Sergey Malykhin | 25 Jun 09:42 2010
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SIP Proxy

Hello,

Is it possible to use SipX for creation Windows SIP Proxy? If yes can you specify which libraries I have to research, because now I'm researching sipXtapi project in MSVC8, but have no idea what to do with it :) and if it usable for SIP Proxy...

TIA

Sergey

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Filipe Paredes | 25 Feb 21:07 2010
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Remote_Held Call State when answering a call

Hi,

We recently found a problem while developing a sipxtapi softphone.  

After answering an incoming call, the Call State changes to REMOTE_HELD instead of CONNECTED. But this only happens sometimes! After a couple of hours it returns to normal.
I didn't find a pattern to why this happens sometimes.

I'm using sipXtapi 3.2 and SipXecs 4.0.2

Thanks in advanced.
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pedro2263 | 10 Feb 18:37 2010
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ACD with G.729

Hello,

I'm trying to use the G.729 codec with the ACD, I apply the XCL-130 patch but it doesn't work, does any body know how to include this codec for use it in sipxacd?

Thanks in advance.

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Manuel Pimenta | 9 Feb 11:06 2010
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Building sipXtapi-media-update branch for Windows

Hey!

 

I have been having loads of issues when trying to compile the media-update branch for windows.

I have installed all of the referenced dependencies, including FFmpeg, but there are both issues with missing files in the mediaLib project and VideoSupport project, as well as compiler errors when interpreting some of the files that have inline assembly code.

I see loads of people posting here that they have managed to build this media-update branch, but I just can’t seem to do it.

 

Could someone please post a list of every dependency that was required to install, including the version? (FFmpeg for example updated its source code, so a lot of the code used in sipXtapi is deprecated…)

 

Thanks in advance,

 

Manuel Pimenta

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birchstreet | 19 Jan 01:02 2010
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sipx expert wanted

Hi there,

I have a client that is looking to roll out a SipX in a hosted configuration. They are looking for a SipX expert to ride shot gun and provide services and development assistance. I am hoping to hear from a few of you who feel you have strong experience with SipX that could fit this bill.

Please email me directly to review.

Bob Smith
birchstreet <at> gmail.com







<div><p>Hi there,<br><br>I have a client that is looking to roll out a SipX in a hosted configuration. They are looking for a SipX expert to ride shot gun and provide services and development assistance. I am hoping to hear from a few of you who feel you have strong experience with SipX that could fit this bill.<br><br>Please email me directly to review.<br><br>Bob Smith<br><a href="mailto:birchstreet <at> gmail.com">birchstreet <at> gmail.com</a><br><br><br><br><br><br><br><br></p></div>
Marcin Głowacki | 11 Jan 11:50 2010
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sipxAudioSetVolume() and sipxAudioSetCallInputDevice()

Hi,

I am wondering if you experience problems with these two functions: sipxAudioSetVolume() and sipxAudioSetCallInputDevice().

I use version 3.2 of sipXtapi in my project and I run it on Windows 7. When I try to set call input device the method returns SIPX_RESULT_INVALID_ARGS although I supply device name retrieved with sipxAudioGetInputDevice(). This also happens on XP SP3.

When I changed from XP to 7 sipxAudioSetVolume() stops at this: assert(iVolume == iLevel). On XP it worked fine.

Do you have any suggestions?

Marcin
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Jan Fricke | 22 Dec 10:49 2009
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Presence problem second notify answered with 481

Hi members,
hope someone can help me.
My setup:
SipX 4.0.1
sipXtapi 3.3 (SVN Rev 11467) compiled with Visual Studio

I try to subscribe presence states of some phones by subscribing the
resourcelist at our sipx.
The subscription is answered with an 202 followed by a Notify message.
The second Notify message from the presence server is answered by
sipxtapi with 481 Transaction does not exist.
Below you find the subscribe and notify messages.

What does transaction does not exist mean? The second notify belongs to
the dialog initiated by the subscribe message (same call-id same from
and to header tags). The CSeq is increased by one and a new via branch
was created. So I see no problem. Same dialog, new transaction.

