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Nasir Javaid

2010-May-12 14:14 UTC

[asterisk-users] asterisk-users Digest, Vol 70, Issue 25

Hi Vardan

I did same as you told and deleted the SIP information in Astdb and
restarted asterisk. but the result was same.

as you said there might be mistake in sip.conf so i am pasting both servers
configuration here..

1- nasir.server.com

[abc]
username=abc
type=friend
secret=mysecret
nat=yes
mailbox=12234568
incominglimit=2
outgoinglimit=2
host=dynamic
dtmfmode=rfc2833
context=payasyougo
canreinvite=yes
callerid="Nasir Qazi" <12234>
accountcode=6:0:abc
amaflags=default
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm


2- 192.168.0.254 (client system)

[abc]
type=peer
username=abc
secret=mysecret
host=nasir.server.com
context=default
dtmfmode=rfc2833
canreinvite=yes
insecure=very
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
;qualify=yes

[caller]
type=friend
secret=123456
host=dynamic
callerid="caller <12129887777>"
context=out
nat=yes
dtmfmode=rfc2833
canreinvite=yes
insecure=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
t38_udptl=yes
qualify=yes


I have registered [caller] on xlite at client system and dialing following
context in local system that will dial [abc]

[out]
exten=> _X.,1,Dial(SIP/${EXTEN}@abc,30,1)
exten=> _X.,n,Hangup


as you can see above *highlighted that context of abc is
payasyougo.*problem is that i want the call to land in that context on
nasir.server.com, which works if i use register string. but without register
string call goes to default context on nasir.server.com

regards,

Nasir Javaid

Message: 19
Date: Tue, 11 May 2010 20:54:30 +0500
From: Vardan <hvardan71 at gmail.com>
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24
To: asterisk-users at lists.digium.com
Message-ID: <hsbujk$qk9$1 at dough.gmane.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hello Nasir

I have some please.
Do so, it help.
Find all records about interexchange beetwen this two server and delete
all records in sip.conf for this both server (first make backup
sip.conf, or any another conf file that you use).
restart asterisk.
look in astdb about this old records, if any found, delete him
Next, create new record in sip.conf on both servers, without
registration string, reload sip.conf.
give him right context from extensions.conf.

Can you do this?

I think is some mistake about configuration in sip.conf, you have I
think two same records (peer or friend).

Vardan
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Nasir Javaid

2010-May-12 14:26 UTC

head link

[asterisk-users] asterisk-users Digest, Vol 70, Issue 25

here i am attaching debug trace of sip in case of sccessfull call when using
register string...


*CLI> [May 12 19:21:14]
<--- SIP read from 192.168.0.254:5060 --->
INVITE sip:17185594743 at nasir.server.com
<sip%3A17185594743 at nasir.server.com>SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport
Max-Forwards: 70
From: "caller" <sip:12129887777 at
192.168.0.254<sip%3A12129887777 at 192.168.0.254>>;tag=as76623e31To: <sip:17185594743 at nasir.server.com <sip%3A17185594743 at
nasir.server.com>>
Contact: <sip:12129887777 at 192.168.0.254 <sip%3A12129887777 at
192.168.0.254>>
Call-ID: 245c407103141a6841c0ac106bd5a53d at 192.168.0.254
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.0
Date: Wed, 12 May 2010 14:20:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 618893758 618893758 IN IP4 192.168.0.254
s=Asterisk PBX 1.6.2.0
c=IN IP4 192.168.0.254
t=0 0
m=audio 11026 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
[May 12 19:21:14] --- (14 headers 13 lines) ---
[May 12 19:21:14] Sending to 192.168.0.254 : 5060 (NAT)
[May 12 19:21:14] Using INVITE request as basis request -
245c407103141a6841c0ac106bd5a53d at 192.168.0.254
[May 12 19:21:14] Found peer 'abc'
[May 12 19:21:14]
<--- Reliably Transmitting (NAT) to 192.168.0.254:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.254:5060
;branch=z9hG4bK3c63f272;received=192.168.0.254;rport=5060
From: "caller" <sip:12129887777 at
192.168.0.254<sip%3A12129887777 at 192.168.0.254>>;tag=as76623e31To: <sip:17185594743 at nasir.server.com <sip%3A17185594743 at
nasir.server.com>>;tag=as0a721b3aCall-ID: 245c407103141a6841c0ac106bd5a53d at 192.168.0.254
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="7bc52d0a"
Content-Length: 0


<------------>
[May 12 19:21:14] Scheduling destruction of SIP dialog '
245c407103141a6841c0ac106bd5a53d at 192.168.0.254' in 32000 ms (Method:
INVITE)
[May 12 19:21:14]
<--- SIP read from 192.168.0.254:5060 --->
ACK sip:17185594743 at nasir.server.com <sip%3A17185594743 at
nasir.server.com>SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport
Max-Forwards: 70
From: "caller" <sip:12129887777 at
192.168.0.254<sip%3A12129887777 at 192.168.0.254>>;tag=as76623e31To: <sip:17185594743 at nasir.server.com <sip%3A17185594743 at
nasir.server.com>>;tag=as0a721b3aContact: <sip:12129887777 at 192.168.0.254 <sip%3A12129887777 at
192.168.0.254>>
Call-ID: 245c407103141a6841c0ac106bd5a53d at 192.168.0.254
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.0
Content-Length: 0


