Re: [asterisk-users] Asterisk Inbound Problem

2007-02-21 Thread Arun Kumar 

My service provider only supports g729 and I tried what you have mentioned
here but same thing is happening here. Is there any why that I can see which
codec my service provider is pushing when I'm receiving call on my asterisk
server. When call comes comes to my server and then I type show g729 it
shows 0/0 out of 15 lic.

thanks
arun

On 2/21/07, Mike Lynchfield [EMAIL PROTECTED] wrote:


Well, could be the fact provider not pushing as g729 or someting else.

Can you set debug 999 and set verbose 999
then redump that ? you are missing the before the answer part also..

Also try G711 first then work your way to other codecs


On 2/20/07, Rajeev Natarajan [EMAIL PROTECTED] wrote:

 Am working with Arun on this project - here's a longer description of
 the problem:

 We've been fighting with our service provider on this issue - we seem to
 be getting a BYE just after we receive an ACK. They claim that it is an
 asterisk issue! The service provider provides only IP based authentication
 for inbound.

 We have used username-password based authentication with the same setup
 with *no problems*  whatsoever!

 If we configure an Audiocodes MEdia gateway to receive the calls, there
 is no issue - so there's something that asterisk is doing? or
 asterisk-Provider gateway combo?

 In our efforts to mask IP, I have used PROVIDER-IP for the IP of my
 service provider (host) and AsteriskIP to indicate my asterisk server

 sip.conf
 [PROVIDER]
 type=peer
 disallow=all
 allow=g729
 context=default
 host=
 fromuser=y.y.y.y
 port=5060
 insecure=very
 canreinvite=no
 nat=yes
 qualify=yes

 CLI output:

-- Executing Answer(SIP/PROVIDER-IP-b7a076a8, ) in new stack
 We're at 124.7.195.102 port 47698
 Adding codec 0x100 (g729) to SDP
 Reliably Transmitting (NAT) to PROVIDER-IP:5060:
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP
 
PROVIDER-IP:5060;branch=z9hG4bK6bd3121243ee9f936c4aeb96d6785b7a;received=PROVIDER-IP

 From: sip:[EMAIL PROTECTED];tag=3380976385-794612
 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1 INVITE
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
 Contact: sip:8009422419@'AsteriskIP'
 Content-Type: application/sdp
 Content-Length: 183

 v=0
 o=root 2172 2172 IN IP4 AsteriskIP
 s=session
 c=IN IP4 AsteriskIP
 t=0 0
 m=audio 47698 RTP/AVP 18
 a=rtpmap:18 G729/8000
 a=fmtp:18 annexb=no
 a=silenceSupp:off - - - -

 ---

  -- Executing Playback(SIP/PROVIDER-IP-b7a076a8, park) in new stack
 -- Playing 'park' (language 'en')
 AstSQL*CLI
 -- SIP read from PROVIDER-IP:5060:
 ACK sip:[EMAIL PROTECTED] SIP/2.0
 Max-Forwards: 5
 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
 From: sip:[EMAIL PROTECTED];tag=3380976385-794612
 Contact: sip:[EMAIL PROTECTED]:5060
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1 ACK
 Via: SIP/2.0/UDP 
221.135.102.100:5060;branch=z9hG4bK02505a1dcc5937d9a648eebc0052b422
 Content-Length: 0


 --- (9 headers 0 lines) ---
 AstSQL*CLI
 -- SIP read from PROVIDER-IP:5060:
 BYE sip:[EMAIL PROTECTED] SIP/2.0
 Max-Forwards: 5
 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
 From: sip:[EMAIL PROTECTED];tag=3380976385-794612
 Contact: sip:[EMAIL PROTECTED]:5060
 Call-ID: [EMAIL PROTECTED]
 CSeq: 2 BYE
 Via: SIP/2.0/UDP 221.135.102.100:5060
 ;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f
 Content-Length: 0


 --- (9 headers 0 lines) ---
 Sending to PROVIDER-IP : 5060 (NAT)
 Transmitting (NAT) to PROVIDER-IP:5060:
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP
 
PROVIDER-IP:5060;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f;received=PROVIDER-IP

 From: sip:[EMAIL PROTECTED];tag=3380976385-794612
 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
 Call-ID: [EMAIL PROTECTED]
 CSeq: 2 BYE
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
 Contact: sip:[EMAIL PROTECTED]
 Content-Length: 0

 


 The following is an ngrep of the traffic for an inbound call - 'U' marks
 the begin of the packet grabbed.