Thanks in advance

Jan

SUBSCRIBE sip:~~rl~F~7800 <at> voip.mydomain.de SIP/2.0
From: <sip:7800 <at> voip.mydomain.de>;tag=35a7b2be8686cf178af07ddbcd51b507
To: <sip:~~rl~F~7800 <at> voip.mydomain.de>
Call-Id:
ssc-7ab8c84ac338b5ab85e57d567839783c <at> 28b70d10c2d661262affe918ab9e2955
Cseq: 2 SUBSCRIBE
Contact: sip:7800 <at> 192.168.1.99:62621
Event: dialog
Accept: application/dialog-info+xml,multipart/related
Expires: 3600
Date: Tue, 22 Dec 2009 09:23:09 GMT
Max-Forwards: 70
User-Agent: sipX/1.3.0 (WinNT)
Supported: replaces, eventlist
Authorization: Digest username="7800", realm="voip.mydomain.de",
nonce="a69e3ede32ff3025ce482637b2f173ed4b308fb3",
uri="sip:~~rl~F~7800 <at> voip.mydomain.de",
response="895069c140cb7986169f703e06d72faa"
Via: SIP/2.0/UDP 192.168.1.99:62621;branch=z9hG4bK-771e24c176d6
Content-Length: 0

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

NOTIFY sip:7800 <at> 192.168.1.99:62621;x-sipX-nonat SIP/2.0
From: <sip:~~rl~F~7800 <at> voip.mydomain.de>;tag=1939c44f
To: <sip:7800 <at> voip.mydomain.de>;tag=35a7b2be8686cf178af07ddbcd51b507
Cseq: 1 NOTIFY
Call-Id:
ssc-7ab8c84ac338b5ab85e57d567839783c <at> 28b70d10c2d661262affe918ab9e2955
Event: dialog
Subscription-State: active;expires=3029
Content-Type:
multipart/related;type="application/rlmi+xml";start="<rlmi <at> voip.mydomain.de>";boundary="b88b51ab"
Contact: <sip:~~rl~F~7800 <at> 192.168.1.1:5140;x-sipX-nonat>
Date: Tue, 22 Dec 2009 09:21:55 GMT
Max-Forwards: 19
User-Agent: sipXecs/4.0.1 sipXecs/rls (Linux)
Accept-Language: en
Require: eventlist
Via: SIP/2.0/UDP
192.168.1.1;branch=z9hG4bK-sipXecs-556a4c572fd52534e95f061c79bd08e9eb14
Via: SIP/2.0/TCP
192.168.1.1:5140;branch=z9hG4bK-sipXecs-64f1e3211777a98386e7d0a930059f81e69a
Content-Length: 1270

--b88b51ab
CONTENT-ID: <0 <at> voip.mydomain.de>
CONTENT-TYPE: application/dialog-info+xml
CONTENT-TRANSFER-ENCODING: binary

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="26"
state="partial" entity="sip:7202 <at> voip.mydomain.de">
<dialog id="id8a7649db"
call-id="c50bf063-a5c70a12-84f25805 <at> 192.168.1.111"
local-tag="3768BB0E-B7DD9C6D" remote-tag="C6519D2C-BFAB40B7"
direction="recipient">
<state>terminated</state>
<local>
<target uri="sip:7202 <at> voip.mydomain.de">
<param pname="x-line-id" pval="0" />
</target>
</local>
<remote>
<identity display="User ext7200">sip:7200 <at> voip.mydomain.de</identity>
<target uri="sip:7200 <at> 192.168.1.111">
<param pname="x-sipX-nonat" pval="" />
</target>
</remote>
</dialog>
</dialog-info>
--b88b51ab
CONTENT-ID: <rlmi <at> voip.mydomain.de>
CONTENT-TYPE: application/rlmi+xml
CONTENT-TRANSFER-ENCODING: binary

