Hi all
After a long iam back to forum
back to my own topic and several readings done on this forum
how people doing same kind of setup what iam trying to achive
so here i have done some good developements
for testing iam doing all in one Server
Step1 :
Installed in Fresh BOX with Debian
Asterisk and A2B working Fine
Step2 : registered with SIP account iam able to make calls successfully
Step3 :
installed Opensips
Made Subscribers to view from A2b Database
Step4 : changed Asterisk port from 5060 to 5062
Step5 : Opensip config made changes to register users with Opensips
and when they dial 001X call send to Asterisk box
route[3]{
if (uri =~ "sip:001[0-9]@*"){
log(1, "Forwarding to Asterisk \n");
rewritehostport("A2b-asterisk-IP:5062");
route(1);
exit;
}
Works Fine, No problems as of now
But to go in advance, i want to use Number of * boxes to achive more Load
Step5 : added Dispatcher Module in the Opensips
loadmodule "dispatcher.so"
.
.
.
modparam("dispatcher","list_file","/usr/local/etc/opensips/dispatcher.cfg")
.
.
.
.
changed route to use dispatcher
route[3]{
if (uri =~ "sip:001[0-9]@*"){
log(1, "Forwarding to Asterisk \n");
ds_select_dst("2","4");
forward();
route(1);
exit;
}
My dispatcher Config Looks like below
dispatcher.cfg
2 sip:a2b-asterisk-ip:5062
2 sip:a2b-asterisk-ip2:5062
I have restarted Opensips
when i dial 0017XXXXXX number the call send Opensips to Asterisk
Jun 30 01:12:28 opensips[25868]: Forwarding to Asterisk
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: set [2]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: alg hash [1]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: selected
[4-2/1] <sip:a2b-asterisk-ip:5062>
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:mk_proxy: doing DNS
lookup...
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:forward_request:
sending:#012INVITE sip:0017XXXXXXXX at opensips-ip:5060
SIP/2.0#015#012Record-Route: <sip:opensips-ip;lr=on>#015#012Via:
SIP/2.0/UDP
opensips-ip;branch=z9hG4bK28178282572929210914#015#012Via: SIP/2.0/UDP
ip-phone-ip:5060;received=ip-phone-ip;branch=z9hG4bK28178282572929210914;rport=5060#015#012From:
4720779942 <sip:4720779942 at opensips-ip:5060>;tag=1966722825#015#012To:
0017325824631 <sip:0017XXXXXXX at opensips-ip:5060>#015#012Call-ID:
32167199575863-11502744529360 at ip-phoneip#015#012CSeq: 2
INVITE#015#012Contact:
<sip:4720779942 at ipphone-ip:5060>#015#012Proxy-Authorization:
Digest username="4720779942", realm="asterisk",
nonce="79ee65ba",
uri="sip:0017XXXXXX at opensips-ip:5060",
response="3e182f165a5663d0b145d6b55d34e94b",
algorithm=MD5#015#012Max-Forwards: 69#015#012Supported:
replaces#015#012User-Agent: Voip Phone 1.0#015#012Allow: INVITE, ACK,
OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK,
UPDATE#015#012Content-Type: application/sdp#015#012Content-Length:
319#015#012#015#012v=0#015#012o=4720779942 17025328 32005127 IN IP4
202.63.111.2#015#012s=A conversation#015#012c=IN IP4 ip-phone-ip#015#012t=0
0#015#012m=audio 10028 RTP/AVP 18 4 8 0 9 101#015#012a=rtpmap:18
G729/8000#015#012a=rtpmap:4 G723/8000#015#012a=rtpmap:8
PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:9
G722/16000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=sendrecv#015#012.
opensips[25868]: DBG:core:forward_request: orig. len=1087, new_len=1220,
proto=1
when i ngrep
------------
U 2009/06/30 01:59:20.770599 ipphone:5060 -> asterisk-a2b-ip:5060
INVITE sip:0017XXXXXXXX at asterisk-a2b-ip:5060 SIP/2.0.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport.
From: 4720779942 <sip:4720779942 at asterisk-a2b-ip:5060>;tag=3037030266.
To: 0017XXXXXXXX <sip:0017XXXXXXXX at asterisk-a2b-ip:5060>.
Call-ID: 14399316162240-7371067914582 at ipphone.
CSeq: 2 INVITE.
Contact: <sip:4720779942 at ipphone:5060>.
Proxy-Authorization: Digest username="4720779942",
realm="asterisk",
nonce="07ba8624", uri="sip:0017XXXXXXXX at
asterisk-a2b-ip:5060",
response="5dbe9b2937d0bc3f6e8d25052fff0b6a", algorithm=MD5.
