gmane.comp.telephony.freeswitch.user

How fs handle the sip-T message?

2013-05-25T02:22:13

Hi, support team

I know FS can pass out SS7 info via properity header.
While, how it handles an incoming SIP-T message with SS7 info encapsulated in "content-type". Will it parse it or discard it?


Thanks
Windy
-------------------
2013-05-25




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How to attche custom "content-type" inFreeswitch SIP message

2013-05-25T01:18:16

Dear support,

I''m trying to encapsulate my private "application/isup" in the SIP Msg. Normally, it should like below example.

It firstly addresses   "Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR". And then can write private "content-type" like "application/sdp".
Now I can see the application/sdp is encapsulated via variable "switch_r_sdp".
Is there anyway to encapsulated other customized "content-type"? 

Appreciate to get your reply.
---------------------------------------------------------------------------------------------------------------
Here is an example of SIP message which encapsulated "applicaiton/sdp" and "application/isup".

INVITE sip: 87896677-ZqO/wWYGQBIAvxtiuMwx3w< at >public.gmane.org;user=phone;SIP/2.0
From: "Caller"<sip:678923001-KdnAoeVyOkTCENZMoErytg< at >public.gmane.org>
To: <sip:87896677-ZqO/wWYGQBIAvxtiuMwx3w< at >public.gmane.org>;user=phone
Call-ID: QRLVLNKeKx-WDHAuwlEkR-EwhPPcTHOP-KdnAoeVyOkTCENZMoErytg< at >public.gmane.org
Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwl

dbh:query - insert id

Lloyd Aloysius — 2013-05-25T00:14:01

Hello All

How to get the id value after insert a record a record using dbh:query

*table_a - columns*.

id - auto increment
field1
field2


dbh:query("insert into table_a ( field1,field2) values ('11','Test')")


After insert how to get the table_a - id value for the inserted record?

Thanks
Lloyd
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WebRTC

mbo — 2013-05-24T18:51:02

I've seen there have been a couple of discussions about WebRTC support earlier this year. Are there any news on that?

Thanks

Markus
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Assertion when Transcoding

Guillermo Ruiz Camauer — 2013-05-24T17:25:06

I am getting the following assertion when attempting to record a G.729 call:

freeswitch: src/switch_channel.c:832: switch_channel_get_variable_dup:
Assertion `channel != ((void *)0)' failed.

The system has a Sangoma D100 card installed.  The call is established
using the G.729 codec, and after playing some prompts, I am attempting to
record an answer:

freeswitch-Q0ErXNX1Rub9dkYlTf3LZg< at >public.gmane.org:5000< at >internal> show codecs
type,name,ikey
codec,ADPCM (IMA),mod_spandsp
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.722,mod_spandsp
codec,G.726 16k,mod_spandsp
codec,G.726 16k (AAL2),mod_spandsp
codec,G.726 24k,mod_spandsp
codec,G.726 24k (AAL2),mod_spandsp
codec,G.726 32k,mod_spandsp
codec,G.726 32k (AAL2),mod_spandsp
codec,G.726 40k,mod_spandsp
codec,G.726 40k (AAL2),mod_spandsp
codec,GSM,mod_spandsp
codec,LPC-10,mod_spandsp
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Sangoma G72

Freeswitch versioning / tagging/releasemanagement

mbo — 2013-05-24T15:44:24

On the wiki I can see that the latest stable freeswitch release is 1.2.9. If I look into git (git tag -l -n1), I can see that there are many tags with a higher version, but older tag date.

v1.3.0          1.3.0 release
v1.3.1          Tagging version 1.3.1
v1.3.10         tag v1.3.10
v1.3.11         tag v1.3.11
v1.3.12         Retag v1.3.12
v1.3.13         tag v1.3.13
v1.3.14         tag v1.3.14
v1.3.15         tag v1.3.15
v1.3.16         tag v1.3.16
v1.3.17-final   tag v1.3.17-final
v1.3.2          Tagging version 1.3.2
v1.3.3          Tagging 1.3.3
v1.3.4          release v1.3.4
v1.3.5          release v1.3.5
v1.3.6          tag v1.3.6
v1.3.7          tag v1.3.7
v1.3.8          tag v1.3.8
v1.3.9          tag v1.3.9
v1.5.0          tag v1.5.0

What about those tags? How is the release management organized? 

