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George Pajari

2009-Apr-08 04:56 UTC

[asterisk-users] Call Pickup Works w/Linksys ATA, not with Cisco 7940G

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I have an Asterisk 1.4.18 with a mix of cordless phones connected using
Linksys SPA2102 ATAs and Cisco 7940G phones. Unit obtains SIP trunking
from an ITSP (server has no PCI boards).<br>
<br>
*8 Call Pickup works fine from any of the phones connected using the
Linksys SPA2102.<br>
<br>
*8 Call Pickup does not work from the Cisco 7940G phones
(chan_sip.c:13977 handle_request_invite: Nothing to pick up for
<a class="moz-txt-link-abbreviated"
href="mailto:000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211">000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211</a>)<br>
<br>
What could the difference be?<br>
<br>
Below you will find:<br>
&nbsp;(a) the "sip show peer nnn" for an ATA extension and a Cisco
extension<br>
&nbsp;(b) the SIP debug trace for (i) a successful call pickup from the ATA
and (ii) an unsuccessful call pickup from the Cisco<br>
<br>
Any light anyone can shed on the perplexing problem would be most
appreciated. I have a forehead-shaped dent in the wall that is growing
larger.<br>
<br>
<br>
<b><u>Linksys SPA2102 ATA "sip show peer
101":</u></b><br>
<pre>
* Name       : 101
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : numberplan-custom-1
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 1
  Pickupgroup  : 1
  Mailbox      : 101
  VM Extension : asterisk
  LastMsgsSent : 0/3
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "Dxx Gxxx" <604-123-4444>
  MaxCallBR    : 384 kbps
  Expire       : 1861
  Insecure     : no
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : 
  Addr->IP     : 192.168.0.205 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 101
  SIP Options  : replaces replace 
  Codecs       : 0x8000e (gsm|ulaw|alaw|h263)
  Codec Order  : (none)
  Auto-Framing:  No 
  Status       : OK (5 ms)
  Useragent    : Linksys/SPA2102-5.2.3
  Reg. Contact : <a class="moz-txt-link-freetext"
 href="sip:101@192.168.0.205:5060">sip:101@192.168.0.205:5060</a>
</pre>
<br>
<br>
<br>
<u><b>Cisco 7940G Phone "sip show peer 106":<br>
</b></u>
<pre>* Name       : 106
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : numberplan-custom-1
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 1
  Pickupgroup  : 1
  Mailbox      : 106
  VM Extension : asterisk
  LastMsgsSent : 3/1
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "Cxxx Nxxx" <6041234567>
  MaxCallBR    : 384 kbps
  Expire       : 247
  Insecure     : no
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : 
  Addr->IP     : 192.168.0.211 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 106
  SIP Options  : (none)
  Codecs       : 0x8000e (gsm|ulaw|alaw|h263)
  Codec Order  : (none)
  Auto-Framing:  No 
  Status       : OK (195 ms)
  Useragent    : Cisco-CP7940G/8.0
  Reg. Contact : <a class="moz-txt-link-freetext"
 href="sip:106@192.168.0.211:5060;transport=udp">sip:106@192.168.0.211:5060;transport=udp</a></pre>
<u><b><br>
<br>
<br>
Successful *8 Call Pickup (SIP Trace)<br>
<br>
</b></u>
<pre><--- SIP read from 192.168.0.205:5060 --->
INVITE <a class="moz-txt-link-freetext"
href="sip:*8@192.168.0.12">sip:*8@192.168.0.12</a> SIP/2.0
Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-e7b4459c
From: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.12"><sip:101@192.168.0.12></a>;tag=22b459c8b65178bco0
To: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>
Remote-Party-ID: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.12"><sip:101@192.168.0.12></a>;screen=yes;party=calling
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:94dc7b8-591d638c@192.168.0.205">94dc7b8-591d638c@192.168.0.205</a>
CSeq: 101 INVITE
Max-Forwards: 70
Contact: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.205:5060"><sip:101@192.168.0.205:5060></a>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.3
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 922981 922981 IN IP4 192.168.0.205
s=-
c=IN IP4 192.168.0.205
t=0 0
m=audio 16412 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (15 headers 20 lines) ---
Sending to 192.168.0.205 : 5060 (no NAT)
Using INVITE request as basis request - <a
class="moz-txt-link-abbreviated"
href="mailto:94dc7b8-591d638c@192.168.0.205">94dc7b8-591d638c@192.168.0.205</a>
Found peer '101'
tg2*CLI> 
<--- Reliably Transmitting (no NAT) to 192.168.0.205:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.0.205:5060;branch=z9hG4bK-e7b4459c;received=192.168.0.205
From: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.12"><sip:101@192.168.0.12></a>;tag=22b459c8b65178bco0
To: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>;tag=as4bf7113a
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:94dc7b8-591d638c@192.168.0.205">94dc7b8-591d638c@192.168.0.205</a>
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="637bf838"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '<a
class="moz-txt-link-abbreviated"
href="mailto:94dc7b8-591d638c@192.168.0.205">94dc7b8-591d638c@192.168.0.205</a>'
in 6400 ms (Method: INVITE)
tg2*CLI> 
<--- SIP read from 192.168.0.205:5060 --->
ACK <a class="moz-txt-link-freetext"
href="sip:*8@192.168.0.12">sip:*8@192.168.0.12</a> SIP/2.0
Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-e7b4459c
From: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.12"><sip:101@192.168.0.12></a>;tag=22b459c8b65178bco0
To: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>;tag=as4bf7113a
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:94dc7b8-591d638c@192.168.0.205">94dc7b8-591d638c@192.168.0.205</a>
CSeq: 101 ACK
Max-Forwards: 70
Contact: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.205:5060"><sip:101@192.168.0.205:5060></a>
User-Agent: Linksys/SPA2102-5.2.3
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
tg2*CLI> 
<--- SIP read from 192.168.0.205:5060 --->
INVITE <a class="moz-txt-link-freetext"
href="sip:*8@192.168.0.12">sip:*8@192.168.0.12</a> SIP/2.0
Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-4be14d40
From: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.12"><sip:101@192.168.0.12></a>;tag=22b459c8b65178bco0
To: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>
Remote-Party-ID: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.12"><sip:101@192.168.0.12></a>;screen=yes;party=calling
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:94dc7b8-591d638c@192.168.0.205">94dc7b8-591d638c@192.168.0.205</a>
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="101",realm="asterisk",nonce="637bf838",uri=<a
class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12">"sip:*8@192.168.0.12"</a>,algorithm=MD5,response="8064fac79e90815539b2ee1aa33df011"
Contact: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.205:5060"><sip:101@192.168.0.205:5060></a>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.3
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 922981 922981 IN IP4 192.168.0.205
s=-
c=IN IP4 192.168.0.205
t=0 0
m=audio 16412 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (16 headers 20 lines) ---
Sending to 192.168.0.205 : 5060 (no NAT)
Using INVITE request as basis request - <a
class="moz-txt-link-abbreviated"
href="mailto:94dc7b8-591d638c@192.168.0.205">94dc7b8-591d638c@192.168.0.205</a>
Found peer '101'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.205:16412
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
Found audio description format G729a for ID 18
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found unknown media description format NSE for ID 100
Got unsupported a:fmtp in SDP offer 
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer 
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x90d
(g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.205:16412
Looking for *8 in numberplan-custom-1 (domain 192.168.0.12)
list_route: hop: <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.205:5060"><sip:101@192.168.0.205:5060></a>
tg2*CLI> 
<--- Transmitting (no NAT) to 192.168.0.205:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.205:5060;branch=z9hG4bK-4be14d40;received=192.168.0.205
From: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.12"><sip:101@192.168.0.12></a>;tag=22b459c8b65178bco0
To: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:94dc7b8-591d638c@192.168.0.205">94dc7b8-591d638c@192.168.0.205</a>
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>
Content-Length: 0


