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Rafael Visser

2012-Aug-26 21:05 UTC

[asterisk-users] the lenght of the uri affects on dialplan?

Hi Gurus..
I use asterisk for just for ivr.
My issue is that when the switch changes it's host name from
MSSASU1.MYDOMAIN to MSSSASU1.MYDOMAiN.COM or MSSSASU1.MYDOMAiN.COM.PY the call
is rejected with "No matching peer" and the
"handle_request_invite: Sending fake auth rejection for device x". It
doesn't match it's own default context.

Also, it has somethig to do with the numbers of digits of the dialed number. Few
digits works ok, 14 to more works wrong.
Do you know what am i missing?
Thanks in advance.









Debug with long hostname (B is considered as an '*')
===============================<--- SIP read from TCP:10.146.9.70:6240
--->
INVITE sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone SIP/2.0
From: <sip:971200152 at MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695
To: <sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone>
Max-Forwards: 70
Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096
Call-ID: 9caX8060616182201-AAAABOPA- at MSSASU1.MYDOMAIN.COM.PY
CSeq: 7313 INVITE
P-Asserted-Identity: <sip:971200152 at MSSASU1.MYDOMAIN.COM.PY;user=phone>
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
P-Charging-Vector:
icid-value=6510081000-0826-16155907;icid-generated-at=MSSASU1.MYDOMAIN.COM.PY;orig-ioi=MSSASU1.MYDOMAIN.COM.PY
Supported: 100rel
Content-Type: application/sdp
Contact: <sip:MSSASU1.MYDOMAIN.COM.PY:5060;transport=UDP>
Content-Length: 414

v=0
o=- 7530078 7530078 IN IP4 MSSASU1.MYDOMAIN.COM.PY
s=-
t=0 0
a=sendrecv
m=audio 13802 RTP/AVP 8 96 18 97
c=IN IP4 10.143.1.67
b=RR:0
b=RS:0
a=rtpmap:8 PCMA/8000
a=rtpmap:96 AMR/8000
a=fmtp:96
mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=maxptime:40
<------------->
--- (15 headers 17 lines) ---
Sending to 10.146.9.70:5060 (no NAT)
Using INVITE request as basis request - 9caX8060616182201-AAAABOPA- at
MSSASU1.MYDOMAIN.COM.PY
################
No matching peer for '971200152' from '10.146.9.70:6240'
[Aug 26 16:15:55] NOTICE[6873]: chan_sip.c:21975 handle_request_invite: Sending
fake auth rej
ection for device <sip:971200152 at
MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695
#################
<--- Reliably Transmitting (no NAT) to 10.146.9.70:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP
MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096;received=10.146.9.70
From: <sip:971200152 at MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695
To: <sip:B56510123456789012345 at
SISIVR03.MYDOMAIN.COM.PY;user=phone>;tag=as4cfd0d54
Call-ID: 9caX8060616182201-AAAABOPA- at MSSASU1.MYDOMAIN.COM.PY
CSeq: 7313 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="35ff0feb"
Content-Length: 0




Short hostname on switch
==============Connected to Asterisk 1.8.7.0 currently running on fdosis-ims1
(pid = 25430)
fdosis-ims1*CLI> core set verbose 1
Verbosity was 0 and is now 1

<--- SIP read from UDP:10.146.9.70:5060 --->
INVITE sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone SIP/2.0
From: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>;tag=0046120455
To: <sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone>
Max-Forwards: 70
Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982
Call-ID: qDaQ1240646182201-AAAAAKDE- at MSSASU1.MYDOMAIN
CSeq: 14481 INVITE
P-Asserted-Identity: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>
Accept: application/sdp
llow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
P-Charging-Vector:
icid-value=A6D6B81000-0826-16440101;icid-generated-at=MSSASU1.MYDOMAIN;orig-ioi=MSSASU1.MYDOMAIN.COM.PY
Supported: 100rel
Content-Type: application/sdp
Contact: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP>
Content-Length: 407

