Hairpinning issue? perhaps?

flan · June 17, 2013, 9:47pm

I have two sites here site-a (asterisk 1.4) site-b (cisco call manager)
site-a has a peer relationship with site-b as does site-b with site-a codecs are configured alike calls between the two sites work well. Station to station transfers work well.

The problem seems to come from when the asterisk server is in the call path and we try to transfer a call through the path in which the call came in on so-to-speak.

Here is the breakdown:

Call comes in from SIP provider @site-A
hits ivr that Dial()s an extension on a sip peer (site-b)
operator at site-b answers the phone needs to transfer back to site-a
operator at site-b blind and / or straight transfers call to extension @site-a.
a. station @site-a rings the user picks up and gets dead air.
b. outside call hears MOH even after the site-a user picks up.

Has anyone encountered a case whereby you have trouble transferring to another sip server through another sip server?

Please let me know if you need any further information. I have looked at a lot of different canreinvite configurations, as well as firewall rules, and the like… Also seen a bunch of stuff that didn’t pertain to the technology (SIP) that I am currently using.

Thanks,
Flan

dejanst · June 18, 2013, 11:20am

Can you please copy/paste the relevant parts of your sip.conf and extensions.conf and the CLI output of “sip set debug on” command?

flan · June 18, 2013, 1:46pm

sip.conf
[penn]
type=friend
host=x.x.x.x
nat=no
canreinvite=update,nonat
dtmfmode=rfc2833
progressinband=never
insecure=port
disallow=all
allow=ulaw
context=internal

extensions.conf
exten => 9146107800,1,Goto(ivr-support,7800,1)

[ivr-support]
exten => 7800,1,Answer()
exten => 7800,n,Set(CALLERID(name)=“SUP:”${CALLERID(name)})
exten => 7800,n,Playback(standby-for-next-engineer-fm)
exten => 7800,n,Dial(SIP/7832&SIP/530@penn,16,t) ; Try EOC and Lynn
exten => 7800,n,Goto(try_queues,7802,1) ; No answer? Try tech queues.

[try_queues]
Note: these are just queues nothing doing here.

; the tech support queues…
exten => 7802,1,Set(CALLERID(name)=“SUP:”${CALLERID(name)})
exten => 7802,n,Queue(techsupport_ringall||||16)
exten => 7802,n,Queue(techsupport_ringmore||||16)
exten => 7802,n,Queue(nightringer||||32767)

I tried to sanitize it with 555-555-5555 and sipprovider and x.x.x.x and penn
555-555-5555 == cell phone calling in from sipprovider
sipprovider == our upstream sip provider
x.x.x.x == mypbx
penn == remote pbx (cisco call manager sip gateway)

/*** the following is because the set isn’t logged in. however 7787 is
2013-06-17 09:43:33] WARNING[1809]: app_dial.c:1183 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
[2013-06-17 09:43:33] WARNING[1809]: chan_sip.c:2921 create_addr: No such host: 7787b
***/

set sip debug ip penn
numbersCLI>
numbersCLI>
numbers*CLI>
– Executing [7800@ivr-support:4] Dial(“SIP/sipprovider-b4ef0140”, “SIP/7832&SIP/530@penn|16|t”) in new stack
– Called 7832
Extension Changed 7832[xand_blf_hints] new state Ringing for Notify User 7801
Audio is at x.x.x.x port 6772
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to penn:5060:
INVITE sip:530@penn SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK6238e204;rport
From: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
To: sip:530@penn
Contact: sip:555-555-5555@x.x.x.x
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Jun 2013 13:43:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 7921 7921 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 6772 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called 530@penn

numbers*CLI>
<— SIP read from penn:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK6238e204;rport
From: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
To: sip:530@penn
Date: Mon, 17 Jun 2013 13:43:23 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from penn:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK6238e204;rport
From: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
To: sip:530@penn;tag=83993EFC-2142
Date: Mon, 17 Jun 2013 13:43:23 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: “Xand Noc” sip:530@penn;party=called;screen=no;privacy=off
Contact: sip:530@penn:5060
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