<------------>
Audio is at 192.168.0.12 port 17634
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
tg2*CLI> 
<--- Reliably Transmitting (no NAT) to 192.168.0.205:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.205:5060;branch=z9hG4bK-4be14d40;received=192.168.0.205
From: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.12"><sip:101@192.168.0.12></a>;tag=22b459c8b65178bco0
To: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>;tag=as49b9fd26
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:94dc7b8-591d638c@192.168.0.205">94dc7b8-591d638c@192.168.0.205</a>
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1941 1941 IN IP4 192.168.0.12
s=session
c=IN IP4 192.168.0.12
t=0 0
m=audio 17634 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
tg2*CLI> 
<--- SIP read from 192.168.0.205:5060 --->
ACK <a class="moz-txt-link-freetext"
href="sip:*8@192.168.0.12">sip:*8@192.168.0.12</a> SIP/2.0
Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-58c05ba7
From: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.12"><sip:101@192.168.0.12></a>;tag=22b459c8b65178bco0
To: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>;tag=as49b9fd26
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:94dc7b8-591d638c@192.168.0.205">94dc7b8-591d638c@192.168.0.205</a>
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest
username="101",realm="asterisk",nonce="637bf838",uri=<a
class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12">"sip:*8@192.168.0.12"</a>,algorithm=MD5,response="8064fac79e90815539b2ee1aa33df011"
Contact: 101 <a class="moz-txt-link-rfc2396E"
href="sip:101@192.168.0.205:5060"><sip:101@192.168.0.205:5060></a>
User-Agent: Linksys/SPA2102-5.2.3
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Scheduling destruction of SIP dialog '<a
class="moz-txt-link-abbreviated"
href="mailto:3966843b4ccc213f5924f6c701536906@192.168.0.12">3966843b4ccc213f5924f6c701536906@192.168.0.12</a>'
in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.0.191:5060:
CANCEL <a class="moz-txt-link-freetext"
href="sip:108@192.168.0.191:5060">sip:108@192.168.0.191:5060</a>
SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK7548dfb8;rport
From: "New User" <a class="moz-txt-link-rfc2396E"
href="sip:6041112222@192.168.0.12"><sip:6041112222@192.168.0.12></a>;tag=as29c3bf9e
To: <a class="moz-txt-link-rfc2396E"
href="sip:108@192.168.0.191:5060"><sip:108@192.168.0.191:5060></a>
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:3966843b4ccc213f5924f6c701536906@192.168.0.12">3966843b4ccc213f5924f6c701536906@192.168.0.12</a>
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '<a
class="moz-txt-link-abbreviated"
href="mailto:3966843b4ccc213f5924f6c701536906@192.168.0.12">3966843b4ccc213f5924f6c701536906@192.168.0.12</a>'
in 6400 ms (Method: INVITE)
    -- SIP/101-08196f30 answered SIP/nv2.xxxl01.a-081c91e0
Audio is at 192.168.0.12 port 12632
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
tg2*CLI> 
<--- Reliably Transmitting (NAT) to 64.34.49.82:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
64.34.49.82:5060;branch=z9hG4bK23f3f87a;received=64.34.49.82;rport=5060
From: "6041112222" <a class="moz-txt-link-rfc2396E"
href="sip:6041112222@64.34.49.82"><sip:6041112222@64.34.49.82></a>;tag=as637ef124
To: <a class="moz-txt-link-rfc2396E"
href="sip:6041112222@96.49.128.97"><sip:6041112222@96.49.128.97></a>;tag=as36bb1ac1
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:07a241f9066a71ac31810e5a35127b7e@64.34.49.82">07a241f9066a71ac31810e5a35127b7e@64.34.49.82</a>
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <a class="moz-txt-link-rfc2396E"
href="sip:6041112222@192.168.0.12"><sip:6041112222@192.168.0.12></a>
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 1941 1941 IN IP4 192.168.0.12
s=session
c=IN IP4 192.168.0.12
t=0 0
m=audio 12632 RTP/AVP 3 0 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Packet2Packet bridging SIP/nv2.tevl01.a-081c91e0 and SIP/101-08196f30

