[Asterisk-Dev] Hold Issue
Alberto Fernandez
asterisk at xynergia.net
Thu Jul 1 12:12:01 MST 2004
When i call an extention, and i put a call on hold or i flash the call
it doesnt let me. Looking at the CLI I noticed that it sayd this.
-- Started music on hold, class 'default', on SIP/192.168.0.158-08ec53e8
-- Stopped music on hold on SIP/192.168.0.158-08ec53e8
I spoke to BKW_ and he asked me if the phones where behind nat. I took
everything and putted it all on the same network. I still had the same
outcome. I made sure that my extentions sayd nat=never, I have tested
with several hardware. If i call from one phone to an other, it works.
But if i call from the PSTN using a cisco 3800, it doesnt work.
It works if the call goes directicly to music on hold.
exten => 2603,1,MusicOnHold(default)
Here is the sip debug ONCE i press hold this happens.
Sip read:
INVITE sip:3052675760 at 192.168.0.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.18:60599;branch=z9hG4bK28577d0306261bb6
From: <sip:2601 at 192.168.0.18:60599>;tag=e837deb6b1f056dd
To: "3052675760" <sip:3052675760 at 192.168.0.157>;tag=as1ffa5810
Contact: <sip:2601 at 192.168.0.18:60599;user=phone>
Call-ID: 2b6451f52fe83fbb2017f86e408442ce at 192.168.0.157
CSeq: 25072 INVITE
User-Agent: Grandstream BT100 1.0.4.71
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 151
v=0
o=2601 8000 8000 IN IP4 192.168.0.18
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 19800 RTP/AVP 0
a=sendonly
a=rtpmap:0 PCMU/8000
a=ptime:20
��12 headers, 9 lines
��Using latest request as basis request
��Sending to 192.168.0.18 : 60599 (non-NAT)
��Found RTP audio format 0
��Peer RTP is at port 0.0.0.0:0
��Found description format PCMU
��Capabilities: us - 0x4(ULAW), peer -
audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
��Non-codec capabilities: us - 0x1(G723), peer -
0x0(EMPTY), combined - 0x0(EMPTY)
�� -- Started music on hold, class 'default', on
SIP/192.168.0.158-08eb0188
��We're at 192.168.0.157 port 11450
��Answering/Requesting with root capability 4
��Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.18:60599;branch=z9hG4bK28577d0306261bb6
From: <sip:2601 at 192.168.0.18:60599>;tag=e837deb6b1f056dd
To: "3052675760" <sip:3052675760 at 192.168.0.157>;tag=as1ffa5810
Call-ID: 2b6451f52fe83fbb2017f86e408442ce at 192.168.0.157
CSeq: 25072 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3052675760 at 192.168.0.157>
Content-Type: application/sdp
Content-Length: 160
v=0
o=root 2730 2733 IN IP4 192.168.0.158
s=session
c=IN IP4 192.168.0.158
t=0 0
m=audio 20588 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
to 192.168.0.18:60599
set_destination: Parsing <sip:3052675760 at 192.168.0.158:5060;user=phone>
for address/port to send to
set_destination: set destination to 192.168.0.158, port 5060
We're at 192.168.0.157 port 12234
Answering with preferred capability 0x4(ULAW)
11 headers, 8 lines
Reliably Transmitting:
INVITE sip:3052675760 at 192.168.0.158:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK26bd546d;rport
From:
<sip:2601 at 192.168.0.157;user=phone;phone-context=local>;tag=as251b0dd8
To: "3052675760" <sip:3052675760 at 192.168.0.158>
Contact: <sip:2601 at 192.168.0.157>
Call-ID: ED798706-19CD11CC-84B9A654-50489A8A at 192.168.0.158
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 160
v=0
o=root 2730 2732 IN IP4 192.168.0.157
s=session
c=IN IP4 192.168.0.157
t=0 0
m=audio 12234 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
(no NAT) to 192.168.0.158:5060
office*CLI>
Sip read:
ACK sip:3052675760 at 192.168.0.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.18:60599;branch=z9hG4bK28577d0306261bb6
From: <sip:2601 at 192.168.0.18:60599>;tag=e837deb6b1f056dd
To: "3052675760" <sip:3052675760 at 192.168.0.157>;tag=as1ffa5810
Contact: <sip:2601 at 192.168.0.18:60599;user=phone>
Call-ID: 2b6451f52fe83fbb2017f86e408442ce at 192.168.0.157
CSeq: 25072 ACK
User-Agent: Grandstream BT100 1.0.4.71
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
��11 headers, 0 lines