<?xml version="1.0"?>
<list xmlns="urn:ietf:params:xml:ns:rlmi"
uri="sip:~~rl~F~7800 <at> voip.mydomain.de" version="0" fullState="true">
  <resource uri="sip:7202 <at> voip.mydomain.de">
    <name>test</name>
    <instance id="s-4374e03528cdbbb1-258,1f60c3ad,2652880-28DC4BE7"
state="active" cid="0 <at> voip.mydomain.de"/>
  </resource>
</list>

--b88b51ab--

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

SIP/2.0 200 OK
From: <sip:~~rl~F~7800 <at> voip.mydomain.de>;tag=1939c44f
To: <sip:7800 <at> voip.mydomain.de>;tag=35a7b2be8686cf178af07ddbcd51b507
Call-Id:
ssc-7ab8c84ac338b5ab85e57d567839783c <at> 28b70d10c2d661262affe918ab9e2955
Cseq: 1 NOTIFY
Via: SIP/2.0/UDP
192.168.1.1;branch=z9hG4bK-sipXecs-556a4c572fd52534e95f061c79bd08e9eb14
Via: SIP/2.0/TCP
192.168.1.1:5140;branch=z9hG4bK-sipXecs-64f1e3211777a98386e7d0a930059f81e69a
Date: Tue, 22 Dec 2009 09:23:09 GMT
Contact: sip:192.168.1.99:62621
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, REGISTER,
SUBSCRIBE, NOTIFY
User-Agent: sipX/1.3.0 (WinNT)
Content-Length: 0

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

NOTIFY sip:7800 <at> 192.168.1.99:62621;x-sipX-nonat SIP/2.0
From: <sip:~~rl~F~7800 <at> voip.mydomain.de>;tag=1939c44f
To: <sip:7800 <at> voip.mydomain.de>;tag=35a7b2be8686cf178af07ddbcd51b507
Cseq: 2 NOTIFY
Call-Id:
ssc-7ab8c84ac338b5ab85e57d567839783c <at> 28b70d10c2d661262affe918ab9e2955
Event: dialog
Subscription-State: active;expires=3020
Content-Type:
multipart/related;type="application/rlmi+xml";start="<rlmi <at> voip.mydomain.de>";boundary="b00a5834"
Contact: <sip:~~rl~F~7800 <at> 192.168.1.1:5140;x-sipX-nonat>
Date: Tue, 22 Dec 2009 09:22:04 GMT
Max-Forwards: 19
User-Agent: sipXecs/4.0.1 sipXecs/rls (Linux)
Accept-Language: en
Require: eventlist
Via: SIP/2.0/UDP
192.168.1.1;branch=z9hG4bK-sipXecs-557da0dfb2e8307a7d1b0e75bbb6578cdcee
Via: SIP/2.0/TCP
192.168.1.1:5140;branch=z9hG4bK-sipXecs-64f447b8e1acd8677beea39556a8880272ec
Content-Length: 1266

--b00a5834
CONTENT-ID: <0 <at> voip.mydomain.de>
CONTENT-TYPE: application/dialog-info+xml
CONTENT-TRANSFER-ENCODING: binary

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="27"
state="partial" entity="sip:7202 <at> voip.mydomain.de">
<dialog id="id9af4cf16"
call-id="7b9035ef-2fafb56e-1703e1f1 <at> 192.168.1.111"
local-tag="6053B995-3B3FB6F0" remote-tag="74365B68-DB3CF203"
direction="recipient">
<state>early</state>
<local>
<target uri="sip:7202 <at> voip.mydomain.de">
<param pname="x-line-id" pval="0" />
</target>
</local>
<remote>
<identity display="User ext7200">sip:7200 <at> voip.mydomain.de</identity>
<target uri="sip:7200 <at> 192.168.1.111">
<param pname="x-sipX-nonat" pval="" />
</target>
</remote>
</dialog>
</dialog-info>
--b00a5834
CONTENT-ID: <rlmi <at> voip.mydomain.de>
CONTENT-TYPE: application/rlmi+xml
CONTENT-TRANSFER-ENCODING: binary