Max-Forwards: 70.
Supported: replaces.
User-Agent: Voip Phone 1.0.
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE,
PRACK, UPDATE.
Content-Type: application/sdp.
Content-Length: 319.
.
v=0.
o=4720779942 69102627 18481147 IN IP4 ipphone.
s=A conversation.
c=IN IP4 ipphone.
t=0 0.
m=audio 10034 RTP/AVP 18 4 8 0 9 101.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:9 G722/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
U 2009/06/30 01:59:20.774528 asterisk-a2b-ip:5060 -> ipphone:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport=5060.
From: 4720779942 <sip:4720779942 at asterisk-a2b-ip:5060>;tag=3037030266.
To: 0017XXXXXXXX <sip:0017XXXXXXXX at asterisk-a2b-ip:5060>.
Call-ID: 14399316162240-7371067914582 at ipphone.
CSeq: 2 INVITE.
Server: OpenSIPS (1.5.1-notls (i386/linux)).
Content-Length: 0.
.
U 2009/06/30 01:59:21.650498 asterisk-a2b-ip:5060 -> ipphone:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
ipphone:5060;received=ipphone;branch=z9hG4bK1984515716453028636;rport=5060.
From: 4720779942 <sip:4720779942 at asterisk-a2b-ip:5060>;tag=3037030266.
To: 0017XXXXXXXX <sip:0017XXXXXXXX at
asterisk-a2b-ip:5060>;tag=as0cb075c5.
Call-ID: 14399316162240-7371067914582 at ipphone.
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="07ba8624".
Content-Length: 0.
------
when i enable debug at Asterisk and Look at i see the below error
---------------------------------------------------------------
<--- SIP read from a2b-asterisk-ip:5060 --->
INVITE sip:0017XXXXXXXXX at a2b-asterisk-ip:5060 SIP/2.0
Record-Route: <sip:a2b-asterisk-ip;lr=on>
Via: SIP/2.0/UDP a2b-asterisk-ip;branch=z9hG4bK166.1b7e2827.0
Via: SIP/2.0/UDP
Ip-phone:5060;received=Ip-phone;branch=z9hG4bK295731884823024293;rport=5060
From: 4720779942 <sip:4720779942 at a2b-asterisk-ip:5060>;tag=12544334
To: 0017XXXXXXXXX <sip:0017XXXXXXXXX at a2b-asterisk-ip:5060>
Call-ID: 16946271051109-143302828620026 at Ip-phone
CSeq: 1 INVITE
Contact: <sip:4720779942 at Ip-phone:5060>
Max-Forwards: 69
Supported: replaces
User-Agent: Voip Phone 1.0
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE,
PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 319
v=0
o=4720779942 31008195 22123120 IN IP4 Ip-phone
s=A conversation
c=IN IP4 Ip-phone
t=0 0
m=audio 10030 RTP/AVP 18 4 8 0 9 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Jun 30 01:15:29] VERBOSE[24612] logger.c: --- (15 headers 14 lines) ---
[Jun 30 01:15:29] VERBOSE[24612] logger.c: Ignoring this INVITE request
[Jun 30 01:15:31] VERBOSE[24612] logger.c: Reliably Transmitting (no NAT) to
termination-provider-ip:5062:
OPTIONS sip:termination-provider-ip:5062 SIP/2.0
Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport
From: "asterisk" <sip:asterisk at
a2b-asterisk-ip:5062>;tag=as4cf91fd8
To: <sip:termination-provider-ip:5062>
Contact: <sip:asterisk at a2b-asterisk-ip:5062>
Call-ID: 65a49c0977c6de0a1d2dbbfe757724bd at a2b-asterisk-ip
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 30 Jun 2009 08:15:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Jun 30 01:15:32] VERBOSE[24612] logger.c:
<--- SIP read from termination-provider-ip:5062 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport=5062
From: "asterisk" <sip:asterisk at
a2b-asterisk-ip:5062>;tag=as4cf91fd8
To:
<sip:termination-provider-ip:5062>;tag=2560d490c3265ff35995c6bbde62a7c3.ee5a
Call-ID: 65a49c0977c6de0a1d2dbbfe757724bd at a2b-asterisk-ip
CSeq: 102 OPTIONS
Content-Length: 0
---------
why does Asterisk sending with out any values
---
From: "asterisk" <sip:asterisk at
a2b-asterisk-ip:5062>;tag=as4cf91fd8
To: <sip:termination-provider-ip:5062>
---
Any suggestions
Ram
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Hi Ram,
Does your OpenSIPS get any SIP reply from Asterisk? or the INVITE is
simply discarded by Asterisk?