Thanks

Markus
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voicemail is not working

juned — 2013-05-24T06:59:31

Hi All,

I am newbie to FS. so as a startup i have installed FS in mu local system to
test out the basic functionality and features. so i have registered default
users 1000 and 1001 in softphone( twinkle
<http://mfnboer.home.xs4all.nl/twinkle/>  ). registration was successful and
calls was also fine but when i tried to check voicemail then it didn't
worked.

what i did to test it out voicemail is, did call to 1001 and let it rang so
after 30 second if no answer is there then voicemail will be activated but
calls are released after 30 seconds.

As per documentation i came to know that by default voicemail is activated
in default extensions is it so ? or i am missing something.

Please point me to right direction, i want to have such a dialplan in which
user can leave voicemail in cases of busy,unavailable and not answering.

Thanks & Regards
Juned 

 



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ESL using bridge app doesn't return whichgateway was used

Clinton Goudie-Nice — 2013-05-23T23:22:03

When you make a bridge command using esl, where you specify multiple
gateways or sip dials separated by or bars, you can't figure out which
gateway was used.

For example, if you bridge to something like this:

sofia/gateway/SBC-GW2/+18019600000|sofia/gateway/SBC-GW1/+18019600000

The call could be bridged to either GW2 or GW1.

When the CHANNEL_BRIDGE event is returned, you can see the original string
in variable_current_application_data, and you may be able to infer the
destination based on IP address, but nothing clearly says what gateway is
used.

If you turn on the all events firehose, you can see the CHANNEL_CREATE
event come over the socket, and it does contain variable_sip_gateway_name
with the actual name of the gateway, however I can't devise a way to access
that data using the org.freeswitch.esl.client library, and even if I could,
I still don't want all events for this system.

Is it possible to get this information returned in any meaningful way
through the ESL layer, either by an api command to

Strange issue with late negotiation

Flavio Goncalves — 2013-05-23T20:38:17

Hello,

I'm having a really strange issue on billing related to late negotiation.
The call end up hanged with no BYE.

There is a dialer sending thousands of calls through and OpenSIPS to a
termination based on FreeSwitch. I have a SIP call flow as below. Some
times during the day, when the volume is high, the UAC drops some 200Ok
 and FS send a reinvite in the middle of the Initial transaction.

----INVITE ---------------   Proxy --INVITE ---------------> FS
<----200 OK --------------   Proxy <---200 OK- ------------- FS
<--REINVITE---------------   Proxy <-REINVITE--------------- FS
---481 leg does not exit->   Proxy ---481------------------> FS
<-- ACK (REINVITE)--------   Proxy <-ACK(REINVITE)---------- FS
----CANCEL --------------->  Proxy ---CANCEL --------------> FS
<---200 Ok ----------------  Proxy <----200 Ok ------------- FS

FS is sending a reinvite before ACK comes from the client.

I have two questions:

1) Is it valid to send a reinvite in the middle of an existing transaction?

According to t

Dialplan not executing on continue_on_fail=true

Ashish gautam — 2013-05-24T12:17:39

Hi,

I have a dialplan as follows:

<include>
<extension name="public_did">
<condition field="destination_number" expression="^(47673501)$">
<action application="answer"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="api_hangup_hook=perl hook.pl"/>
<action application="set" data="session_in_hangup_hook=true"/>
<action application="perl" data="perl/ash.pl"/>
</condition>
</extension>
</include>

when the called party does not pick up the phone or is busy, the dialplan
does not proceed and hook.pl does not get executed.