<?xml version="1.0"?>
<list xmlns="urn:ietf:params:xml:ns:rlmi"
uri="sip:~~rl~F~7800 <at> voip.mydomain.de" version="1" fullState="false">
  <resource uri="sip:7202 <at> voip.mydomain.de">
    <name>test</name>
    <instance id="s-4374e03528cdbbb1-258,1f60c3ad,2652880-28DC4BE7"
state="active" cid="0 <at> voip.mydomain.de"/>
  </resource>
</list>

--b00a5834--

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

SIP/2.0 481 Transaction Does Not Exist
From: <sip:~~rl~F~7800 <at> voip.mydomain.de>;tag=1939c44f
To: <sip:7800 <at> voip.mydomain.de>;tag=35a7b2be8686cf178af07ddbcd51b507
Call-Id:
ssc-7ab8c84ac338b5ab85e57d567839783c <at> 28b70d10c2d661262affe918ab9e2955
Cseq: 2 NOTIFY
Via: SIP/2.0/UDP
192.168.1.1;branch=z9hG4bK-sipXecs-557da0dfb2e8307a7d1b0e75bbb6578cdcee
Via: SIP/2.0/TCP
192.168.1.1:5140;branch=z9hG4bK-sipXecs-64f447b8e1acd8677beea39556a8880272ec
Date: Tue, 22 Dec 2009 09:23:18 GMT
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, REGISTER,
SUBSCRIBE, NOTIFY
User-Agent: sipX/1.3.0 (WinNT)
Content-Length: 0

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Anup | 18 Dec 12:46 2009

How to play the RTP packets as soon as it comes to the UDP buffer.

Consider the following scenario: As soon as a rtp packet arrives, i want to decode and play the Rtp packets using waveoutwrite().
Assume the voice data is encoded using Speex(narrow band). Can anyone help me with what code i must include to the get the decoding done.
 
Thanks
Anup
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A S | 7 Dec 18:01 2009
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Conference with one-way audio?

Greetings,
 
found SipXtapi recently...trying to find my way around it. Please bear with me...
 
I am trying to create a special "conference" where an audio from callers is mixed at the receiving end (the host that creates a conference), but not sent back to the callers. Tried doing it with multiple non-conferenced calls and the MpMediaTask, but per documentation, only one resource graph is processed at a time.
 
Also tried disabling the ToSpeaker resources in the call flow graphs, but this seems to stop both incoming and outgoing video.
 
Is there a way to set up this "one-way conference" with SipXtapi? Should I be trying to somehow connect the flow graphs of the conferenced calls and disable the resources that are responsible for sending audio back to conference participants (how?)
 
Help much appreciated.
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Paulo Vicentini | 26 Nov 21:32 2009
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[sipxtapi] QOP - Quality of Protection

Hi 
Do you know if sipxtapi (sipXtackLib) is compatible (workable) with qop (Quality of Protection)?
It seems that HttpMessage::buildMd5Digest is considering qop parameter..but I need  to check what is happening.
Regards
Paulo
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Jan Thiemo Fricke | 14 Nov 01:10 2009
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Problems with some end devices and sipxtapi

Hi all,
recently I've noticed some problems between sipxtapi and a couple of phones.
First some details of my test installation:

sipxtapi 3.3 (2.11.2009) with a self written wrapper class for .NET
Internal samplerate set to 16kHz. Codecs G711/G722
Phones: Polycom 430/550, Siemens Optipoint 410 all phones with latest
firmware
SIP Server: SipX 4.0.2

SipXtapi crashes at calls with the Polycom 430/550 if I press hold at
the polycom. Sometimes I can hear music-on-hold for a very short time
before it crashes. With Snom, Grandstream, Patton, sipxtapi on the other
side everything works fine.

Another problem is an incompatiblity with Siemens Optipoint 410. With a
old firmware calling each other was not possible. With a new firmware
sipxtapi can call the Siemens and backwards. But sipxtapi seems to send
the BYE message with a request header like
1234 <at> ip-of-the-siemens-telephone and the siemens answers with 404 Not found.

Has anybody experienced such problems?

I will create some log-files of these things and post which version and
firmware version of each device is used. But maybe there is someone who
has done this before because he had these problems too.

Thanks in advance and best regards

Jan
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Gmane