<------------->
— (13 headers 0 lines) —
– SIP/penn-b57237d8 is ringing
– SIP/7832-09224b70 is ringing
numbers*CLI>
<— SIP read from penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK6238e204;rport
From: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
To: sip:530@penn;tag=83993EFC-2142
Date: Mon, 17 Jun 2013 13:43:23 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: “Xand Noc” sip:530@penn;party=called;screen=no;privacy=off
Contact: sip:530@penn:5060
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uas
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 1827 2011 IN IP4 penn
s=SIP Call
c=IN IP4 penn
t=0 0
m=audio 16688 RTP/AVP 0 101
c=IN IP4 penn
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
— (19 headers 11 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port penn:16688
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port penn:16688
list_route: hop: sip:530@penn:5060
set_destination: Parsing sip:530@penn:5060 for address/port to send to
set_destination: set destination to penn, port 5060
Transmitting (no NAT) to penn:5060:
ACK sip:530@penn:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK06731406;rport
From: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
To: sip:530@penn;tag=83993EFC-2142
Contact: sip:555-555-5555@x.x.x.x
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

-- SIP/penn-b57237d8 answered SIP/sipprovider-b4ef0140

Extension Changed 7832[xand_blf_hints] new state Idle for Notify User 7801
– Registered SIP ‘7824’ at 216.150.131.207 port 32983 expires 60
[2013-06-17 09:43:30] NOTICE[7930]: chan_sip.c:12669 handle_response_peerpoke: Peer ‘7824’ is now Reachable. (16ms / 2000ms)
numbers*CLI>
<— SIP read from penn:50182 —>
INVITE sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B42155D
Remote-Party-ID: “Xand Noc” sip:530@penn;party=calling;screen=no;privacy=off
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:32 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3068555450-3599372770-2242030795-1396059506
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1371476612
Contact: sip:530@penn:5060
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 249

v=0
o=CiscoSystemsSIP-GW-UserAgent 1827 2012 IN IP4 penn
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 16688 RTP/AVP 0 101
c=IN IP4 0.0.0.0
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=inactive

<------------->
— (20 headers 12 lines) —
Sending to penn : 5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 0.0.0.0:16688
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 0.0.0.0:16688
numbers*CLI>
<— Transmitting (no NAT) to penn:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B42155D;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Length: 0

<------------>
Audio is at x.x.x.x port 6772
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
numbers*CLI>
<— Reliably Transmitting (no NAT) to penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B42155D;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 7921 7922 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 6772 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=inactive

<------------>
– Started music on hold, class ‘default’, on SIP/sipprovider-b4ef0140
numbers*CLI>
<— SIP read from penn:50182 —>
ACK sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B432069
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:32 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

<------------->
— (10 headers 0 lines) —
numbers*CLI>
<— SIP read from penn:50182 —>
INVITE sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B44C87
Remote-Party-ID: “Xand Noc” sip:530@penn;party=calling;screen=no;privacy=off
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:32 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3068555450-3599372770-2242030795-1396059506
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1371476612
Contact: sip:530@penn:5060
Expires: 180
Allow-Events: telephone-event
Content-Length: 0

<------------->
— (19 headers 0 lines) —
Sending to penn : 5060 (no NAT)
numbers*CLI>
<— Transmitting (no NAT) to penn:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B44C87;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Length: 0

<------------>
Audio is at x.x.x.x port 6772
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
numbers*CLI>
<— Reliably Transmitting (no NAT) to penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B44C87;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 7921 7923 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 6772 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
numbers*CLI>
<— SIP read from penn:50182 —>
ACK sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B451097
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:32 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 259

v=0
o=CiscoSystemsSIP-GW-UserAgent 1827 2013 IN IP4 penn
s=SIP Call
c=IN IP4 penn
t=0 0
m=audio 16688 RTP/AVP 0 101
c=IN IP4 penn
a=sendonly
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
— (11 headers 12 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port penn:16688
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port penn:16688
– Stopped music on hold on SIP/sipprovider-b4ef0140
– Started music on hold, class ‘default’, on SIP/sipprovider-b4ef0140
numbers*CLI>
<— SIP read from penn:50182 —>
INVITE sip:7787@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4625DE
Remote-Party-ID: “Xand Noc” sip:530@penn;party=calling;screen=yes;privacy=off
From: “Xand Noc” sip:530@penn;tag=83996638-1FFC
To: sip:7787@x.x.x.x
Date: Mon, 17 Jun 2013 13:43:33 GMT
Call-ID: BCE7EBAC-D68A11E2-85AEACCB-53362D72@penn
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3852760576-0000065536-0000001447-0242095276
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1371476613
Contact: sip:530@penn:5060
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0

