thr3ads.net - asterisk users - [Asterisk-Users] SIP to SIP calls have no audio until put on hold and taken back off [Feb 2005]

If this information is useful, please help other people find it:
Share via:
Dave Ludlow
2005-Feb-20 09:11 UTC
[Asterisk-Users] SIP to SIP calls have no audio until put on hold and taken back off

A previous poster mentioned the same thing, with no response:

http://lists.digium.com/pipermail/asterisk-users/2004-
December/080161.html

Fresh asterisk 1.0.5 install on FC3, started with "make samples",
nothing fancy.  It's so bland, I'm surprised the list isn't full of
people having the same trouble.

I have several Uniden UIP200 phones and a single Grandstream BudgetTone
100.  Any combination does the same thing.

Calls started from within asterisk (*.call files, transfers, directory)
work fine.  I've tried all combinations of codecs, with no change.

This is my first serious attempt with *, so don't be afraid to assume
I'm a moron.

Relevent config snippets and a "set verbose 100" and SIP DEBUG console
dump follow.

*** sip.conf ***
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes

[1010]
type=friend
host=dynamic
username=1010
secret=password
context=default
dtmfmode=rfc2833

<1011-1019 are all basically the same as 1010>

*** extensions.conf ***
[default]
exten => 1010,1,Dial(SIP/1010,20,tr)
exten => 1011,1,Dial(SIP/1011,20,tr)
<etc>

*** console dump of call, hold, unhold, hangup ***
*** Asterisk on 192.168.200.0, phones on 192.168.201.0,
*** connected by VPN, same thing happens when on one lan

Sip read:
INVITE sip:1010@192.168.200.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1ae8b35c5d8d1ae5
From: <sip:1019@192.168.200.100>;tag=9970b15421c8f59c
To: <sip:1010@192.168.200.100>
Contact: <sip:1019@192.168.201.111>
Supported: replaces
Call-ID: 3b9c9ea24231eb6f@192.168.201.111
CSeq: 22567 INVITE
User-Agent: Grandstream BT100 1.0.5.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 354

v=0
=1019 0 8000 IN IP4 192.168.201.111
s=SIP Call
c=IN IP4 192.168.201.111
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 9
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=rtpmap:9 G722/8000
a=ptime:20

13 headers, 17 lines
Using latest request as basis request
Sending to 192.168.201.111 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 15
Found RTP audio format 99
Found RTP audio format 9
Peer audio RTP is at port 192.168.201.111:5004
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Found description format iLBC
Found description format G722
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
- 0x0 (nothing)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1ae8b35c5d8d1ae5
From: <sip:1019@192.168.200.100>;tag=9970b15421c8f59c
To: <sip:1010@192.168.200.100>;tag=as45319780
Call-ID: 3b9c9ea24231eb6f@192.168.201.111
CSeq: 22567 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1010@192.168.200.100>
Proxy-Authenticate: Digest realm="asterisk",
nonce="499f7907"
Content-Length: 0


 to 192.168.201.111:5060
Scheduling destruction of call '3b9c9ea24231eb6f@192.168.201.111' in
15000 ms
Found user '1019'
asterisk*CLI>

Sip read:
ACK sip:1010@192.168.200.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1ae8b35c5d8d1ae5
From: <sip:1019@192.168.200.100>;tag=9970b15421c8f59c
To: <sip:1010@192.168.200.100>;tag=as45319780
Contact: <sip:1019@192.168.201.111>
Call-ID: 3b9c9ea24231eb6f@192.168.201.111
CSeq: 22567 ACK
User-Agent: Grandstream BT100 1.0.5.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


11 headers, 0 lines
asterisk*CLI>

Sip read:
INVITE sip:1010@192.168.200.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1d2ba72e99353828
From: <sip:1019@192.168.200.100>;tag=9970b15421c8f59c
To: <sip:1010@192.168.200.100>
Contact: <sip:1019@192.168.201.111>
Supported: replaces
Proxy-Authorization: DIGEST username="1019",
realm="asterisk",
algorithm=MD5, uri="sip:1010@192.168.200.100",
nonce="499f7907",
response="80ba81f6c2dc429b45c8bb6d57c9b7d6"
Call-ID: 3b9c9ea24231eb6f@192.168.201.111
CSeq: 22568 INVITE
User-Agent: Grandstream BT100 1.0.5.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 354

v=0
o=1019 1 8000 IN IP4 192.168.201.111
s=SIP Call
c=IN IP4 192.168.201.111
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 9
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=rtpmap:9 G722/8000
a=ptime:20

