date:20150310

Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

2015-03-10 Thread mahan77

Hello Terrance,

Thank you for your time to replay back. 
It was basic dispatcher Config file and posted in the first place. 

This is my sip trace.

interface: eth0 (192.168.1.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )

U 192.168.1.64:5060 -> 192.168.1.150:5060
.
Gb.."
?.

U 87.117.75.1:5060 -> 192.168.1.150:5060
INVITE sip:442032912345@192.168.1.150:5060 SIP/2.0.
Via: SIP/2.0/UDP 87.117.75.1:5060;rport;branch=z9hG4bK9gZ0vUKSc0Q3m.
Max-Forwards: 68.
From: "07720212345" ;tag=1Z1Z5983N78ZN.
To: .
Call-ID: fe855718-40e4-1233-4c90-aba651435a79.
CSeq: 72611343 INVITE.
Contact: .
User-Agent: TelNG GW.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Session-Expires: 1800;refresher=uac.
Min-SE: 120.
Privacy: none.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 225.
X-3C-ACCOUNT: 8166.
X-3C-DIRECTION: in.
X-FS-Support: update_display,send_info.
P-Asserted-Identity: "07720212345" .
.
v=0.
o=FreeSWITCH 1425866039 1425866040 IN IP4 87.117.75.1.
s=FreeSWITCH.
c=IN IP4 87.117.75.1.
t=0 0.
m=audio 28776 RTP/AVP 8 0 18 101 13.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 192.168.1.150:5060 -> 192.168.1.85:5060
INVITE sip:442032912345@192.168.1.150:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK9gZ0vUKSc0Q3m.
Via: SIP/2.0/UDP
87.117.75.1:5060;received=87.117.75.1;rport=5060;branch=z9hG4bK9gZ0vUKSc0Q3m.
Max-Forwards: 67.
From: "07720212345" ;tag=1Z1Z5983N78ZN.
To: .
Call-ID: fe855718-40e4-1233-4c90-aba651435a79.
CSeq: 72611343 INVITE.
Contact: .
User-Agent: TelNG GW.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Session-Expires: 1800;refresher=uac.
Min-SE: 120.
Privacy: none.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 225.
X-3C-ACCOUNT: 8166.
X-3C-DIRECTION: in.
X-FS-Support: update_display,send_info.
P-Asserted-Identity: "07720212345" .
.
v=0.
o=FreeSWITCH 1425866039 1425866040 IN IP4 87.117.75.1.
s=FreeSWITCH.
c=IN IP4 87.117.75.1.
t=0 0.
m=audio 28776 RTP/AVP 8 0 18 101 13.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 192.168.1.85:5060 -> 192.168.1.150:5060
OPTIONS sip:2001@192.168.1.64:5060;ob SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK2460f825;rport.
Max-Forwards: 70.
From: "asterisk" ;tag=as2b561864.
To: .
Contact: .
Call-ID: 6432c1a87201b6d9051df2e71c82e6a7@192.168.1.85:5060.
CSeq: 102 OPTIONS.
User-Agent: Asterisk PBX SVN-branch-13-r432059.
Date: Mon, 09 Mar 2015 09:53:35 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces, timer.
Content-Length: 0.
.


U 192.168.1.85:5060 -> 192.168.1.150:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
192.168.1.150:5060;branch=z9hG4bK9gZ0vUKSc0Q3m;received=192.168.1.150;rport=5060.
Via: SIP/2.0/UDP
87.117.75.1:5060;received=87.117.75.1;rport=5060;branch=z9hG4bK9gZ0vUKSc0Q3m.
From: "07720212345" ;tag=1Z1Z5983N78ZN.
To: ;tag=as63edc664.
Call-ID: fe855718-40e4-1233-4c90-aba651435a79.
CSeq: 72611343 INVITE.
Server: Asterisk PBX SVN-branch-13-r432059.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces, timer.
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c8254d0".
Content-Length: 0.
.


U 192.168.1.150:5060 -> 192.168.1.85:5060
OPTIONS sip:2001@192.168.1.64:5060;ob SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK2460f825.
Via: SIP/2.0/UDP
192.168.1.85:5060;received=192.168.1.85;branch=z9hG4bK2460f825;rport=5060.
Max-Forwards: 69.
From: "asterisk" ;tag=as2b561864.
To: .
Contact: .
Call-ID: 6432c1a87201b6d9051df2e71c82e6a7@192.168.1.85:5060.
CSeq: 102 OPTIONS.
User-Agent: Asterisk PBX SVN-branch-13-r432059.
Date: Mon, 09 Mar 2015 09:53:35 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces, timer.
Content-Length: 0.
.