<------------->
— (20 headers 0 lines) —
Sending to penn : 5060 (no NAT)
Using INVITE request as basis request - BCE7EBAC-D68A11E2-85AEACCB-53362D72@penn
Found peer 'penn’
Looking for 7787 in internal (domain x.x.x.x)
list_route: hop: sip:530@penn:5060
numbers*CLI>
<— Transmitting (no NAT) to penn:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4625DE;received=penn
From: “Xand Noc” sip:530@penn;tag=83996638-1FFC
To: sip:7787@x.x.x.x
Call-ID: BCE7EBAC-D68A11E2-85AEACCB-53362D72@penn
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7787@x.x.x.x
Content-Length: 0

<------------>
– Executing [7787@internal:1] Dial(“SIP/penn-b5c14d08”, “SIP/7787&SIP/7787a&SIP/7787b&SIP/7787c&SIP/7787d|30|t”) in new stack
– Called 7787
[2013-06-17 09:43:33] WARNING[1809]: app_dial.c:1183 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
[2013-06-17 09:43:33] WARNING[1809]: chan_sip.c:2921 create_addr: No such host: 7787b
[2013-06-17 09:43:33] WARNING[1809]: app_dial.c:1183 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
[2013-06-17 09:43:33] WARNING[1809]: chan_sip.c:2921 create_addr: No such host: 7787c
[2013-06-17 09:43:33] WARNING[1809]: app_dial.c:1183 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
[2013-06-17 09:43:33] WARNING[1809]: chan_sip.c:2921 create_addr: No such host: 7787d
[2013-06-17 09:43:33] WARNING[1809]: app_dial.c:1183 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
Extension Changed 7787[xand_blf_hints] new state Ringing for Notify User 7801
– SIP/7787-090274c8 is ringing
numbers*CLI>
<— Transmitting (no NAT) to penn:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4625DE;received=penn
From: “Xand Noc” sip:530@penn;tag=83996638-1FFC
To: sip:7787@x.x.x.x;tag=as46408165
Call-ID: BCE7EBAC-D68A11E2-85AEACCB-53362D72@penn
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7787@x.x.x.x
Content-Length: 0

<------------>
numbers*CLI>
<— SIP read from penn:50182 —>
INVITE sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B479AA
Remote-Party-ID: “Xand Noc” sip:530@penn;party=calling;screen=no;privacy=off
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:34 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3068555450-3599372770-2242030795-1396059506
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1371476614
Contact: sip:530@penn:5060
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 1827 2014 IN IP4 penn
s=SIP Call
c=IN IP4 penn
t=0 0
m=audio 16688 RTP/AVP 0 101
c=IN IP4 penn
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
— (20 headers 11 lines) —
Sending to penn : 5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port penn:16688
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port penn:16688
numbers*CLI>
<— Transmitting (no NAT) to penn:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B479AA;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Length: 0

<------------>
Audio is at x.x.x.x port 6772
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
numbers*CLI>
<— Reliably Transmitting (no NAT) to penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B479AA;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 7921 7924 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 6772 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Stopped music on hold on SIP/sipprovider-b4ef0140
numbers*CLI>
<— SIP read from penn:50182 —>
ACK sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B48935
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:34 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: telephone-event
Content-Length: 0

<------------->
— (10 headers 0 lines) —
numbers*CLI>
<— SIP read from penn:50182 —>
INVITE sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B49AE9
Remote-Party-ID: “Xand Noc” sip:530@penn;party=calling;screen=no;privacy=off
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:34 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3068555450-3599372770-2242030795-1396059506
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 104 INVITE
Max-Forwards: 70
Timestamp: 1371476614
Contact: sip:530@penn:5060
Expires: 180
Allow-Events: telephone-event
Content-Length: 0