</pre>
<u><b>Successful *8 Call Pickup (SIP
Trace)</b></u><br>
<pre>x<--- SIP read from 192.168.0.211:50386 --->
INVITE <a class="moz-txt-link-freetext"
href="sip:*8@192.168.0.12">sip:*8@192.168.0.12</a> SIP/2.0
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK29b51f96
From: "Cxxx Nxxxx" <a class="moz-txt-link-rfc2396E"
href="sip:106@192.168.0.12"><sip:106@192.168.0.12></a>;tag=000d6556eeb302bd081e4f74-20817e6e
To: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211">000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211</a>
Max-Forwards: 70
Date: Wed, 08 Apr 2009 00:54:50 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <a class="moz-txt-link-rfc2396E"
href="sip:106@192.168.0.211:5060;transport=udp"><sip:106@192.168.0.211:5060;transport=udp></a>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 275
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 25698 0 IN IP4 192.168.0.211
s=SIP Call
t=0 0
m=audio 19514 RTP/AVP 0 8 18 101
c=IN IP4 192.168.0.211
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (17 headers 13 lines) ---
Sending to 192.168.0.211 : 5060 (no NAT)
Using INVITE request as basis request - <a
class="moz-txt-link-abbreviated"
href="mailto:000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211">000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211</a>
Found no matching peer or user for '192.168.0.211:50386'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.211:19514
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Got unsupported a:fmtp in SDP offer 
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer 
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.211:19514
Looking for *8 in numberplan-custom-1 (domain 192.168.0.12)
list_route: hop: <a class="moz-txt-link-rfc2396E"
href="sip:106@192.168.0.211:5060;transport=udp"><sip:106@192.168.0.211:5060;transport=udp></a>
tg2*CLI> 
<--- Transmitting (no NAT) to 192.168.0.211:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.211:5060;branch=z9hG4bK29b51f96;received=192.168.0.211
From: "Cxxx Nxxxx" <a class="moz-txt-link-rfc2396E"
href="sip:106@192.168.0.12"><sip:106@192.168.0.12></a>;tag=000d6556eeb302bd081e4f74-20817e6e
To: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211">000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211</a>
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>
Content-Length: 0