<------------->
[May 12 19:21:14] --- (10 headers 0 lines) ---
[May 12 19:21:14]
<--- SIP read from 192.168.0.254:5060 --->
INVITE sip:17185594743 at nasir.server.com
<sip%3A17185594743 at nasir.server.com>SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK05611806;rport
Max-Forwards: 70
From: "caller" <sip:12129887777 at
192.168.0.254<sip%3A12129887777 at 192.168.0.254>>;tag=as76623e31To: <sip:17185594743 at nasir.server.com <sip%3A17185594743 at
nasir.server.com>>
Contact: <sip:12129887777 at 192.168.0.254 <sip%3A12129887777 at
192.168.0.254>>
Call-ID: 245c407103141a6841c0ac106bd5a53d at 192.168.0.254
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.0
Proxy-Authorization: Digest username="abc",
realm="asterisk", algorithm=MD5,
uri="sip:17185594743 at nasir.server.com <sip%3A17185594743 at
nasir.server.com>",
nonce="7bc52d0a",
response="f138ecd92bb706207a7b8d00f1c1bed7"
Date: Wed, 12 May 2010 14:20:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 618893758 618893759 IN IP4 192.168.0.254
s=Asterisk PBX 1.6.2.0
c=IN IP4 192.168.0.254
t=0 0
m=audio 11026 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
[May 12 19:21:14] --- (15 headers 13 lines) ---
[May 12 19:21:14] Sending to 192.168.0.254 : 5060 (NAT)
[May 12 19:21:14] Using INVITE request as basis request -
245c407103141a6841c0ac106bd5a53d at 192.168.0.254
[May 12 19:21:14] Found peer 'abc'
[May 12 19:21:14] Found RTP audio format 0
[May 12 19:21:14] Found RTP audio format 3
[May 12 19:21:14] Found RTP audio format 101
[May 12 19:21:14] Peer audio RTP is at port 192.168.0.254:11026
[May 12 19:21:14] Found description format PCMU for ID 0
[May 12 19:21:14] Found description format GSM for ID 3
[May 12 19:21:14] Found description format telephone-event for ID 101
[May 12 19:21:14] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer -
audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
[May 12 19:21:14] Non-codec capabilities (dtmf): us - 0x1 (telephone-event),
peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May 12 19:21:14] Peer audio RTP is at port 192.168.0.254:11026
[May 12 19:21:14] Looking for 17185594743 in payasyougo (domain
nasir.server.com)
[May 12 19:21:14] WARNING[3785]: chan_sip.c:3930 sip_new: setting callerid
number to 12129339037
[May 12 19:21:14] list_route: hop:
<sip:12129887777 at 192.168.0.254<sip%3A12129887777 at
192.168.0.254>>[May 12 19:21:14]
<--- Transmitting (NAT) to 192.168.0.254:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.254:5060
;branch=z9hG4bK05611806;received=192.168.0.254;rport=5060
From: "caller" <sip:12129887777 at
192.168.0.254<sip%3A12129887777 at 192.168.0.254>>;tag=as76623e31To: <sip:17185594743 at nasir.server.com <sip%3A17185594743 at
nasir.server.com>>
Call-ID: 245c407103141a6841c0ac106bd5a53d at 192.168.0.254
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:17185594743 at nasir.server.com<sip%3A17185594743 at
nasir.server.com>>Content-Length: 0



On Wed, May 12, 2010 at 7:14 PM, Nasir Javaid <nasirjavaidnasir at
gmail.com>wrote:
> Hi Vardan
>
> I did same as you told and deleted the SIP information in Astdb and
> restarted asterisk. but the result was same.
>
> as you said there might be mistake in sip.conf so i am pasting both servers
> configuration here..
>
> 1- nasir.server.com
>
> [abc]
> username=abc
> type=friend
> secret=mysecret
> nat=yes
> mailbox=12234568
> incominglimit=2
> outgoinglimit=2
> host=dynamic
> dtmfmode=rfc2833
> context=payasyougo
> canreinvite=yes
> callerid="Nasir Qazi" <12234>
> accountcode=6:0:abc
> amaflags=default
>
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
>
>
> 2- 192.168.0.254 (client system)
>
>
> [abc]
> type=peer
> username=abc
> secret=mysecret
> host=nasir.server.com
>
> context=default
> dtmfmode=rfc2833
> canreinvite=yes
> insecure=very
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> nat=yes
> ;qualify=yes
>
> [caller]
> type=friend
> secret=123456
> host=dynamic
> callerid="caller <12129887777>"
> context=out
> nat=yes
> dtmfmode=rfc2833
> canreinvite=yes
> insecure=no
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> t38_udptl=yes
> qualify=yes
>
>
> I have registered [caller] on xlite at client system and dialing following
> context in local system that will dial [abc]
>
> [out]
> exten=> _X.,1,Dial(SIP/${EXTEN}@abc,30,1)
> exten=> _X.,n,Hangup
>
>
> as you can see above *highlighted that context of abc is
payasyougo.*problem is that i want the call to land in that context on
> nasir.server.com, which works if i use register string. but without
> register string call goes to default context on nasir.server.com
>
> regards,
>
> Nasir Javaid
>
>
> Message: 19
> Date: Tue, 11 May 2010 20:54:30 +0500
> From: Vardan <hvardan71 at gmail.com>
> Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24
> To: asterisk-users at lists.digium.com
> Message-ID: <hsbujk$qk9$1 at dough.gmane.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hello Nasir
>
> I have some please.
> Do so, it help.
> Find all records about interexchange beetwen this two server and delete
> all records in sip.conf for this both server (first make backup
> sip.conf, or any another conf file that you use).
> restart asterisk.
> look in astdb about this old records, if any found, delete him
> Next, create new record in sip.conf on both servers, without
> registration string, reload sip.conf.
> give him right context from extensions.conf.
>
> Can you do this?
>
> I think is some mistake about configuration in sip.conf, you have I
> think two same records (peer or friend).
>
> Vardan
>-------------- next part --------------
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Vardan