<------------->
— (19 headers 0 lines) —
Sending to penn : 5060 (no NAT)
numbers*CLI>
<— Transmitting (no NAT) to penn:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B49AE9;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Length: 0

<------------>
Audio is at x.x.x.x port 6772
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
numbers*CLI>
<— Reliably Transmitting (no NAT) to penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B49AE9;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 7921 7925 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 6772 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
Extension Changed 7787[xand_blf_hints] new state InUse for Notify User 7801
– SIP/7787-090274c8 answered SIP/penn-b5c14d08
Audio is at x.x.x.x port 9662
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
numbers*CLI>
<— Reliably Transmitting (no NAT) to penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4625DE;received=penn
From: “Xand Noc” sip:530@penn;tag=83996638-1FFC
To: sip:7787@x.x.x.x;tag=as46408165
Call-ID: BCE7EBAC-D68A11E2-85AEACCB-53362D72@penn
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7787@x.x.x.x
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 7921 7921 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 9662 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
numbers*CLI>
<— SIP read from penn:50182 —>
ACK sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4A203
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:34 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Max-Forwards: 70
CSeq: 104 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 259

v=0
o=CiscoSystemsSIP-GW-UserAgent 1827 2015 IN IP4 penn
s=SIP Call
c=IN IP4 penn
t=0 0
m=audio 16688 RTP/AVP 0 101
c=IN IP4 penn
a=sendonly
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
— (11 headers 12 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port penn:16688
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port penn:16688
– Started music on hold, class ‘default’, on SIP/sipprovider-b4ef0140
numbers*CLI>
<— SIP read from penn:50182 —>
INVITE sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4B18DC
Remote-Party-ID: “Xand Noc” sip:530@penn;party=calling;screen=no;privacy=off
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:34 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3068555450-3599372770-2242030795-1396059506
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 105 INVITE
Max-Forwards: 70
Timestamp: 1371476614
Contact: sip:530@penn:5060
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 249

v=0
o=CiscoSystemsSIP-GW-UserAgent 1827 2016 IN IP4 penn
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 16688 RTP/AVP 0 101
c=IN IP4 0.0.0.0
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=inactive

<------------->
— (20 headers 12 lines) —
Sending to penn : 5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 0.0.0.0:16688
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 0.0.0.0:16688
numbers*CLI>
<— Transmitting (no NAT) to penn:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4B18DC;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 105 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Length: 0

<------------>
Audio is at x.x.x.x port 6772
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4B18DC;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 105 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 7921 7926 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 6772 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=inactive

<------------>
– Stopped music on hold on SIP/sipprovider-b4ef0140
– Started music on hold, class ‘default’, on SIP/sipprovider-b4ef0140

<— SIP read from penn:50182 —>
ACK sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4C2389
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:34 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Max-Forwards: 70
CSeq: 105 ACK
Allow-Events: telephone-event
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from penn:50182 —>
INVITE sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4D10D5
Remote-Party-ID: “Xand Noc” sip:530@penn;party=calling;screen=no;privacy=off
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:34 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3068555450-3599372770-2242030795-1396059506
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 106 INVITE
Max-Forwards: 70
Timestamp: 1371476614
Contact: sip:530@penn:5060
Expires: 180
Allow-Events: telephone-event
Content-Length: 0

<------------->
— (19 headers 0 lines) —
Sending to penn : 5060 (no NAT)
numbers*CLI>
<— Transmitting (no NAT) to penn:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4D10D5;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 106 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Length: 0

<------------>
Audio is at x.x.x.x port 6772
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
numbers*CLI>
<— Reliably Transmitting (no NAT) to penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4D10D5;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 106 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 7921 7927 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 6772 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
numbers*CLI>
<— SIP read from penn:50182 —>
ACK sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4EDB3
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:34 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
Max-Forwards: 70
CSeq: 106 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 1827 2017 IN IP4 penn
s=SIP Call
c=IN IP4 penn
t=0 0
m=audio 16688 RTP/AVP 0 101
c=IN IP4 penn
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
— (11 headers 11 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port penn:16688
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port penn:16688
numbers*CLI>
<— SIP read from penn:50182 —>
ACK sip:7787@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B4FEDA
From: “Xand Noc” sip:530@penn;tag=83996638-1FFC
To: sip:7787@x.x.x.x;tag=as46408165
Date: Mon, 17 Jun 2013 13:43:33 GMT
Call-ID: BCE7EBAC-D68A11E2-85AEACCB-53362D72@penn
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244