<------------>
[Apr  7 17:58:50] NOTICE[2097]: chan_sip.c:13977 handle_request_invite: Nothing
to pick up for <a class="moz-txt-link-abbreviated"
href="mailto:000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211">000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211</a>

<--- Reliably Transmitting (no NAT) to 192.168.0.211:5060 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP
192.168.0.211:5060;branch=z9hG4bK29b51f96;received=192.168.0.211
From: "Cxxx Nxxxx" <a class="moz-txt-link-rfc2396E"
href="sip:106@192.168.0.12"><sip:106@192.168.0.12></a>;tag=000d6556eeb302bd081e4f74-20817e6e
To: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>;tag=as52f770b6
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211">000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211</a>
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>
Content-Length: 0


<------------>
tg2*CLI> 
<--- SIP read from 192.168.0.211:50387 --->
ACK <a class="moz-txt-link-freetext"
href="sip:*8@192.168.0.12">sip:*8@192.168.0.12</a> SIP/2.0
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK29b51f96
From: "Cxxx Nxxxx" <a class="moz-txt-link-rfc2396E"
href="sip:106@192.168.0.12"><sip:106@192.168.0.12></a>;tag=000d6556eeb302bd081e4f74-20817e6e
To: <a class="moz-txt-link-rfc2396E"
href="sip:*8@192.168.0.12"><sip:*8@192.168.0.12></a>;tag=as52f770b6
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211">000d6556-eeb3001c-76b88543-7f51d8f8@192.168.0.211</a>
Date: Wed, 08 Apr 2009 00:54:50 GMT
CSeq: 101 ACK
Content-Length: 0


<------------->

</pre>
<br>
<br>
<pre class="moz-signature" cols="72">-- 
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
  <a class="moz-txt-link-abbreviated"
href="http://www.netvoice.ca">www.netvoice.ca</a>  <a
 class="moz-txt-link-abbreviated"
href="http://www.ip-centrex.ca">www.ip-centrex.ca</a>  <a
 class="moz-txt-link-abbreviated"
href="http://www.ip-pbx.ca">www.ip-pbx.ca</a>  <a
 class="moz-txt-link-abbreviated"
href="http://www.vpas.ca">www.vpas.ca</a>
    <a class="moz-txt-link-abbreviated"
href="http://www.digium.ca">www.digium.ca</a> <a
 class="moz-txt-link-abbreviated"
href="http://www.grandstream.ca">www.grandstream.ca</a> <a
 class="moz-txt-link-abbreviated"
href="http://www.sipura.ca">www.sipura.ca</a> <a
 class="moz-txt-link-abbreviated"
href="http://www.snom.ca">www.snom.ca</a>
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
</pre>
</body>
</html>

Vincent Li

2009-Apr-08 18:54 UTC

head link

[asterisk-users] Call Pickup Works w/Linksys ATA, not with Cisco 7940G

On Tue, 7 Apr 2009, George Pajari wrote:
> I have an Asterisk 1.4.18 with a mix of cordless phones connected using 
Linksys SPA2102 ATAs and> Cisco 7940G phones. Unit obtains SIP trunking from an ITSP (server has 
no PCI boards).>
> *8 Call Pickup works fine from any of the phones connected using the 
Linksys SPA2102.>
> *8 Call Pickup does not work from the Cisco 7940G phones 
(chan_sip.c:13977> handle_request_invite: Nothing to pick up for 
000d6556-eeb3001c-76b88543-7f51d8f8 at 192.168.0.211)>
Seems someone else had the same problem back in 2004 and got no answer.

http://lists.digium.com/pipermail/asterisk-users/2004-April/036869.html


Vincent Li
System Administrator
BRC,UBC
perl
-e'print"\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012"'

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