U 192.168.1.150:5060 -> 87.117.75.1:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
87.117.75.1:5060;received=87.117.75.1;rport=5060;branch=z9hG4bK9gZ0vUKSc0Q3m.
From: "07720212345" ;tag=1Z1Z5983N78ZN.
To: ;tag=as63edc664.
Call-ID: fe855718-40e4-1233-4c90-aba651435a79.
CSeq: 72611343 INVITE.
Server: Asterisk PBX SVN-branch-13-r432059.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces, timer.
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c8254d0".
Content-

[OpenSIPS-Users] dialog errors

2015-03-10 Thread Edwin

A few times a day I see dialog warnings in my logging:

Mar 9 08:00:04 [18379]: WARNING:dialog:w_unset_dlg_profile: cannot get
string for value
Mar 9 08:00:05 [18385]: WARNING:dialog:w_unset_dlg_profile: cannot get
string for value
Mar 9 08:00:07 [18391]: WARNING:dialog:w_unset_dlg_profile: cannot get
string for value
Mar 9 08:05:18 [18374]: WARNING:dialog:w_unset_dlg_profile: cannot get
string for value
Mar 9 08:05:19 [18391]: WARNING:dialog:w_unset_dlg_profile: cannot get
string for value
Mar 9 08:05:21 [18390]: WARNING:dialog:w_unset_dlg_profile: cannot get
string for value

I can not find more information about this errors in the opensips
documentation or on the mailing lists. Can anyone give me a hint what could
be wrong or where to look? (I can give more info about database or config).





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View this message in context: 
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Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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[OpenSIPS-Users] Blox SBC

2015-03-10 Thread Varadhan Work

Hello,

I'm Varadhan,

We have come up with an idea to build opensource SBC with opensips as core
SIP session route, thanks to opensips and its community.

Now we have launched beta version of Blox (www.blox.org)

Blox is a Session Border Controller(SBC) used to control VoIP signaling and
media streams. SBC is responsible for setting up, conducting, and tearing
down calls. SBC allows owners to control the types of call that can be
placed through the networks and also overcome some of the problems caused
by firewalls and NAT for VoIP calls. A common location for a stand-alone
SBC is a connection point, called a border, between a private local area
network (LAN) and the Internet. SBC polices real-time voice traffic between
IP network borders ensuring your private network is robustly secure and
fully manageable.
You can download and install Blox via www.blox.org, please post your
valuable feedback.

Thanks & Regards,
Varadhan M
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[OpenSIPS-Users] How to remove/update dialog

2015-03-10 Thread Hamid Hashmi


route[sip]{...t_on_failure("1");$DLG_timeout = 120;
create_dialog("B"); t_relay();}
failure_route[1]{... if(t_check_status("some reasson")){   
route(pstn);  }...}
route[pstn]{...t_on_failure("2");$DLG_timeout = 60;
create_dialog("B");t_relay();}
How to remove previous dialog from table "dialog" ? OR is there any method to 
update the timeout value in dialog without calling create_dialog("B") ?
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Re: [OpenSIPS-Users] Blox SBC

2015-03-10 Thread Terrance Devor

Hello,

Just visited your site and say that you can only download the .iso.
Do you make the full source available?


Terrance
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Re: [OpenSIPS-Users] Call transfer problem

2015-03-10 Thread Schneur Rosenberg

Sorry I meant to place call on hold, not transfer

On Mon, Mar 9, 2015 at 8:40 PM, Schneur Rosenberg 
wrote:

> I have a OpenSIPS server that acts as a load balancing server for a couple
> of asterisk servers, and lately I'm having a issue with asterisk responding
> with a 488 Not acceptable here for transfer requests when it comes to port
> 5060 on the Opensips server, but if I send it to another port it works
> fine, I realized that when using port 5060 the sdp has port 0  ie  (m=audio
> 0 RTP/AVP 0 8 101.) but when using another port it does have a port number
> (ie m=audio 16422 RTP/AVP 0 8 101.)
>
> Im attaching both traces, I changed ips for security 212.212.212.212. is
> the opensips server, 212.212.212.213 is the asterisk server,
> and 91.176.221.245 is the end user.
>
> Can anyone please help me make this work on port 5060 too, and can you
> please explain why it would act differently, the NAT is also handeled
> differently as can be seen in the VIA header (maybe router is trying to use
> some ALG?)
>
> Here is the trace using port 5060
>
> U 91.176.221.245:49242 -> 212.212.212.212:5060
> INVITE sip:1917425@212.212.212.213:5060;nat=yes SIP/2.0.
> Via: SIP/2.0/UDP 91.176.221.245:32781;branch=z9hG4bK-a6635dda.
> From: "EXT 101" ;tag=3189bbd5cf53a984o0.
> To: ;tag=as7f636018.
> Call-ID: f8a392e9-f1139d80@192.168.0.101.
> CSeq: 103 INVITE.
> Max-Forwards: 70.
> Route:  ;lr;ftag=3189bbd5cf53a984o0;did=d97.6c4d2422>.
> Proxy-Authorization: Digest
>
>
> username="WindowP1",realm="myserver",nonce="54f468a5000180cd88bd6cead37ab9d1920cc4c54a0ecdd3",uri="sip:1917425@212.212.
>
> 212.213:5060",algorithm=MD5,response="0a5a118e2b318621f32b5d60b600229c".
> Contact: "EXT 101" .
> Expires: 30.
> User-Agent: Cisco/SPA525G2-7.5.6.
> Content-Length: 226.
> Content-Type: application/sdp.
> .
> v=0.
> o=- 33523926 33523927 IN IP4 192.168.0.101.
> s=-.
> c=IN IP4 0.0.0.0.
> t=0 0.
> m=audio 0 RTP/AVP 0 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:30.
> a=sendonly.
>
> U 212.212.212.212:5060 -> 91.176.221.245:49242
> SIP/2.0 100 Giving a try.
> Via: SIP/2.0/UDP 91.176.221.245:32781
> ;received=91.176.221.245;rport=49242;branch=z9hG4bK-a6635dda.
> From: "EXT 101" ;tag=3189bbd5cf53a984o0.
> To: ;tag=as7f636018.
> Call-ID: f8a392e9-f1139d80@192.168.0.101.
> CSeq: 103 INVITE.
> Server: OpenSIPS (1.7.2-notls (x86_64/linux)).
> Content-Length: 0.
> .
>
> U 212.212.212.212:5060 -> 212.212.212.213:5060
> INVITE sip:1917425@212.212.212.213:5060 SIP/2.0.
> Record-Route: .
> Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe66e.0702ec92.0.
> Via: SIP/2.0/UDP 91.176.221.245:32781
> ;rport=49242;received=91.176.221.245;branch=z9hG4bK-a6635dda.
> From: "EXT 101" ;tag=3189bbd5cf53a984o0.
> To: ;tag=as7f636018.
> Call-ID: f8a392e9-f1139d80@192.168.0.101.
> CSeq: 103 INVITE.
> Max-Forwards: 69.
> Proxy-Authorization: Digest
>
>
> username="WindowP1",realm="myserver",nonce="54f468a5000180cd88bd6cead37ab9d1920cc4c54a0ecdd3",uri="sip:1917425@212.212.
>
> 212.213:5060",algorithm=MD5,response="0a5a118e2b318621f32b5d60b600229c".
> Contact: "EXT 101" .
> Expires: 30.
> User-Agent: Cisco/SPA525G2-7.5.6.
> Content-Length: 226.
> Content-Type: application/sdp.
> .
> v=0.
> o=- 33523926 33523927 IN IP4 192.168.0.101.
> s=-.
> c=IN IP4 0.0.0.0.
> t=0 0.
> m=audio 0 RTP/AVP 0 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:30.
> a=sendonly.
>
> U 212.212.212.213:5060 -> 212.212.212.212:5060
> SIP/2.0 488 Not acceptable here.
> Via: SIP/2.0/UDP
> 212.212.212.212;branch=z9hG4bKe66e.0702ec92.0;received=212.212.212.212;rport=5060.
> Via: SIP/2.0/UDP 91.176.221.245:32781
> ;rport=49242;received=91.176.221.245;branch=z9hG4bK-a6635dda.
> From: "EXT 101" ;tag=3189bbd5cf53a984o0.
> To: ;tag=as7f636018.
> Call-ID: f8a392e9-f1139d80@192.168.0.101.
> CSeq: 103 INVITE.
> Server: SIP Server 9.21/CS.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH.
> Supported: replaces, timer.
> X-Asterisk-HangupCause: Normal Clearing.
> X-Asterisk-HangupCauseCode: 16.
> Content-Length: 0.
> .
> here is the trace using port 5744
> U 91.176.221.245:63719 -> 212.212.212.212:5744
> INVITE sip:1917425@212.212.212.213:5060;nat=yes SIP/2.0.
> Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-6f0161f0.
> From: "EXT 101" ;tag=f8c2b14a55549ac2o0.
> To: ;tag=as1fcb82f4.
> Call-ID: 5832b9ea-a4a32aa2@192.168.0.101.
> CSeq: 103 INVITE.
> Max-Forwards: 70.
> Route:  ;lr;ftag=f8c2b14a55549ac2o0;did=9ae.775cc5c5>.
> Proxy-Authorization: Digest
> username="WindowP1",realm="myserver",nonce="54f46d7a081272baa39f9e34a0ee81f471dec7b034ba",uri="
> sip:1917425@212.212.212.213:5060
> ",algorithm=MD5,response="4096f485853bf6a209424c36dac2342b".
> Contact: "EXT 101" .
> Expires: 30.
> User-Agent: Cisco/SPA525G2-7.5.6.
> Content-Length: 224.
> Content-Type: application/sdp.
> .
> v=0.
> o=- 30981