Regards,
Bogdan
Hi all
After a long iam back to forum
back to my own topic and several readings done on this forum
how people doing same kind of setup what iam trying to achive
so here i have done some good developements
for testing iam doing all in one Server
Step1 :
Installed in Fresh BOX with Debian
Asterisk and A2B working Fine
Step2 : registered with SIP account iam able to make calls successfully
Step3 :
installed Opensips
Made Subscribers to view from A2b Database
Step4 : changed Asterisk port from 5060 to 5062
Step5 : Opensip config made changes to register users with Opensips
and when they dial 001X call send to Asterisk box
route[3]{
if (uri =~ "sip:001[0-9]@*"){
log(1, "Forwarding to Asterisk \n");
rewritehostport("A2b-asterisk-IP:5062");
route(1);
exit;
}
Works Fine, No problems as of now
But to go in advance, i want to use Number of * boxes to achive more Load
Step5 : added Dispatcher Module in the Opensips
loadmodule "dispatcher.so"
.
.
.
modparam("dispatcher","list_file","/usr/local/etc/opensips/dispatcher.cfg")
.
.
.
.
changed route to use dispatcher
route[3]{
if (uri =~ "sip:001[0-9]@*"){
log(1, "Forwarding to Asterisk \n");
ds_select_dst("2","4");
forward();
route(1);
exit;
}
My dispatcher Config Looks like below
dispatcher.cfg
2 sip:a2b-asterisk-ip:5062
2 sip:a2b-asterisk-ip2:5062
I have restarted Opensips
when i dial 0017XXXXXX number the call send Opensips to Asterisk
Jun 30 01:12:28 opensips[25868]: Forwarding to Asterisk
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: set [2]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: alg hash [1]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: selected
[4-2/1] <sip:a2b-asterisk-ip:5062>
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:mk_proxy: doing DNS
lookup...
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:forward_request:
sending:#012INVITE sip:0017XXXXXXXX at opensips-ip:5060
SIP/2.0#015#012Record-Route: <sip:opensips-ip;lr=on>#015#012Via:
SIP/2.0/UDP
opensips-ip;branch=z9hG4bK28178282572929210914#015#012Via: SIP/2.0/UDP
ip-phone-ip:5060;received=ip-phone-ip;branch=z9hG4bK28178282572929210914;rport=5060#015#012From:
4720779942 <sip:4720779942 at opensips-ip:5060>;tag=1966722825#015#012To:
0017325824631 <sip:0017XXXXXXX at opensips-ip:5060>#015#012Call-ID:
32167199575863-11502744529360 at ip-phoneip#015#012CSeq: 2
INVITE#015#012Contact:
<sip:4720779942 at ipphone-ip:5060>#015#012Proxy-Authorization:
Digest username="4720779942", realm="asterisk",
nonce="79ee65ba",
uri="sip:0017XXXXXX at opensips-ip:5060",
response="3e182f165a5663d0b145d6b55d34e94b",
algorithm=MD5#015#012Max-Forwards: 69#015#012Supported:
replaces#015#012User-Agent: Voip Phone 1.0#015#012Allow: INVITE, ACK,
OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK,
UPDATE#015#012Content-Type: application/sdp#015#012Content-Length:
319#015#012#015#012v=0#015#012o=4720779942 17025328 32005127 IN IP4
202.63.111.2#015#012s=A conversation#015#012c=IN IP4 ip-phone-ip#015#012t=0
0#015#012m=audio 10028 RTP/AVP 18 4 8 0 9 101#015#012a=rtpmap:18
G729/8000#015#012a=rtpmap:4 G723/8000#015#012a=rtpmap:8
PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:9
G722/16000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=sendrecv#015#012.
opensips[25868]: DBG:core:forward_request: orig. len=1087, new_len=1220,
proto=1
when i ngrep
------------
U 2009/06/30 01:59:20.770599 ipphone:5060 -> asterisk-a2b-ip:5060
INVITE sip:0017XXXXXXXX at asterisk-a2b-ip:5060 SIP/2.0.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport.
From: 4720779942 <sip:4720779942 at asterisk-a2b-ip:5060>;tag=3037030266.
To: 0017XXXXXXXX <sip:0017XXXXXXXX at asterisk-a2b-ip:5060>.
Call-ID: 14399316162240-7371067914582 at ipphone.
CSeq: 2 INVITE.
Contact: <sip:4720779942 at ipphone:5060>.
Proxy-Authorization: Digest username="4720779942",
realm="asterisk",
nonce="07ba8624", uri="sip:0017XXXXXXXX at
asterisk-a2b-ip:5060",
response="5dbe9b2937d0bc3f6e8d25052fff0b6a", algorithm=MD5.