 U PROVIDER-IP:5060 - AsteriskIP:5060
   INVITE sip:800942@AsteriskIP SIP/2.0..Max-Forwards:
 5..Session-Expires: 3600;Refresher=uac..Suppor ted: timer..To: 
 sip:[EMAIL PROTECTED]:5060..From:
 sip:PROVIDER-IP;tag=3380960452-790279..Co ntact:
 sip:PROVIDER-IP:5060..Remote-Party-Id:
 sip:PROVIDER-IP;party=calling;screen=no;privacy =off..Call-ID:
 [EMAIL PROTECTED]: 1
 INVITE..Via: SIP/2.0/UDP 221. 
135.102.100:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4..Allow-Events:
 telephone-event..Content-T ype: application/sdp..Content-Length:
 206v=0..o=nextone-msw1 1774 4816 IN IP4 PROVIDER-IP..s=sip call..c=IN
 IP4 PROV-IP-2..t=0 0..m=audio 18932 RTP/AVP 18 19..a=ptime:20..a=rtpmap:19
 CN/8000..a=fm tp:18 annexb=yes..a=rtpmap:18 G729/8000..


 #
 U AsteriskIP:5060 - PROVIDER-IP:5060
   SIP/2.0 100 Trying..Via: SIP/2.0/UDP

Re: [asterisk-users] Asterisk Inbound Problem

2007-02-20 Thread Rajeev Natarajan 

Am working with Arun on this project - here's a longer description of the
problem:

We've been fighting with our service provider on this issue - we seem to be
getting a BYE just after we receive an ACK. They claim that it is an
asterisk issue! The service provider provides only IP based authentication
for inbound.

We have used username-password based authentication with the same setup with
*no problems*  whatsoever!

If we configure an Audiocodes MEdia gateway to receive the calls, there is
no issue - so there's something that asterisk is doing? or asterisk-Provider
gateway combo?

In our efforts to mask IP, I have used PROVIDER-IP for the IP of my service
provider (host) and AsteriskIP to indicate my asterisk server

sip.conf
[PROVIDER]
type=peer
disallow=all
allow=g729
context=default
host=
fromuser=y.y.y.y
port=5060
insecure=very
canreinvite=no
nat=yes
qualify=yes

CLI output:

  -- Executing Answer(SIP/PROVIDER-IP-b7a076a8, ) in new stack
We're at 124.7.195.102 port 47698
Adding codec 0x100 (g729) to SDP
Reliably Transmitting (NAT) to PROVIDER-IP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
PROVIDER-IP:5060;branch=z9hG4bK6bd3121243ee9f936c4aeb96d6785b7a;received=PROVIDER-IP
From: sip:[EMAIL PROTECTED];tag=3380976385-794612
To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:8009422419@'AsteriskIP'
Content-Type: application/sdp
Content-Length: 183

v=0
o=root 2172 2172 IN IP4 AsteriskIP
s=session
c=IN IP4 AsteriskIP
t=0 0
m=audio 47698 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -

---

-- Executing Playback(SIP/PROVIDER-IP-b7a076a8, park) in new stack
   -- Playing 'park' (language 'en')
AstSQL*CLI
-- SIP read from PROVIDER-IP:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Max-Forwards: 5
To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
From: sip:[EMAIL PROTECTED];tag=3380976385-794612
Contact: sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
Via: SIP/2.0/UDP 221.135.102.100:5060
;branch=z9hG4bK02505a1dcc5937d9a648eebc0052b422
Content-Length: 0


--- (9 headers 0 lines) ---
AstSQL*CLI
-- SIP read from PROVIDER-IP:5060:
BYE sip:[EMAIL PROTECTED] SIP/2.0
Max-Forwards: 5
To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
From: sip:[EMAIL PROTECTED];tag=3380976385-794612
Contact: sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 2 BYE
Via: SIP/2.0/UDP 221.135.102.100:5060
;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f
Content-Length: 0


--- (9 headers 0 lines) ---
Sending to PROVIDER-IP : 5060 (NAT)
Transmitting (NAT) to PROVIDER-IP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
PROVIDER-IP:5060;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f;received=PROVIDER-IP
From: sip:[EMAIL PROTECTED];tag=3380976385-794612
To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
Call-ID: [EMAIL PROTECTED]
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


The following is an ngrep of the traffic for an inbound call - 'U' marks the
begin of the packet grabbed.