[OpenSIPS-Users] Problem with placing call on hold

2015-03-10 Thread Schneur Rosenberg

I have a OpenSIPS server that acts as a load balancing server for a couple
of asterisk servers, and lately I'm having a issue with asterisk responding
with a 488 Not acceptable here for hold requests when it comes to port 5060
on the Opensips server, but if I send it to another port it works fine, I
realized that when using port 5060 the sdp has port 0  ie  (m=audio 0
RTP/AVP 0 8 101.) but when using another port it does have a port number
(ie m=audio 16422 RTP/AVP 0 8 101.)

Im attaching both traces, I changed ips for security 212.212.212.212. is
the opensips server, 212.212.212.213 is the asterisk server,
and 91.176.221.245 is the end user.

Can anyone please help me make this work on port 5060 too, and can you
please explain why it would act differently, the NAT is also handeled
differently as can be seen in the VIA header (maybe router is trying to use
some ALG?)

Here is the trace using port 5060

U 91.176.221.245:49242 -> 212.212.212.212:5060
INVITE sip:1917425@212.212.212.213:5060;nat=yes SIP/2.0.
Via: SIP/2.0/UDP 91.176.221.245:32781;branch=z9hG4bK-a6635dda.
From: "EXT 101" ;tag=3189bbd5cf53a984o0.
To: ;tag=as7f636018.
Call-ID: f8a392e9-f1139d80@192.168.0.101.
CSeq: 103 INVITE.
Max-Forwards: 70.
Route: .
Proxy-Authorization: Digest

username="WindowP1",realm="myserver",nonce="54f468a5000180cd88bd6cead37ab9d1920cc4c54a0ecdd3",uri="sip:1917425@212.212.

212.213:5060",algorithm=MD5,response="0a5a118e2b318621f32b5d60b600229c".
Contact: "EXT 101" .
Expires: 30.
User-Agent: Cisco/SPA525G2-7.5.6.
Content-Length: 226.
Content-Type: application/sdp.
.
v=0.
o=- 33523926 33523927 IN IP4 192.168.0.101.
s=-.
c=IN IP4 0.0.0.0.
t=0 0.
m=audio 0 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendonly.

U 212.212.212.212:5060 -> 91.176.221.245:49242
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 91.176.221.245:32781
;received=91.176.221.245;rport=49242;branch=z9hG4bK-a6635dda.
From: "EXT 101" ;tag=3189bbd5cf53a984o0.
To: ;tag=as7f636018.
Call-ID: f8a392e9-f1139d80@192.168.0.101.
CSeq: 103 INVITE.
Server: OpenSIPS (1.7.2-notls (x86_64/linux)).
Content-Length: 0.
.

U 212.212.212.212:5060 -> 212.212.212.213:5060
INVITE sip:1917425@212.212.212.213:5060 SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe66e.0702ec92.0.
Via: SIP/2.0/UDP 91.176.221.245:32781
;rport=49242;received=91.176.221.245;branch=z9hG4bK-a6635dda.
From: "EXT 101" ;tag=3189bbd5cf53a984o0.
To: ;tag=as7f636018.
Call-ID: f8a392e9-f1139d80@192.168.0.101.
CSeq: 103 INVITE.
Max-Forwards: 69.
Proxy-Authorization: Digest

username="WindowP1",realm="myserver",nonce="54f468a5000180cd88bd6cead37ab9d1920cc4c54a0ecdd3",uri="sip:1917425@212.212.

