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

[Nasir Javaid]

Hi again,

below is debug trace of * cli when i remove register string from sip.conf


*CLI> [May 12 19:33:06]
<--- 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=z9hG4bK56e3b44a;rport
Max-Forwards: 70
From: "caller" <sip:12129887777 at 192.168.0.254<sip%3A12129887777 at 192.168.0.254>
>;tag=as5b6db7a2
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: 23c4c49b329104d31ad6822c02cb84f5 at 192.168.0.254
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.0
Date: Wed, 12 May 2010 14:32:18 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 814806874 814806874 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 17632 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 19:33:06] --- (14 headers 13 lines) ---
[May 12 19:33:06] Sending to 192.168.0.254 : 5060 (NAT)
[May 12 19:33:06] Using INVITE request as basis request -
23c4c49b329104d31ad6822c02cb84f5 at 192.168.0.254
[May 12 19:33:06] Found no matching peer or user for '192.168.0.254:5060'
[May 12 19:33:06] Found RTP audio format 0
[May 12 19:33:06] Found RTP audio format 3
[May 12 19:33:06] Found RTP audio format 101
[May 12 19:33:06] Peer audio RTP is at port 192.168.0.254:17632
[May 12 19:33:06] Found description format PCMU for ID 0
[May 12 19:33:06] Found description format GSM for ID 3
[May 12 19:33:06] Found description format telephone-event for ID 101
[May 12 19:33:06] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer -
audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
[May 12 19:33:06] Non-codec capabilities (dtmf): us - 0x1 (telephone-event),
peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May 12 19:33:06] Peer audio RTP is at port 192.168.0.254:17632
[May 12 19:33:06] Looking for 17185594743 in default (domain
nasir.server.com)
[May 12 19:33:06] WARNING[4113]: chan_sip.c:3930 sip_new: setting callerid
number to 12129887777
[May 12 19:33:06] list_route: hop:
<sip:12129887777 at 192.168.0.254<sip%3A12129887777 at 192.168.0.254>
>
[May 12 19:33:06]
<--- Transmitting (NAT) to 192.168.0.254:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.254:5060
;branch=z9hG4bK56e3b44a;received=192.168.0.254;rport=5060
From: "caller" <sip:12129887777 at 192.168.0.254<sip%3A12129887777 at 192.168.0.254>
>;tag=as5b6db7a2
To: <sip:17185594743 at nasir.server.com <sip%3A17185594743 at nasir.server.com>>
Call-ID: 23c4c49b329104d31ad6822c02cb84f5 at 192.168.0.254
CSeq: 102 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 7:26 PM, Nasir Javaid <nasirjavaidnasir at gmail.com>wrote:

> 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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