v=0
o=CiscoSystemsSIP-GW-UserAgent 0 7017 IN IP4 penn
s=SIP Call
c=IN IP4 penn
t=0 0
m=audio 26430 RTP/AVP 0 101
c=IN IP4 penn
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
— (12 headers 11 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port penn:26430
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port penn:26430
== Spawn extension (internal, 7787, 1) exited non-zero on 'SIP/penn-b5c14d08’
Extension Changed 7787[xand_blf_hints] new state Idle for Notify User 7801
Scheduling destruction of SIP dialog ‘BCE7EBAC-D68A11E2-85AEACCB-53362D72@penn’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:530@penn:5060 for address/port to send to
set_destination: set destination to penn, port 5060
Reliably Transmitting (no NAT) to penn:5060:
BYE sip:530@penn:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK2d8dfccc;rport
From: sip:7787@x.x.x.x;tag=as46408165
To: “Xand Noc” sip:530@penn;tag=83996638-1FFC
Call-ID: BCE7EBAC-D68A11E2-85AEACCB-53362D72@penn
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

numbers*CLI>
<— SIP read from penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK2d8dfccc;rport
From: sip:7787@x.x.x.x;tag=as46408165
To: “Xand Noc” sip:530@penn;tag=83996638-1FFC
Date: Mon, 17 Jun 2013 13:43:37 GMT
Call-ID: BCE7EBAC-D68A11E2-85AEACCB-53362D72@penn
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0

<------------->
— (10 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
numbers*CLI>
<— SIP read from penn:50182 —>
BYE sip:555-555-5555@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B51AE
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Date: Mon, 17 Jun 2013 13:43:34 GMT
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
User-Agent: Cisco-SIPGateway/IOS-12.x
Timestamp: 1371476617
CSeq: 107 BYE
Max-Forwards: 70
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to penn : 5060 (no NAT)
numbers*CLI>
<— Transmitting (no NAT) to penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B51AE;received=penn
From: sip:530@penn;tag=83993EFC-2142
To: “SUP:WIRELESS CALLER” sip:555-555-5555@x.x.x.x;tag=as56e5e4f2
Call-ID: 455c6c5864b3b2d4149345610b79e108@x.x.x.x
CSeq: 107 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555-555-5555@x.x.x.x
Content-Length: 0

<------------>
== Spawn extension (ivr-support, 7800, 4) exited non-zero on ‘SIP/sipprovider-b4ef0140’
– Stopped music on hold on SIP/sipprovider-b4ef0140
Really destroying SIP dialog ‘BCE7EBAC-D68A11E2-85AEACCB-53362D72@penn’ Method: ACK
Really destroying SIP dialog ‘455c6c5864b3b2d4149345610b79e108@x.x.x.x’ Method: BYE
numbers*CLI>
<— SIP read from penn:50182 —>
INVITE sip:7839@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B52740
Remote-Party-ID: “Betsy Mertz” sip:349@penn;party=calling;screen=no;privacy=off
From: sip:349@penn;tag=838BADC4-D2C
To: “Maria DiBlasi” sip:7839@x.x.x.x;tag=as17f90eaf
Date: Mon, 17 Jun 2013 13:43:39 GMT
Call-ID: 2248fa4f27cb5927729519895e1d1289@x.x.x.x
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2767065097-3599241698-2238819531-1396059506
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1371476619
Contact: sip:349@penn:5060
Expires: 180
Allow-Events: telephone-event
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 3839 1370 IN IP4 penn
s=SIP Call
c=IN IP4 penn
t=0 0
m=audio 22028 RTP/AVP 0 101
c=IN IP4 penn
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
— (21 headers 11 lines) —
Sending to penn : 5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port penn:22028
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port penn:22028
numbers*CLI>
<— Transmitting (no NAT) to penn:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B52740;received=penn
From: sip:349@penn;tag=838BADC4-D2C
To: “Maria DiBlasi” sip:7839@x.x.x.x;tag=as17f90eaf
Call-ID: 2248fa4f27cb5927729519895e1d1289@x.x.x.x
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7839@x.x.x.x
Content-Length: 0