212.213:5060",algorithm=MD5,response="0a5a118e2b318621f32b5d60b600229c".
Contact: "EXT 101" .
Expires: 30.
User-Agent: Cisco/SPA525G2-7.5.6.
Content-Length: 226.
Content-Type: application/sdp.
.
v=0.
o=- 33523926 33523927 IN IP4 192.168.0.101.
s=-.
c=IN IP4 0.0.0.0.
t=0 0.
m=audio 0 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendonly.

U 212.212.212.213:5060 -> 212.212.212.212:5060
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/UDP
212.212.212.212;branch=z9hG4bKe66e.0702ec92.0;received=212.212.212.212;rport=5060.
Via: SIP/2.0/UDP 91.176.221.245:32781
;rport=49242;received=91.176.221.245;branch=z9hG4bK-a6635dda.
From: "EXT 101" ;tag=3189bbd5cf53a984o0.
To: ;tag=as7f636018.
Call-ID: f8a392e9-f1139d80@192.168.0.101.
CSeq: 103 INVITE.
Server: SIP Server 9.21/CS.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.
here is the trace using port 5744
U 91.176.221.245:63719 -> 212.212.212.212:5744
INVITE sip:1917425@212.212.212.213:5060;nat=yes SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-6f0161f0.
From: "EXT 101" ;tag=f8c2b14a55549ac2o0.
To: ;tag=as1fcb82f4.
Call-ID: 5832b9ea-a4a32aa2@192.168.0.101.
CSeq: 103 INVITE.
Max-Forwards: 70.
Route: .
Proxy-Authorization: Digest
username="WindowP1",realm="myserver",nonce="54f46d7a081272baa39f9e34a0ee81f471dec7b034ba",uri="
sip:1917425@212.212.212.213:5060
",algorithm=MD5,response="4096f485853bf6a209424c36dac2342b".
Contact: "EXT 101" .
Expires: 30.
User-Agent: Cisco/SPA525G2-7.5.6.
Content-Length: 224.
Content-Type: application/sdp.
.
v=0.
o=- 30981 30982 IN IP4 192.168.0.101.
s=-.
c=IN IP4 0.0.0.0.
t=0 0.
m=audio 16422 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendonly.

U 212.212.212.212:5744 -> 91.176.221.245:63719
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 192.168.0.101:5060
;received=91.176.221.245;rport=63719;branch=z9hG4bK-6f0161f0.
From: "EXT 101" ;tag=f8c2b14a55549ac2o0.
To: ;tag=as1fcb82f4.
Call-ID: 5832b9ea-a4a32aa2@192.168

[OpenSIPS-Users] ERROR:siptrace:pipport2su: bad protocol

2015-03-10 Thread Satish Patel

I am configuring siptrace with homer server and getting following error,

 ERROR:siptrace:pipport2su: bad protocol
 ERROR:siptrace:pipport2su: bad protocol
 ERROR:siptrace:pipport2su: bad protocol

Razvan fix following patch but still getting above error in log
https://github.com/OpenSIPS/opensips/commit/178b0cc26b05a81947de150fe1c2df36d600ccaa

Am i missing something?
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Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

2015-03-10 Thread Terrance Devor

On Tue, Mar 10, 2015 at 6:06 AM, mahan77  wrote:

> Hello Terrance,
>
> Thank you for your time to replay back.
> It was basic dispatcher Config file and posted in the first place.
>
> This is my sip trace.
>
> interface: eth0 (192.168.1.0/255.255.255.0)
> filter: (ip or ip6) and ( port 5060 )
>
> U 192.168.1.64:5060 -> 192.168.1.150:5060
> .
> Gb.."
> ?.
>
> U 87.117.75.1:5060 -> 192.168.1.150:5060
> INVITE sip:442032912345@192.168.1.150:5060 SIP/2.0.
> Via: SIP/2.0/UDP 87.117.75.1:5060;rport;branch=z9hG4bK9gZ0vUKSc0Q3m.
> Max-Forwards: 68.
> From: "07720212345" ;tag=1Z1Z5983N78ZN.
> To: .
> Call-ID: fe855718-40e4-1233-4c90-aba651435a79.
> CSeq: 72611343 INVITE.
> Contact: .
> User-Agent: TelNG GW.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, hold, conference, presence, dialog, line-seize,
> call-info, sla, include-session-description, presence.winfo,
> message-summary, refer.
> Session-Expires: 1800;refresher=uac.
> Min-SE: 120.
> Privacy: none.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 225.
> X-3C-ACCOUNT: 8166.
> X-3C-DIRECTION: in.
> X-FS-Support: update_display,send_info.
> P-Asserted-Identity: "07720212345" .
> .
> v=0.
> o=FreeSWITCH 1425866039 1425866040 IN IP4 87.117.75.1.
> s=FreeSWITCH.
> c=IN IP4 87.117.75.1.
> t=0 0.
> m=audio 28776 RTP/AVP 8 0 18 101 13.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 192.168.1.150:5060 -> 192.168.1.85:5060
> INVITE sip:442032912345@192.168.1.150:5060 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK9gZ0vUKSc0Q3m.
> Via: SIP/2.0/UDP
> 87.117.75.1:5060
> ;received=87.117.75.1;rport=5060;branch=z9hG4bK9gZ0vUKSc0Q3m.
> Max-Forwards: 67.
> From: "07720212345" ;tag=1Z1Z5983N78ZN.
> To: .
> Call-ID: fe855718-40e4-1233-4c90-aba651435a79.
> CSeq: 72611343 INVITE.
> Contact: .
> User-Agent: TelNG GW.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, hold, conference, presence, dialog, line-seize,
> call-info, sla, include-session-description, presence.winfo,
> message-summary, refer.
> Session-Expires: 1800;refresher=uac.
> Min-SE: 120.
> Privacy: none.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 225.
> X-3C-ACCOUNT: 8166.
> X-3C-DIRECTION: in.
> X-FS-Support: update_display,send_info.
> P-Asserted-Identity: "07720212345" .
> .
> v=0.
> o=FreeSWITCH 1425866039 1425866040 IN IP4 87.117.75.1.
> s=FreeSWITCH.
> c=IN IP4 87.117.75.1.
> t=0 0.
> m=audio 28776 RTP/AVP 8 0 18 101 13.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 192.168.1.85:5060 -> 192.168.1.150:5060
> OPTIONS sip:2001@192.168.1.64:5060;ob SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK2460f825;rport.
> Max-Forwards: 70.
> From: "asterisk" ;tag=as2b561864.
> To: .
> Contact: .
> Call-ID: 6432c1a87201b6d9051df2e71c82e6a7@192.168.1.85:5060.
> CSeq: 102 OPTIONS.
> User-Agent: Asterisk PBX SVN-branch-13-r432059.
> Date: Mon, 09 Mar 2015 09:53:35 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE.
> Supported: replaces, timer.
> Content-Length: 0.
> .
>
>
> U 192.168.1.85:5060 -> 192.168.1.150:5060
> SIP/2.0 401 Unauthorized.
> Via: SIP/2.0/UDP
> 192.168.1.150:5060
> ;branch=z9hG4bK9gZ0vUKSc0Q3m;received=192.168.1.150;rport=5060.
> Via: SIP/2.0/UDP
> 87.117.75.1:5060
> ;received=87.117.75.1;rport=5060;branch=z9hG4bK9gZ0vUKSc0Q3m.
> From: "07720212345" ;tag=1Z1Z5983N78ZN.
> To: ;tag=as63edc664.
> Call-ID: fe855718-40e4-1233-4c90-aba651435a79.
> CSeq: 72611343 INVITE.
> Server: Asterisk PBX SVN-branch-13-r432059.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE.
> Supported: replaces, timer.
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c8254d0".
> Content-Length: 0.
> .
>
>
> U 192.168.1.150:5060 -> 192.168.1.85:5060
> OPTIONS sip:2001@192.168.1.64:5060;ob SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK2460f825.
> Via: SIP/2.0/UDP
> 192.168.1.85:5060;received=192.168.1.85;branch=z9hG4bK2460f825;rport=5060.
> Max-Forwards: 69.
> From: "asterisk" ;tag=as2b561864.
> To: .
> Contact: .
> Call-ID: 6432c1a87201b6d9051df2e71c82e6a7@192.168.1.85:5060.
> CSeq: 102 OPTIONS.
> User-Agent: Asterisk PBX SVN-branch-13-r432059.
> Date: Mon, 09 Mar 2015 09:53:35 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE.
> Supported: replaces, timer.
> Content-Length: 0.
> .
>
>
> U 192.168.1.150:5060 -> 87.117.75.1:5060
> SIP/2.0 401 Unauthorized.
> Via: SIP/2.0/UDP
> 87.117.75.1:5060
> ;received=87.117.75.1;rport=5060;branch=z9hG4bK9gZ0vUKSc0Q3m.
> From: "07720212345"