U PROVIDER-IP:5060 - AsteriskIP:5060
 INVITE sip:800942@AsteriskIP SIP/2.0..Max-Forwards:
5..Session-Expires: 3600;Refresher=uac..Suppor ted: timer..To: 
sip:[EMAIL PROTECTED]:5060..From:
sip:PROVIDER-IP;tag=3380960452-790279..Co ntact:
sip:PROVIDER-IP:5060..Remote-Party-Id:
sip:PROVIDER-IP;party=calling;screen=no;privacy =off..Call-ID:
[EMAIL PROTECTED]: 1 INVITE..Via:
SIP/2.0/UDP 221.
135.102.100:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4..Allow-Events:
telephone-event..Content-T ype: application/sdp..Content-Length:
206v=0..o=nextone-msw1 1774 4816 IN IP4 PROVIDER-IP..s=sip call..c=IN
IP4 PROV-IP-2..t=0 0..m=audio 18932 RTP/AVP 18 19..a=ptime:20..a=rtpmap:19
CN/8000..a=fm tp:18 annexb=yes..a=rtpmap:18 G729/8000..


#
U AsteriskIP:5060 - PROVIDER-IP:5060
 SIP/2.0 100 Trying..Via: SIP/2.0/UDP
PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4;
received=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To:
 sip:[EMAIL PROTECTED] 11.2:5060..Call-ID:
[EMAIL PROTECTED]: 1
INVITE..User-Agent: Ast erisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY..Contact: 
sip:[EMAIL PROTECTED]..Content-Length:
0


#
U AsteriskIP:5060 - PROVIDER-IP:5060
 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4
;received=PROVIDER-IP..From:
sip:PROVIDER-IP;tag=3380960452-790279..To: 
sip:[EMAIL PROTECTED]:5060;tag=as78bcde29..Call-ID:
[EMAIL PROTECTED]: 1 INVITE.
.User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY..Contact:  sip:800942@AsteriskIP..Content-Length:
0



#
U

Re: [asterisk-users] Asterisk Inbound Problem

2007-02-20 Thread Mike Lynchfield 

Well, could be the fact provider not pushing as g729 or someting else.

Can you set debug 999 and set verbose 999
then redump that ? you are missing the before the answer part also..

Also try G711 first then work your way to other codecs


On 2/20/07, Rajeev Natarajan [EMAIL PROTECTED] wrote:


Am working with Arun on this project - here's a longer description of the
problem:

We've been fighting with our service provider on this issue - we seem to
be getting a BYE just after we receive an ACK. They claim that it is an
asterisk issue! The service provider provides only IP based authentication
for inbound.

We have used username-password based authentication with the same setup
with *no problems*  whatsoever!

If we configure an Audiocodes MEdia gateway to receive the calls, there is
no issue - so there's something that asterisk is doing? or asterisk-Provider
gateway combo?

In our efforts to mask IP, I have used PROVIDER-IP for the IP of my
service provider (host) and AsteriskIP to indicate my asterisk server

sip.conf
[PROVIDER]
type=peer
disallow=all
allow=g729
context=default
host=
fromuser=y.y.y.y
port=5060
insecure=very
canreinvite=no
nat=yes
qualify=yes

CLI output:

   -- Executing Answer(SIP/PROVIDER-IP-b7a076a8, ) in new stack
We're at 124.7.195.102 port 47698
Adding codec 0x100 (g729) to SDP
Reliably Transmitting (NAT) to PROVIDER-IP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
PROVIDER-IP:5060;branch=z9hG4bK6bd3121243ee9f936c4aeb96d6785b7a;received=PROVIDER-IP

From: sip:[EMAIL PROTECTED];tag=3380976385-794612
To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:8009422419@'AsteriskIP'
Content-Type: application/sdp
Content-Length: 183

v=0
o=root 2172 2172 IN IP4 AsteriskIP
s=session
c=IN IP4 AsteriskIP
t=0 0
m=audio 47698 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -

---

 -- Executing Playback(SIP/PROVIDER-IP-b7a076a8, park) in new stack
-- Playing 'park' (language 'en')
AstSQL*CLI
-- SIP read from PROVIDER-IP:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Max-Forwards: 5
To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
From: sip:[EMAIL PROTECTED];tag=3380976385-794612
Contact: sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
Via: SIP/2.0/UDP 
221.135.102.100:5060;branch=z9hG4bK02505a1dcc5937d9a648eebc0052b422
Content-Length: 0


--- (9 headers 0 lines) ---
AstSQL*CLI
-- SIP read from PROVIDER-IP:5060:
BYE sip:[EMAIL PROTECTED] SIP/2.0
Max-Forwards: 5
To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
From: sip:[EMAIL PROTECTED];tag=3380976385-794612
Contact: sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 2 BYE
Via: SIP/2.0/UDP 221.135.102.100:5060
;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f
Content-Length: 0


--- (9 headers 0 lines) ---
Sending to PROVIDER-IP : 5060 (NAT)
Transmitting (NAT) to PROVIDER-IP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
PROVIDER-IP:5060;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f;received=PROVIDER-IP

From: sip:[EMAIL PROTECTED];tag=3380976385-794612
To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855
Call-ID: [EMAIL PROTECTED]
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0



The following is an ngrep of the traffic for an inbound call - 'U' marks
the begin of the packet grabbed.