v=0
o=- 8986991 8986991 IN IP4 MSSASU1.MYDOMAIN
s=-
t=0 0
a=sendrecv
m=audio 30838 RTP/AVP 8 96 18 97
c=IN IP4 10.143.1.68
b=RR:0
b=RS:0
a=rtpmap:8 PCMA/8000
a=rtpmap:96 AMR/8000
a=fmtp:96
mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=maxptime:40
<------------->
--- (15 headers 17 lines) ---
Sending to 10.146.9.70:5060 (no NAT)
Using INVITE request as basis request - qDaQ1240646182201-AAAAAKDE- at
MSSASU1.MYDOMAIN
Found peer 'sip.ericsson' for '971200152' from 10.146.9.70:5060
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 18
Found RTP audio format 97
Found audio description format PCMA for ID 8
Found unknown media description format AMR for ID 96
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 97
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1
(telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 10.143.1.68:30838
Looking for B56510123456789012345 in incoming-sip-ericsson (domain
SISIVR03.MYDOMAIN.COM.PY)
list_route: hop: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP>

<--- Transmitting (no NAT) to 10.146.9.70:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982;received=10.146.9.70
From: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>;tag=0046120455
To: <sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone>
Call-ID: qDaQ1240646182201-AAAAAKDE- at MSSASU1.MYDOMAIN
CSeq: 14481 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:B56510123456789012345 at 10.146.9.132:5060>
Content-Length: 0


 		 	   		  
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Faisal Hanif

2012-Aug-26 22:42 UTC

head link

[asterisk-users] the lenght of the uri affects on dialplan?

mention the complete scnario and your sip.conf.

Regards,

Faisal 
(sent from phone)

