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

Wed May 12 09:26:26 CDT 2010

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=as76623e31
To: <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=as76623e31
To: <sip:17185594743 at nasir.server.com <sip%3A17185594743 at nasir.server.com>
>;tag=as0a721b3a
Call-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=as76623e31
To: <sip:17185594743 at nasir.server.com <sip%3A17185594743 at nasir.server.com>
>;tag=as0a721b3a
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 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=as76623e31
To: <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=as76623e31
To: <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
>
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