U PROVIDER-IP:5060 - AsteriskIP:5060
  INVITE sip:800942@AsteriskIP SIP/2.0..Max-Forwards:
5..Session-Expires: 3600;Refresher=uac..Suppor ted: timer..To: 
sip:[EMAIL PROTECTED]:5060..From:
sip:PROVIDER-IP;tag=3380960452-790279..Co ntact:
sip:PROVIDER-IP:5060..Remote-Party-Id:
sip:PROVIDER-IP;party=calling;screen=no;privacy =off..Call-ID:
[EMAIL PROTECTED]: 1 INVITE..Via:
SIP/2.0/UDP 221. 
135.102.100:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4..Allow-Events:
telephone-event..Content-T ype: application/sdp..Content-Length:
206v=0..o=nextone-msw1 1774 4816 IN IP4 PROVIDER-IP..s=sip call..c=IN
IP4 PROV-IP-2..t=0 0..m=audio 18932 RTP/AVP 18 19..a=ptime:20..a=rtpmap:19
CN/8000..a=fm tp:18 annexb=yes..a=rtpmap:18 G729/8000..


#
U AsteriskIP:5060 - PROVIDER-IP:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP
PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4;
received=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To:
 sip:[EMAIL PROTECTED] 11.2:5060..Call-ID:
[EMAIL PROTECTED]: 1
INVITE..User-Agent: Ast erisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY..Contact:  sip:[EMAIL PROTECTED]..Content-Length:
0


#
U AsteriskIP:5060 - PROVIDER-IP:5060
  SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4

[asterisk-users] Asterisk Inbound Problem

2007-02-19 Thread Arun Kumar 

HI

I've configred an Incoming DID in my asterisk and when I call from outside I
see call is coming to my Asterisk server and then from asterisk it rings on
a particulat exten but when I pickup the call the call get disconnect
immediate and on the other end it keep trying (ringing).

here is my exten.conf:

exten = _80.,1,Answer
exten = _80.,2,Dial(IAX2/2001)

did starts with 80 and any call comes for my number they are sending to my
asterisk IP.

thanks
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Re: [asterisk-users] Asterisk Inbound Problem

2007-02-19 Thread Mark Phillips 
Without seeing your config files my guess would be that this is
something to do with a bad codec negotiation.

I'd bet that your IAX phone is using ulaw and your DID provider is using
something else like G729.

Mark

On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote:
 HI
 
 I've configred an Incoming DID in my asterisk and when I call from
 outside I see call is coming to my Asterisk server and then from
 asterisk it rings on a particulat exten but when I pickup the call the
 call get disconnect immediate and on the other end it keep trying
 (ringing). 
 
 here is my exten.conf:
 
 exten = _80.,1,Answer
 exten = _80.,2,Dial(IAX2/2001)
 
 did starts with 80 and any call comes for my number they are sending
 to my asterisk IP.
 
 thanks
 
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 asterisk-users mailing list
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Re: [asterisk-users] Asterisk Inbound Problem

2007-02-19 Thread Arun Kumar 

Instead of forwarding to IAX softphone if I'll play some music same thing is
happening in this case also.

On 2/20/07, Mark Phillips [EMAIL PROTECTED] wrote:


Without seeing your config files my guess would be that this is
something to do with a bad codec negotiation.

I'd bet that your IAX phone is using ulaw and your DID provider is using
something else like G729.

Mark

On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote:
 HI

 I've configred an Incoming DID in my asterisk and when I call from
 outside I see call is coming to my Asterisk server and then from
 asterisk it rings on a particulat exten but when I pickup the call the
 call get disconnect immediate and on the other end it keep trying
 (ringing).

 here is my exten.conf:

 exten = _80.,1,Answer
 exten = _80.,2,Dial(IAX2/2001)

 did starts with 80 and any call comes for my number they are sending
 to my asterisk IP.

 thanks

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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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