14 headers, 17 lines
Using latest request as basis request
Sending to 192.168.201.111 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 15
Found RTP audio format 99
Found RTP audio format 9
Peer audio RTP is at port 192.168.201.111:5004
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Found description format iLBC
Found description format G722
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
- 0x0 (nothing)
Found user '1019'
Looking for 1010 in default
list_route: hop: <sip:1019@192.168.201.111>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1d2ba72e99353828
From: <sip:1019@192.168.200.100>;tag=9970b15421c8f59c
To: <sip:1010@192.168.200.100>;tag=as4a6e9e69
Call-ID: 3b9c9ea24231eb6f@192.168.201.111
CSeq: 22568 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1010@192.168.200.100>
Content-Length: 0


 to 192.168.201.111:5060
    -- Executing Dial("SIP/1019-2da8", "SIP/1010|20|tr") in
new stack
We're at 192.168.200.100 port 18284
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:1010@192.168.201.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.100:5060;branch=z9hG4bK065f8ed2
From: "1019" <sip:1019@192.168.200.100>;tag=as18eaba8f
To: <sip:1010@192.168.201.110:5060>
Contact: <sip:1019@192.168.200.100>
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 20 Feb 2005 02:01:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 5394 5394 IN IP4 192.168.200.100
s=session
c=IN IP4 192.168.200.100
t=0 0
m=audio 18284 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 192.168.201.110:5060
    -- Called 1010
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1d2ba72e99353828
From: <sip:1019@192.168.200.100>;tag=9970b15421c8f59c
To: <sip:1010@192.168.200.100>;tag=as4a6e9e69
Call-ID: 3b9c9ea24231eb6f@192.168.201.111
CSeq: 22568 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1010@192.168.200.100>
Content-Length: 0


 to 192.168.201.111:5060
asterisk*CLI>

Sip read:
SIP/2.0 100 Trying
To: <sip:1010@192.168.201.110:5060>;tag=3e68de2444adcd12482755b9adb9b72a
Via: SIP/2.0/UDP 192.168.200.100:5060;branch=z9hG4bK065f8ed2
From: "1019" <sip:1019@192.168.200.100>;tag=as18eaba8f
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
CSeq: 102 INVITE
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK


9 headers, 0 lines
asterisk*CLI>

Sip read:
SIP/2.0 180 Ringing
To: <sip:1010@192.168.201.110:5060>;tag=3e68de2444adcd12482755b9adb9b72a
Contact: <sip:1010@192.168.201.110:5060>
Via: SIP/2.0/UDP 192.168.200.100:5060;branch=z9hG4bK065f8ed2
From: "1019" <sip:1019@192.168.200.100>;tag=as18eaba8f
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
CSeq: 102 INVITE
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK


10 headers, 0 lines
    -- SIP/1010-11a6 is ringing
asterisk*CLI>

Sip read:
SIP/2.0 200 OK
To: <sip:1010@192.168.201.110:5060>;tag=3e68de2444adcd12482755b9adb9b72a
Contact: <sip:1010@192.168.201.110:5060>
Via: SIP/2.0/UDP 192.168.200.100:5060;branch=z9hG4bK065f8ed2
From: "1019" <sip:1019@192.168.200.100>;tag=as18eaba8f
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 228
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK

v=0
=- 5394 408891 IN IP4 192.168.201.110
s=session
c=IN IP4 192.168.201.110
t=0 0
m=audio 25022 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=silenceSupp:off - - - -

11 headers, 10 lines
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.201.110:25022
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
list_route: hop: <sip:1010@192.168.201.110:5060>
set_destination: Parsing <sip:1010@192.168.201.110:5060> for
address/port to send to
set_destination: set destination to 192.168.201.110, port 5060
Transmitting:
ACK sip:1010@192.168.201.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.100:5060;branch=z9hG4bK5d24050f
From: "1019" <sip:1019@192.168.200.100>;tag=as18eaba8f
To: <sip:1010@192.168.201.110:5060>;tag=3e68de2444adcd12482755b9adb9b72a
Contact: <sip:1019@192.168.200.100>
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.201.110:5060
    -- SIP/1010-11a6 answered SIP/1019-2da8
We're at 192.168.200.100 port 15918
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1d2ba72e99353828
From: <sip:1019@192.168.200.100>;tag=9970b15421c8f59c
To: <sip:1010@192.168.200.100>;tag=as4a6e9e69
Call-ID: 3b9c9ea24231eb6f@192.168.201.111
CSeq: 22568 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1010@192.168.200.100>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 5394 5394 IN IP4 192.168.200.100
s=session
c=IN IP4 192.168.200.100
t=0 0
m=audio 15918 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 192.168.201.111:5060
    -- Attempting native bridge of SIP/1019-2da8 and SIP/1010-11a6
asterisk*CLI>

Sip read:
ACK sip:1010@192.168.200.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bKcc477b3272f20746
From: <sip:1019@192.168.200.100>;tag=9970b15421c8f59c
To: <sip:1010@192.168.200.100>;tag=as4a6e9e69
Contact: <sip:1019@192.168.201.111>
Proxy-Authorization: DIGEST username="1019",
realm="asterisk",
algorithm=MD5, uri="sip:1010@192.168.200.100",
nonce="499f7907",
response="d33feeaabc1504babfa1361c11aa9157"
Call-ID: 3b9c9ea24231eb6f@192.168.201.111
CSeq: 22568 ACK
User-Agent: Grandstream BT100 1.0.5.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