Re: [OpenSIPS-Users] ERROR:siptrace:pipport2su: bad protocol

2015-03-10 Thread Satish Patel

I set debug=6 and here is the logs most of time error pop'ed up near "tm"
module


DBG:tm:run_trans_callbacks: trans=0x7f9570f06710, callback type 1024, id 1
entered
DBG:siptrace:trace_onreq_out: trace on req out
DBG:core:parse_headers: flags=40
DBG:siptrace:trace_msg_out: trace msg out
ERROR:siptrace:pipport2su: bad protocol


On Tue, Mar 10, 2015 at 12:12 PM, Satish Patel  wrote:

> I am configuring siptrace with homer server and getting following error,
>
>  ERROR:siptrace:pipport2su: bad protocol
>  ERROR:siptrace:pipport2su: bad protocol
>  ERROR:siptrace:pipport2su: bad protocol
>
> Razvan fix following patch but still getting above error in log
> https://github.com/OpenSIPS/opensips/commit/178b0cc26b05a81947de150fe1c2df36d600ccaa
>
> Am i missing something?
>
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Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

2015-03-10 Thread mahan77

Hello Terrance.Yes it’s the basics scripts.I’m trying to setup OpenSIPS as a
proxy to asterisk via dispatcher modules. First part I will able to register
sippers on asterisk via dispatcher modules. I’m trying work my way up. This
is my second part of the test; send incoming calls to asterisk IVR.  If you
have any working dispatcher scripts can you post or mail me.  It will help
me to understand more.Many thanks.Sathees



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Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

2015-03-10 Thread Terrance Devor

Hello Mahan,

My suggestion would be to do a lot more research:
i)
http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration
ii)
http://www.voip-sip.org/wp-content/uploads/2011/08/Building-Telephony-Systems-with-OpenSIPS-1.6.pdf

Take a few weeks and get to know opensips and build up your config to do
what you
require from it as a proxy. Without more work on your part, I really can't
offer any
help myself. Maybe others on here can.

If this is too hard a task for you. you can possibly seek an
individual/company that can
offer consulting work for your company.

Terrance


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Re: [OpenSIPS-Users] Safe Place/Method to Fix Contact

2015-03-10 Thread Terrance Devor

Hello Răzvan,

Thank you for your response. I think I am making a mess of this. Maybe
if I explain what I am trying to accomplish. Please consider the following:

U 2015/03/10 15:31:47.591498 25.21.74.12:5060 -> 192.168.2.5:5060
INVITE sip:9187321212@74.75.31.22 SIP/2.0.
Contact: .

U 2015/03/10 15:31:47.598538 192.168.2.5:5060 -> 192.168.2.10:5060
INVITE sip:9187321212@192.168.2.10:5060 SIP/2.0.
Contact: .

U 2015/03/10 15:31:47.857774 192.168.2.10:5060 -> 192.168.2.5:5060
SIP/2.0 100 Trying.
Contact: .

U 2015/03/10 15:31:47.858765 192.168.2.10:5060 -> 192.168.2.5:5060
SIP/2.0 200 OK.
Contact: .

U 2015/03/10 15:31:47.903559 192.168.2.5:5060 -> 25.21.74.12:5060
SIP/2.0 200 OK.
Contact: .

U 2015/03/10 15:31:47.945974 25.21.74.12:5060 -> 192.168.2.5:5060
ACK sip:192.168.2.20:5060 SIP/2.0.
Contact: .

U 2015/03/10 15:31:48.204696 192.168.2.5:5060 -> 192.168.2.10:5060
ACK sip:9187321212@192.168.2.10:5060 SIP/2.0.
Contact: .

U 2015/03/10 15:32:31.488749 25.21.74.12:5060 -> 192.168.2.5:5060
BYE sip:192.168.2.20:5060 SIP/2.0.
Contact: .

U 2015/03/10 15:32:31.658439 192.168.2.5:5060 -> 192.168.2.10:5060
BYE sip:9187321212@192.168.2.10:5060 SIP/2.0.
Contact: .

25,21.74.12 Underline Carrier
192.168.2.5 Opensips
192.168.2.10 Gateway

The contacts that I am attempting to do is modify is:

Message Two:
Original: Contact: .
Modified: Contact: .