Rafael Visser <rafael_visser at hotmail.com> wrote:
>
>Hi Gurus..
>I use asterisk for just for ivr.
>My issue is that when the switch changes it's host name from
MSSASU1.MYDOMAIN to MSSSASU1.MYDOMAiN.COM or MSSSASU1.MYDOMAiN.COM.PY the call
is rejected with "No matching peer" and the
"handle_request_invite: Sending fake auth rejection for device x". It
doesn't match it's own default context.
>
>Also, it has somethig to do with the numbers of digits of the dialed number.
Few digits works ok, 14 to more works wrong.
>Do you know what am i missing?
>Thanks in advance.
>
>
>
>
>
>
>
>
>
>Debug with long hostname (B is considered as an '*')
>===============================><--- SIP read from
TCP:10.146.9.70:6240 --->
>INVITE sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone
SIP/2.0
>From: <sip:971200152 at
MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695
>To: <sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone>
>Max-Forwards: 70
>Via: SIP/2.0/TCP
MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096
>Call-ID: 9caX8060616182201-AAAABOPA- at MSSASU1.MYDOMAIN.COM.PY
>CSeq: 7313 INVITE
>P-Asserted-Identity: <sip:971200152 at
MSSASU1.MYDOMAIN.COM.PY;user=phone>
>Accept: application/sdp
>Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
>P-Charging-Vector:
icid-value=6510081000-0826-16155907;icid-generated-at=MSSASU1.MYDOMAIN.COM.PY;orig-ioi=MSSASU1.MYDOMAIN.COM.PY
>Supported: 100rel
>Content-Type: application/sdp
>Contact: <sip:MSSASU1.MYDOMAIN.COM.PY:5060;transport=UDP>
>Content-Length: 414
>
>v=0
>o=- 7530078 7530078 IN IP4 MSSASU1.MYDOMAIN.COM.PY
>s=-
>t=0 0
>a=sendrecv
>m=audio 13802 RTP/AVP 8 96 18 97
>c=IN IP4 10.143.1.67
>b=RR:0
>b=RS:0
>a=rtpmap:8 PCMA/8000
>a=rtpmap:96 AMR/8000
>a=fmtp:96
mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
>a=rtpmap:18 G729/8000
>a=fmtp:18 annexb=yes
>a=rtpmap:97 telephone-event/8000
>a=fmtp:97 0-15
>a=maxptime:40
><------------->
>--- (15 headers 17 lines) ---
>Sending to 10.146.9.70:5060 (no NAT)
>Using INVITE request as basis request - 9caX8060616182201-AAAABOPA- at
MSSASU1.MYDOMAIN.COM.PY
>################
>No matching peer for '971200152' from '10.146.9.70:6240'
>[Aug 26 16:15:55] NOTICE[6873]: chan_sip.c:21975 handle_request_invite:
Sending fake auth rej
>ection for device <sip:971200152 at
MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695
>#################
><--- Reliably Transmitting (no NAT) to 10.146.9.70:5060 --->
>SIP/2.0 401 Unauthorized
>Via: SIP/2.0/TCP
MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096;received=10.146.9.70
>From: <sip:971200152 at
MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695
>To: <sip:B56510123456789012345 at
SISIVR03.MYDOMAIN.COM.PY;user=phone>;tag=as4cfd0d54
>Call-ID: 9caX8060616182201-AAAABOPA- at MSSASU1.MYDOMAIN.COM.PY
>CSeq: 7313 INVITE
>Server: Asterisk PBX 1.8.7.0
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
>Supported: replaces, timer
>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="35ff0feb"
>Content-Length: 0
>
>
>
>
>Short hostname on switch
>==============>Connected to Asterisk 1.8.7.0 currently running on
fdosis-ims1 (pid = 25430)
>fdosis-ims1*CLI> core set verbose 1
>Verbosity was 0 and is now 1
>
><--- SIP read from UDP:10.146.9.70:5060 --->
>INVITE sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone
SIP/2.0
>From: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>;tag=0046120455
>To: <sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone>
>Max-Forwards: 70
>Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982
>Call-ID: qDaQ1240646182201-AAAAAKDE- at MSSASU1.MYDOMAIN
>CSeq: 14481 INVITE
>P-Asserted-Identity: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>
>Accept: application/sdp
>llow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
>P-Charging-Vector:
icid-value=A6D6B81000-0826-16440101;icid-generated-at=MSSASU1.MYDOMAIN;orig-ioi=MSSASU1.MYDOMAIN.COM.PY
>Supported: 100rel
>Content-Type: application/sdp
>Contact: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP>
>Content-Length: 407
>
>v=0
>o=- 8986991 8986991 IN IP4 MSSASU1.MYDOMAIN
>s=-
>t=0 0
>a=sendrecv
>m=audio 30838 RTP/AVP 8 96 18 97
>c=IN IP4 10.143.1.68
>b=RR:0
>b=RS:0
>a=rtpmap:8 PCMA/8000
>a=rtpmap:96 AMR/8000
>a=fmtp:96
mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
>a=rtpmap:18 G729/8000
>a=fmtp:18 annexb=yes
>a=rtpmap:97 telephone-event/8000
>a=fmtp:97 0-15
>a=maxptime:40
><------------->
>--- (15 headers 17 lines) ---
>Sending to 10.146.9.70:5060 (no NAT)
>Using INVITE request as basis request - qDaQ1240646182201-AAAAAKDE- at
MSSASU1.MYDOMAIN
>Found peer 'sip.ericsson' for '971200152' from
10.146.9.70:5060
>Found RTP audio format 8
>Found RTP audio format 96
>Found RTP audio format 18
>Found RTP audio format 97
>Found audio description format PCMA for ID 8
>Found unknown media description format AMR for ID 96
>Found audio description format G729 for ID 18
>Found audio description format telephone-event for ID 97
>Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108
(alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
>Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1
(telephone-event|), combined - 0x0 (nothing)
>Peer audio RTP is at port 10.143.1.68:30838
>Looking for B56510123456789012345 in incoming-sip-ericsson (domain
SISIVR03.MYDOMAIN.COM.PY)
>list_route: hop: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP>
>
><--- Transmitting (no NAT) to 10.146.9.70:5060 --->
>SIP/2.0 100 Trying
>Via: SIP/2.0/UDP
MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982;received=10.146.9.70
>From: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>;tag=0046120455
>To: <sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone>
>Call-ID: qDaQ1240646182201-AAAAAKDE- at MSSASU1.MYDOMAIN
>CSeq: 14481 INVITE
>Server: Asterisk PBX 1.8.7.0
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
>Supported: replaces, timer
>Contact: <sip:B56510123456789012345 at 10.146.9.132:5060>
>Content-Length: 0
>
>
> 		 	   		  
>--
>_____________________________________________________________________
>-- Bandwidth and Colocation Provided by http://www.api-digital.com --
>New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

shengbin zhu

2012-Aug-27 03:28 UTC

head link

[asterisk-users] the lenght of the uri affects on dialplan?