Please help

problem bridging lots of dtmf in a session

Michel Brabants — 2013-05-24T11:53:41

Hello,

we have a problem with dtmf not being bridged by freeswitch when there is
lots of dtmf in the call. Freeswitch resides between an asterisk gateway
(which sends lot of duplicated dtmf) and an ivr. When there is a lot of
dtmf being bridged in a single call, we can see in a wireshark-trace that
it stops bridging the dtmf at a certain moment. I can still see the dtmf
being received in the freeswitch-logs, bit I'm not seeing it in the
wireshark-trace towards the ivr.

We're using freeswitch 1.2.7.

Thanks,

Michel
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Best performing codec on mobile/slow networks

Daniel Ivanov — 2013-05-24T09:50:36

I've officially given up on AMR, due to it's licensing issues and sketchy
supports for my mobile clients. I am looking for new options now as
CsipSimple has a new codec pack supporting G.726.1, opus and codec2. Which
do you think would perform best on mobile networks? And which has FS
support for transcoding as well?
_________________________________________________________________________
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Centralized SIP directory

mehroz — 2013-05-24T09:08:22

Hi all,

I have a DB server, storing SIP users and auth information.

2 freeswitch servers configured with mod_xml_curl , and an attempt of
registration on both servers looks for SIP authentication from DB server. At
the moment, i am defining 2 domains on DB server (FS-1 IP and FS-2 IP), and
creating unique user , 2 times to address both domains.

I want to get rid of creating 2 users for both domains. What is the best
approach.




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_________________________________________________________________________
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http://www.cudatel.com

Official FreeSWITCH Sites
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http://wiki.freeswitch.org
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FreeS

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

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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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

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"

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

The number of registered phones

andpe — 2013-05-23T08:17:19

Hi
What is the possible number of registered phones to FreeSWITCH? I know it may depend on many factors. Where can I find this information? Is 2000 or 3000 users a lot or not? I'm not asking about the performance of connections here.
Regards
Andy_________________________________________________________________________
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consulting-YF8E+gPBBv73h3GqohbjpQ< at >public.gmane.org
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Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg

Adam Raszynski — 2013-05-23T08:11:57

Hi,

I have very strange problem with audio after upgrading of FS to the newest
GIT version.

Previous version was quite old, but worked without problems for a long time:
FreeSWITCH version: 1.0.head (git-313b164 2011-11-26 08-53-01 -0600)

Current version is:
FreeSWITCH version: 1.5.1b ()

All configuration files are the same (exact copy)

System: Debian Lenny 32-bit
All calls are made over SIP

Symptoms:
- IVR prompt playback is OK
- FAX sending/receiving is OK
- Calling to remote servers via VoIP provider is OK
- The only problem is calling to local user and it's very strange. When
external user calls local user (SIP call, via IVR, then bridge), audio
stream from external user is OK, but audio stream from local user sent to
external user is very delayed. Delay increases during a call. At the
beginning is about 1 second, all the time delay is increasing. Audio is
reaching external user but after about 30 seconds of call stream is delayed
for about 10 seconds. At the same time audio stream from external use

FreeSwitch Proxy + RTPProxy Media server

adahary — 2013-05-22T20:21:01

I would like to integrate FS with RTPproxy like openSips and kamailio are
well integrated with it.
FS should handle the SIP signaling and the RTPproxy should relay the RTP
stream from A to B:
A.sip <=> FS <=>  B.sip   
FS = PASS-THRU
A.rtp <=> RTPproxy <=> B.rtp.

I understand that FS should ask the RTPproxy to allocate UDP ports for both
endpoint and then pass-thru-bridge them to cummunicate directly through the
RTPproxy.

What I cannot yet figure out is how replace both A and B ip/port sets in SDP
with the RTPproxy ip/port-udp.
I'v read about switch_r_sdp (Leg.A) and switch_m_sdp (Leg.B !?) but couldn't
figure it out.

how should I use switch_r_sdp (or/and switch_m_sdp) in my dialplan when when
A calls B?
Tried the following with no help.
<action application="set">
</action>
<action application="set">
</action>

Any tip/help/advise will be appriciated.

Regards

Assaf




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