now bridged call is OK, but when local user uses IVR provided by
bind-meta-app it gets distorted audio, when leaving IVR and switching back
to the caller audio is OK again

2013/5/23 Adam Raszynski <netcentrica-Re5JQEeQqe8AvxtiuMwx3w< at >public.gmane.org>

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Re: Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg

Adam Raszynski — 2013-05-23T10:24:40

I changed outbound-codec-prefs to PCMU< at >20i,PCMA< at >20i so timing on inbound
and outbound legs is the same

Info about async PTIME does not appear now in console, so I think now there
are synchronous

But the problem is not solved - audio stream from local user is still
lagging
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