Hi,Rafael

I found some difference in INVITE message when the switch is long name, see
inline mark in red.
I guess the root cause is the server
MSSSASU1.MYDOMAiN.COM.PY<http://msssasu1.mydomain.com.py/>  doesn't
support tcp.

Best Regards

Ben

2012/8/27 Rafael Visser <rafael_visser at hotmail.com>
>  Hi Gurus..
> I use asterisk for just for ivr.
> My issue is that when the switch changes it's host name from
> MSSASU1.MYDOMAIN to MSSSASU1.MYDOMAiN.COM or MSSSASU1.MYDOMAiN.COM.PY the
> call is rejected with "No matching peer" and the
"handle_request_invite:
> Sending fake auth rejection for device x". It doesn't match
it's own
> default context.
>
> Also, it has somethig to do with the numbers of digits of the dialed
> number. Few digits works ok, 14 to more works wrong.
> Do you know what am i missing?
> Thanks in advance.
>
>
>
>
>
>
>
>
>
> Debug with long hostname (B is considered as an '*')
> ===============================> <--- SIP read from
TCP:10.146.9.70:6240 --->
> INVITE sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone
> SIP/2.0
> From: <sip:971200152 at
MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695
> To: <sip:B56510123456789012345 at
SISIVR03.MYDOMAIN.COM.PY;user=phone>
> Max-Forwards: 70
> Via: SIP/2.0/TCP
MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096
>  [Ben] --here indicate the SIP use TCP as transport protocol, but the
> Contact header field is UDP protocol, they are mismatch, you can check the
> same INVITE message when the switch is short name,[ they are matched.
> Call-ID: 9caX8060616182201-AAAABOPA- at MSSASU1.MYDOMAIN.COM.PY
> CSeq: 7313 INVITE
> P-Asserted-Identity: <sip:971200152 at
MSSASU1.MYDOMAIN.COM.PY;user=phone>
> Accept: application/sdp
> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
> P-Charging-Vector:
icid-value=6510081000-0826-16155907;icid-generated-at>
MSSASU1.MYDOMAIN.COM.PY;orig-ioi=MSSASU1.MYDOMAIN.COM.PY
> Supported: 100rel
> Content-Type: application/sdp
> Contact: <sip:MSSASU1.MYDOMAIN.COM.PY:5060;transport=UDP>
> Content-Length: 414
>
> v=0
> o=- 7530078 7530078 IN IP4 MSSASU1.MYDOMAIN.COM.PY
> s=-
> t=0 0
> a=sendrecv
> m=audio 13802 RTP/AVP 8 96 18 97
> c=IN IP4 10.143.1.67
> b=RR:0
> b=RS:0
> a=rtpmap:8 PCMA/8000
> a=rtpmap:96 AMR/8000
> a=fmtp:96
>
mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
> a=rtpmap:97 telephone-event/8000
> a=fmtp:97 0-15
> a=maxptime:40
> <------------->
> --- (15 headers 17 lines) ---
> Sending to 10.146.9.70:5060 (no NAT)
> Using INVITE request as basis request -
> 9caX8060616182201-AAAABOPA- at MSSASU1.MYDOMAIN.COM.PY
> ################
> No matching peer for '971200152' from '10.146.9.70:6240'
> [Aug 26 16:15:55] NOTICE[6873]: chan_sip.c:21975 handle_request_invite:
> Sending fake auth rej
> ection for device <sip:971200152 at MSSASU1.MYDOMAIN.COM.PY
> ;user=phone>;tag=3016589695
> #################
> <--- Reliably Transmitting (no NAT) to 10.146.9.70:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060
> ;branch=z9hG4bK00000035391821780096;received=10.146.9.70
> From: <sip:971200152 at
MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695
> To: <sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY
> ;user=phone>;tag=as4cfd0d54
> Call-ID: 9caX8060616182201-AAAABOPA- at MSSASU1.MYDOMAIN.COM.PY
> CSeq: 7313 INVITE