Max-Forwards: 70.
Supported: replaces.
User-Agent: Voip Phone 1.0.
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE,
PRACK, UPDATE.
Content-Type: application/sdp.
Content-Length: 319.
.
v=0.
o=4720779942 69102627 18481147 IN IP4 ipphone.
s=A conversation.
c=IN IP4 ipphone.
t=0 0.
m=audio 10034 RTP/AVP 18 4 8 0 9 101.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:9 G722/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
U 2009/06/30 01:59:20.774528 asterisk-a2b-ip:5060 -> ipphone:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport=5060.
From: 4720779942 <sip:4720779942 at asterisk-a2b-ip:5060>;tag=3037030266.
To: 0017XXXXXXXX <sip:0017XXXXXXXX at asterisk-a2b-ip:5060>.
Call-ID: 14399316162240-7371067914582 at ipphone.
CSeq: 2 INVITE.
Server: OpenSIPS (1.5.1-notls (i386/linux)).
Content-Length: 0.
.
U 2009/06/30 01:59:21.650498 asterisk-a2b-ip:5060 -> ipphone:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
ipphone:5060;received=ipphone;branch=z9hG4bK1984515716453028636;rport=5060.
From: 4720779942 <sip:4720779942 at asterisk-a2b-ip:5060>;tag=3037030266.
To: 0017XXXXXXXX <sip:0017XXXXXXXX at
asterisk-a2b-ip:5060>;tag=as0cb075c5.
Call-ID: 14399316162240-7371067914582 at ipphone.
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="07ba8624".
Content-Length: 0.
------
when i enable debug at Asterisk and Look at i see the below error
---------------------------------------------------------------
<--- SIP read from a2b-asterisk-ip:5060 --->
INVITE sip:0017XXXXXXXXX at a2b-asterisk-ip:5060 SIP/2.0
Record-Route: <sip:a2b-asterisk-ip;lr=on>
Via: SIP/2.0/UDP a2b-asterisk-ip;branch=z9hG4bK166.1b7e2827.0
Via: SIP/2.0/UDP
Ip-phone:5060;received=Ip-phone;branch=z9hG4bK295731884823024293;rport=5060
From: 4720779942 <sip:4720779942 at a2b-asterisk-ip:5060>;tag=12544334
To: 0017XXXXXXXXX <sip:0017XXXXXXXXX at a2b-asterisk-ip:5060>
Call-ID: 16946271051109-143302828620026 at Ip-phone
CSeq: 1 INVITE
Contact: <sip:4720779942 at Ip-phone:5060>
Max-Forwards: 69
Supported: replaces
User-Agent: Voip Phone 1.0
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE,
PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 319
v=0
o=4720779942 31008195 22123120 IN IP4 Ip-phone
s=A conversation
c=IN IP4 Ip-phone
t=0 0
m=audio 10030 RTP/AVP 18 4 8 0 9 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Jun 30 01:15:29] VERBOSE[24612] logger.c: --- (15 headers 14 lines) ---
[Jun 30 01:15:29] VERBOSE[24612] logger.c: Ignoring this INVITE request
[Jun 30 01:15:31] VERBOSE[24612] logger.c: Reliably Transmitting (no NAT) to
termination-provider-ip:5062:
OPTIONS sip:termination-provider-ip:5062 SIP/2.0
Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport
From: "asterisk" <sip:asterisk at
a2b-asterisk-ip:5062>;tag=as4cf91fd8
To: <sip:termination-provider-ip:5062>
Contact: <sip:asterisk at a2b-asterisk-ip:5062>
Call-ID: 65a49c0977c6de0a1d2dbbfe757724bd at a2b-asterisk-ip
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 30 Jun 2009 08:15:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Jun 30 01:15:32] VERBOSE[24612] logger.c:
<--- SIP read from termination-provider-ip:5062 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport=5062
From: "asterisk" <sip:asterisk at
a2b-asterisk-ip:5062>;tag=as4cf91fd8
To:
<sip:termination-provider-ip:5062>;tag=2560d490c3265ff35995c6bbde62a7c3.ee5a
Call-ID: 65a49c0977c6de0a1d2dbbfe757724bd at a2b-asterisk-ip
CSeq: 102 OPTIONS
Content-Length: 0
---------
why does Asterisk sending with out any values
---
From: "asterisk" <sip:asterisk at
a2b-asterisk-ip:5062>;tag=as4cf91fd8
To: <sip:termination-provider-ip:5062>
---
Any suggestions
Ram