12 headers, 0 lines
asterisk*CLI>

Sip read:
INVITE sip:1019@192.168.200.100 SIP/2.0
Via: SIP/2.0/UDP
192.168.201.110:5060;branch=z9hG4bKu040bf54feb3b4e70311c2ee7afc7e6b4
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
CSeq: 161018 INVITE
From: Jennifer_Smith
<sip:1010@192.168.201.110:5060>;tag=3e68de2444adcd12482755b9adb9b72a
To: <sip:1019@192.168.200.100>;tag=as18eaba8f
Contact: <sip:1010@192.168.201.110:5060>
Content-Type: application/sdp
User-Agent: Uniden SIP Phone p2 Ver BS4.63
Max-Forwards: 70
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Content-Length: 220

v=0
o=- 5394 408892 IN IP4 192.168.201.110
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 25022 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=silenceSupp:off - - - -

13 headers, 10 lines
Using latest request as basis request
Sending to 192.168.201.110 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 0.0.0.0:25022
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
    -- Started music on hold, class 'default', on SIP/1019-2da8
We're at 192.168.200.100 port 18284
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.201.110:5060;branch=z9hG4bKu040bf54feb3b4e70311c2ee7afc7e6b4
From: Jennifer_Smith
<sip:1010@192.168.201.110:5060>;tag=3e68de2444adcd12482755b9adb9b72a
To: <sip:1019@192.168.200.100>;tag=as18eaba8f
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
CSeq: 161018 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1019@192.168.200.100>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 5394 5395 IN IP4 192.168.200.100
s=session
c=IN IP4 192.168.200.100
t=0 0
m=audio 18284 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 192.168.201.110:5060
asterisk*CLI>

Sip read:
ACK sip:1019@192.168.200.100 SIP/2.0
Via: SIP/2.0/UDP
192.168.201.110:5060;branch=z9hG4bKj37a435302983ead28327abb7373f8195
CSeq: 161018 ACK
To: <sip:1019@192.168.200.100>;tag=as18eaba8f
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
From: Jennifer_Smith
<sip:1010@192.168.201.110:5060>;tag=3e68de2444adcd12482755b9adb9b72a
User-Agent: Uniden SIP Phone p2 Ver BS4.63


7 headers, 0 lines
asterisk*CLI>

Sip read:
INVITE sip:1019@192.168.200.100 SIP/2.0
Via: SIP/2.0/UDP
192.168.201.110:5060;branch=z9hG4bKr73acabf453c1824b49c9246400733406
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
CSeq: 161019 INVITE
From: Jennifer_Smith
<sip:1010@192.168.201.110:5060>;tag=3e68de2444adcd12482755b9adb9b72a
To: <sip:1019@192.168.200.100>;tag=as18eaba8f
Contact: <sip:1010@192.168.201.110:5060>
Content-Type: application/sdp
User-Agent: Uniden SIP Phone p2 Ver BS4.63
Max-Forwards: 70
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Content-Length: 274

v=0
o=- 716620892 408893 IN IP4 192.168.201.110
s=-
c=IN IP4 192.168.201.110
t=0 0
m=audio 25022 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=sendrecv
a=ptime:20

13 headers, 13 lines
Using latest request as basis request
Sending to 192.168.201.110 : 5060 (non-NAT)
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 192.168.201.110:25022
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
    -- Stopped music on hold on SIP/1019-2da8
We're at 192.168.200.100 port 18284
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.201.110:5060;branch=z9hG4bKr73acabf453c1824b49c9246400733406
From: Jennifer_Smith
<sip:1010@192.168.201.110:5060>;tag=3e68de2444adcd12482755b9adb9b72a
To: <sip:1019@192.168.200.100>;tag=as18eaba8f
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
CSeq: 161019 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1019@192.168.200.100>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 5394 5396 IN IP4 192.168.200.100
s=session
c=IN IP4 192.168.200.100
t=0 0
m=audio 18284 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 192.168.201.110:5060
asterisk*CLI>

Sip read:
ACK sip:1019@192.168.200.100 SIP/2.0
Via: SIP/2.0/UDP
192.168.201.110:5060;branch=z9hG4bKe0e8451c3a03b5aa46d28ed2f8d7f5ecb
CSeq: 161019 ACK
To: <sip:1019@192.168.200.100>;tag=as18eaba8f
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
From: Jennifer_Smith
<sip:1010@192.168.201.110:5060>;tag=3e68de2444adcd12482755b9adb9b72a
User-Agent: Uniden SIP Phone p2 Ver BS4.63


7 headers, 0 lines


-- 
Dave Ludlow <vendor@adsllc.com>
Advaned Digital Services LLC
Seemingly Similar Threads

Search for more reasonably related threads
asterisk users - Feb 2005 - SIP to SIP calls have no audio until put on hold and taken back off

[Asterisk-Users] SIP to SIP calls have no audio until put on hold and taken back off

Seemingly Similar Threads

Wisdom of the Ancients