<------------>
Audio is at x.x.x.x port 7522
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
numbers*CLI>
<— Reliably Transmitting (no NAT) to penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B52740;received=penn
From: sip:349@penn;tag=838BADC4-D2C
To: “Maria DiBlasi” sip:7839@x.x.x.x;tag=as17f90eaf
Call-ID: 2248fa4f27cb5927729519895e1d1289@x.x.x.x
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7839@x.x.x.x
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 7921 7922 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 7522 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
numbers*CLI>
<— SIP read from penn:50182 —>
ACK sip:7839@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP penn:5060;branch=z9hG4bK2B53574
From: sip:349@penn;tag=838BADC4-D2C
To: “Maria DiBlasi” sip:7839@x.x.x.x;tag=as17f90eaf
Date: Mon, 17 Jun 2013 13:43:39 GMT
Call-ID: 2248fa4f27cb5927729519895e1d1289@x.x.x.x
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Reliably Transmitting (no NAT) to penn:5060:
OPTIONS sip:penn SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK7a01a058;rport
From: “asterisk” sip:asterisk@x.x.x.x;tag=as2507a465
To: sip:penn
Contact: sip:asterisk@x.x.x.x
Call-ID: 72dde7df0b749b5f3112904f65b9e89a@x.x.x.x
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Jun 2013 13:43:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

numbers*CLI>
<— SIP read from penn:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK7a01a058;rport
From: “asterisk” sip:asterisk@x.x.x.x;tag=as2507a465
To: sip:penn;tag=83998468-515
Date: Mon, 17 Jun 2013 13:43:41 GMT
Call-ID: 72dde7df0b749b5f3112904f65b9e89a@x.x.x.x
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces,sdp-anat
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 450

v=0
o=CiscoSystemsSIP-GW-UserAgent 7035 9755 IN IP4 penn
s=SIP Call
c=IN IP4 penn
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 penn
m=image 0 udptl t38
c=IN IP4 penn
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:180
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
— (14 headers 18 lines) —
Really destroying SIP dialog ‘72dde7df0b749b5f3112904f65b9e89a@x.x.x.x’ Method: OPTIONS
numbers*CLI> exit
Executing last minute cleanups

dejanst · June 20, 2013, 9:14am

Try setting type=peer for penn in sip.conf. Probably won’t do much difference, but type=peer is usually less problematic than type=friend.

I am not understanding why the Cisco Manager sends so many SIP Invites on the second call. Can you think of a reason?

This message also strikes me as little odd:

<--- SIP read from penn:50182 --->

Is Cisco sending SIP Invite’s on another port than 5060?

flan · July 11, 2013, 8:50pm

Nope I can’t I am at wits end with this. We are going to try to upgrade from 1.4 to 1.6 and see if that fixes it. However if that doesn’t do it? Not sure what to try next. I will try to switch it to peer and see if that helps.

Also, hairpinning should work in this case right?

dejanst · July 12, 2013, 1:58pm

If you want to do an upgrade - upgrade it to 1.8 or 1.11 (one of the Long Term Releases). But I think that the problem is on the Cisco side …

flan · August 2, 2013, 1:47pm

What would make you think that its something up with the cisco side?

flan · August 16, 2013, 4:19am

Just an update on this thread. We decided to move forward with the upgrade to 1.8.11 and there were some syntactical issues that I didn’t see anyone talk about. Be that as it may, the upgrade in and of itself didn’t fix the “possible” hairpinning issue that we were having. However, the situation has changed a bit.
Here is the new break down;
call in from cell phone through sip provider.
call hits PBX and gets dial()'d out to the same cisco sip server as stated earlier.
operator picks up the phone “hey how are ya” blah blah blah.
4 digit transfers the call back across the callpath to an extension hanging off of the original sip server.
The call completes
pick up the line and we get one way audio. I can hear the person calling in however the person on the cell phone cannot hear me.

Codec issue perhaps? I am interested in what the community thinks. Its late so I will have to do debugs in the morning.

Thanks and let me know what ya’ll think.

Le meas,
Flan