Message Five:
Original: Contact: .
Modified: Contact: .

Message Seven:
Original: Contact: .
Modified: Contact: .

Message Nine:
Original: Contact: .
Modified: Contact: .

When I attempt to making such modifications, I am receiving errors such as:

Failed to match the sequential request to a known dialog
Attempt to route with preloaded Route's

What are the requirements to shape the Contact header to point to the right
IP and still maintain the preloaded
route.

Kind Regards,

Terrance
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Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

2015-03-10 Thread mahan77

Thank youi Will look into 



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[OpenSIPS-Users] Can somebody explain of this error?

2015-03-10 Thread mahan77

Hello,Can somebody explain of this error?DBG:core:get_hdr_field:  [38];
uri=[sip:442032912...@ddns.domain.com] DBG:core:get_hdr_field: to body
[]Many thanks Sathees



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Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

2015-03-10 Thread Terrance Devor

The  OpenSIPS project owners offer some excellent consulting solutions
that will save you a lot of time.

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Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

2015-03-10 Thread mahan77

Hello again,I love to learn. That’s why I’m trying myself.  Thank you
againsathees



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Re: [OpenSIPS-Users] Can somebody explain of this error?

2015-03-10 Thread Jeff Pyle

Sathees,

That's not an error, just a debug statement.  (Notice the DBG in
DBG:core:get_hdr_field.)  If you don't want to see those, set the debug=
value lower in opensips.cfg.  I believe 3 is a normal running value.  2 is
even less verbose.

- Jeff

On Tue, Mar 10, 2015 at 5:07 PM, mahan77  wrote:

> Hello, Can somebody explain of this error? DBG:core:get_hdr_field:  [38];
> uri=[sip:442032912...@ddns.domain.com] DBG:core:get_hdr_field: to body [<
> sip:442032912...@ddns.domain.com>] Many thanks Sathees
> --
> View this message in context: Can somebody explain of this error?
> 
> Sent from the OpenSIPS - Users mailing list archive
> 
> at Nabble.com.
>
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[OpenSIPS-Users] set_advertised_address in on_reply

2015-03-10 Thread Terrance Devor

Hello Everyone,

When trying to change the record_route using set_advertised_address in the
on_reply route there is no change. I am testing using some test headers and
know that the location is getting triggered however, cannot change the RR
using
set_advertised_address. Is this possible?

Kind Regards,

Terrance.
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Re: [OpenSIPS-Users] Blox SBC

2015-03-10 Thread Varadhan Work

Source code

https://github.com/blox-org

On Tue, Mar 10, 2015 at 5:40 PM, Varadhan Work 
wrote:

> Hello,
>
> I'm Varadhan,
>
> We have come up with an idea to build opensource SBC with opensips as core
> SIP session route, thanks to opensips and its community.
>
> Now we have launched beta version of Blox (www.blox.org)
>
> Blox is a Session Border Controller(SBC) used to control VoIP signaling
> and media streams. SBC is responsible for setting up, conducting, and
> tearing down calls. SBC allows owners to control the types of call that can
> be placed through the networks and also overcome some of the problems
> caused by firewalls and NAT for VoIP calls. A common location for a
> stand-alone SBC is a connection point, called a border, between a private
> local area network (LAN) and the Internet. SBC polices real-time voice
> traffic between IP network borders ensuring your private network is
> robustly secure and fully manageable.
> You can download and install Blox via www.blox.org, please post your
> valuable feedback.
>
> Thanks & Regards,
> Varadhan M
>
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Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

[OpenSIPS-Users] dialog errors

[OpenSIPS-Users] Blox SBC

[OpenSIPS-Users] How to remove/update dialog

Re: [OpenSIPS-Users] Blox SBC

Re: [OpenSIPS-Users] Call transfer problem

[OpenSIPS-Users] Problem with placing call on hold

[OpenSIPS-Users] ERROR:siptrace:pipport2su: bad protocol

Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

Re: [OpenSIPS-Users] ERROR:siptrace:pipport2su: bad protocol

Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

Re: [OpenSIPS-Users] Safe Place/Method to Fix Contact

Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

[OpenSIPS-Users] Can somebody explain of this error?

Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

Re: [OpenSIPS-Users] incoming DID will not pass to asterisk

Re: [OpenSIPS-Users] Can somebody explain of this error?

[OpenSIPS-Users] set_advertised_address in on_reply

Re: [OpenSIPS-Users] Blox SBC

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