2010-May-12 14:27 UTC

head link

[asterisk-users] asterisk-users Digest, Vol 70, Issue 25

Look, you do again with registration.
remove any registration information.
Look this config, I think it can help you


Server1:

sip.conf

[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729

extensions.conf

[calltoserver2]
  exten =>  _X.,1,Noop(Call to server2)
  exten =>  _X.,2,Dial(SIP/interboxserver2/${EXTEN})
  exten =>  _X.,3,Hangup

[callfromserver2]

exten => _X.,1,Noop(Call from server2)
exten => _X.,2,Dial(SIP/${EXTEN})
exten => _X.,3,Hangup


Server2:

sip.conf

[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729

extensions.conf

[calltoserver1]
  exten =>  _X.,1,Noop(Call to server1)
  exten =>  _X.,2,Dial(SIP/interboxserver1/${EXTEN})
  exten =>  _X.,3,Hangup

[callfromserver1]

exten => _X.,1,Noop(Call from server1)
exten => _X.,2,Dial(SIP/${EXTEN})
exten => _X.,3,Hangup


Try so, I think it must work.
And also, look and delete any another records in both servers in 
sip.conf about this servers settings.

Vardan

Nasir Javaid wrote:> Hi Vardan
>
> I did same as you told and deleted the SIP information in Astdb and
> restarted asterisk. but the result was same.
>
> as you said there might be mistake in sip.conf so i am pasting both
> servers configuration here..
>
> 1- nasir.server.com <http://nasir.server.com>
>
> [abc]
> username=abc
> type=friend
> secret=mysecret
> nat=yes
> mailbox=12234568
> incominglimit=2
> outgoinglimit=2
> host=dynamic
> dtmfmode=rfc2833
> context=payasyougo
> canreinvite=yes
> callerid="Nasir Qazi" <12234>
> accountcode=6:0:abc
> amaflags=default
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
>
>
> 2- 192.168.0.254 (client system)
>
> [abc]
> type=peer
> username=abc
> secret=mysecret
> host=nasir.server.com <http://nasir.server.com>
> context=default
> dtmfmode=rfc2833
> canreinvite=yes
> insecure=very
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> nat=yes
> ;qualify=yes
>
> [caller]
> type=friend
> secret=123456
> host=dynamic
> callerid="caller <12129887777>"
> context=out
> nat=yes
> dtmfmode=rfc2833
> canreinvite=yes
> insecure=no
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> t38_udptl=yes
> qualify=yes
>
>
> I have registered [caller] on xlite at client system and dialing
> following context in local system that will dial [abc]
>
> [out]
> exten=> _X.,1,Dial(SIP/${EXTEN}@abc,30,1)
> exten=> _X.,n,Hangup
>
>
> as you can see above *highlighted that context of abc is payasyougo.*
> problem is that i want the call to land in that context on
> nasir.server.com <http://nasir.server.com>, which works if i use
> register string. but without register string call goes to default
> context on nasir.server.com <http://nasir.server.com>
>
> regards,
>
> Nasir Javaid
>
> Message: 19
> Date: Tue, 11 May 2010 20:54:30 +0500
> From: Vardan <hvardan71 at gmail.com <mailto:hvardan71 at
gmail.com>>
> Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24
> To: asterisk-users at lists.digium.com <mailto:asterisk-users at
lists.digium.com>
> Message-ID: <hsbujk$qk9$1 at dough.gmane.org <mailto:1 at
dough.gmane.org>>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hello Nasir
>
> I have some please.
> Do so, it help.
> Find all records about interexchange beetwen this two server and delete
> all records in sip.conf for this both server (first make backup
> sip.conf, or any another conf file that you use).
> restart asterisk.
> look in astdb about this old records, if any found, delete him
> Next, create new record in sip.conf on both servers, without
> registration string, reload sip.conf.
> give him right context from extensions.conf.
>
> Can you do this?
>
> I think is some mistake about configuration in sip.conf, you have I
> think two same records (peer or friend).
>
> Vardan
>

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