> Server: Asterisk PBX 1.8.7.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="35ff0feb"
> Content-Length: 0
>
>
>
>
> Short hostname on switch
> ==============> Connected to Asterisk 1.8.7.0 currently running on
fdosis-ims1 (pid > 25430)
> fdosis-ims1*CLI> core set verbose 1
> Verbosity was 0 and is now 1
>
> <--- SIP read from UDP:10.146.9.70:5060 --->
> INVITE sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone
> SIP/2.0
> From: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>;tag=0046120455
> To: <sip:B56510123456789012345 at
SISIVR03.MYDOMAIN.COM.PY;user=phone>
> Max-Forwards: 70
> Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982
> Call-ID: qDaQ1240646182201-AAAAAKDE- at MSSASU1.MYDOMAIN
> CSeq: 14481 INVITE
> P-Asserted-Identity: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>
> Accept: application/sdp
> llow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
> P-Charging-Vector:
>
icid-value=A6D6B81000-0826-16440101;icid-generated-at=MSSASU1.MYDOMAIN;orig-ioi>
MSSASU1.MYDOMAIN.COM.PY
> Supported: 100rel
> Content-Type: application/sdp
> Contact: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP>
> Content-Length: 407
>
> v=0
> o=- 8986991 8986991 IN IP4 MSSASU1.MYDOMAIN
> s=-
> t=0 0
> a=sendrecv
> m=audio 30838 RTP/AVP 8 96 18 97
> c=IN IP4 10.143.1.68
> b=RR:0
> b=RS:0
> a=rtpmap:8 PCMA/8000
> a=rtpmap:96 AMR/8000
> a=fmtp:96
>
mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
> a=rtpmap:97 telephone-event/8000
> a=fmtp:97 0-15
> a=maxptime:40
> <------------->
> --- (15 headers 17 lines) ---
> Sending to 10.146.9.70:5060 (no NAT)
> Using INVITE request as basis request -
> qDaQ1240646182201-AAAAAKDE- at MSSASU1.MYDOMAIN
> Found peer 'sip.ericsson' for '971200152' from
10.146.9.70:5060
> Found RTP audio format 8
> Found RTP audio format 96
> Found RTP audio format 18
> Found RTP audio format 97
> Found audio description format PCMA for ID 8
> Found unknown media description format AMR for ID 96
> Found audio description format G729 for ID 18
> Found audio description format telephone-event for ID 97
> Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108
> (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1
> (telephone-event|), combined - 0x0 (nothing)
> Peer audio RTP is at port 10.143.1.68:30838
> Looking for B56510123456789012345 in incoming-sip-ericsson (domain
> SISIVR03.MYDOMAIN.COM.PY)
> list_route: hop: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP>
>
> <--- Transmitting (no NAT) to 10.146.9.70:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
>
MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982;received=10.146.9.70
> From: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>;tag=0046120455
> To: <sip:B56510123456789012345 at
SISIVR03.MYDOMAIN.COM.PY;user=phone>
> Call-ID: qDaQ1240646182201-AAAAAKDE- at MSSASU1.MYDOMAIN
> CSeq: 14481 INVITE
> Server: Asterisk PBX 1.8.7.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> Contact: <sip:B56510123456789012345 at 10.146.9.132:5060>
> Content-Length: 0
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
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asterisk users - Aug 2012 - the lenght of the uri affects on dialplan?

[asterisk-users] the lenght of the uri affects on dialplan?

[asterisk-users] the lenght of the uri affects on dialplan?

[asterisk-users] the lenght of the uri affects on dialplan?