office*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK26bd546d;rport
From:
<sip:2601 at 192.168.0.157;user=phone;phone-context=local>;tag=as251b0dd8
To: "3052675760" <sip:3052675760 at 192.168.0.158>
Date: Sun, 07 Mar 1993 04:50:06 GMT
Call-ID: ED798706-19CD11CC-84B9A654-50489A8A at 192.168.0.158
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Contact: <sip:3052675760 at 192.168.0.158:5060;user=phone>
CSeq: 103 INVITE
Content-Type: application/sdp
Content-Length: 135
v=0
o=CiscoSystemsSIP-GW-UserAgent 6124 502 IN IP4 192.168.0.158
s=SIP Call
c=IN IP4 192.168.0.158
t=0 0
m=audio 20892 RTP/AVP 0
��11 headers, 6 lines
��Found RTP audio format 0
��Peer RTP is at port 192.168.0.158:0
��Capabilities: us - 0x4(ULAW), peer -
audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
��Non-codec capabilities: us - 0x1(G723), peer -
0x0(EMPTY), combined - 0x0(EMPTY)
��set_destination: Parsing
<sip:3052675760 at 192.168.0.158:5060;user=phone> for address/port to send
to
��set_destination: set destination to 192.168.0.158, port
5060
��Transmitting:
ACK sip:3052675760 at 192.168.0.158:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK33c10579;rport
From:
<sip:2601 at 192.168.0.157;user=phone;phone-context=local>;tag=as251b0dd8
To: "3052675760" <sip:3052675760 at 192.168.0.158>
Contact: <sip:2601 at 192.168.0.157>
Call-ID: ED798706-19CD11CC-84B9A654-50489A8A at 192.168.0.158
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.0.158:5060
set_destination: Parsing <sip:2601 at 192.168.0.18:60599;user=phone> for
address/port to send to
set_destination: set destination to 192.168.0.18, port 60599
We're at 192.168.0.157 port 11450
Answering/Requesting with root capability 4
11 headers, 8 lines
Reliably Transmitting:
INVITE sip:2601 at 192.168.0.18:60599 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK66fa4ef1;rport
From: "3052675760" <sip:3052675760 at 192.168.0.157>;tag=as1ffa5810
To: <sip:2601 at 192.168.0.18:60599>;tag=e837deb6b1f056dd
Contact: <sip:3052675760 at 192.168.0.157>
Call-ID: 2b6451f52fe83fbb2017f86e408442ce at 192.168.0.157
CSeq: 105 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 160
v=0
o=root 2730 2734 IN IP4 192.168.0.158
s=session
c=IN IP4 192.168.0.158
t=0 0
m=audio 20892 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
(no NAT) to 192.168.0.18:60599
office*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK66fa4ef1;rport
From: "3052675760" <sip:3052675760 at 192.168.0.157>;tag=as1ffa5810
To: <sip:2601 at 192.168.0.18:60599>;tag=e837deb6b1f056dd
Call-ID: 2b6451f52fe83fbb2017f86e408442ce at 192.168.0.157
CSeq: 105 INVITE
User-Agent: Grandstream BT100 1.0.4.71
Contact: <sip:2601 at 192.168.0.18:60599;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 144
v=0
o=2601 8000 8000 IN IP4 192.168.0.18
s=SIP Call
c=IN IP4 192.168.0.18
t=0 0
m=audio 19800 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
��11 headers, 8 lines
��Found RTP audio format 0
��Peer RTP is at port 192.168.0.18:0
��Found description format PCMU
��Capabilities: us - 0x4(ULAW), peer -
audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
��Non-codec capabilities: us - 0x1(G723), peer -
0x0(EMPTY), combined - 0x0(EMPTY)
�� -- Stopped music on hold on
SIP/192.168.0.158-08eb0188
��set_destination: Parsing
<sip:2601 at 192.168.0.18:60599;user=phone> for address/port to send to
��set_destination: set destination to 192.168.0.18, port
60599
��Transmitting:
ACK sip:2601 at 192.168.0.18:60599 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK0771d028;rport
From: "3052675760" <sip:3052675760 at 192.168.0.157>;tag=as1ffa5810
To: <sip:2601 at 192.168.0.18:60599>;tag=e837deb6b1f056dd
Contact: <sip:3052675760 at 192.168.0.157>
Call-ID: 2b6451f52fe83fbb2017f86e408442ce at 192.168.0.157
CSeq: 105 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.0.18:60599
set_destination: Parsing <sip:3052675760 at 192.168.0.158:5060;user=phone>
for address/port to send to
set_destination: set destination to 192.168.0.158, port 5060
We're at 192.168.0.157 port 12234
Answering with preferred capability 0x4(ULAW)
11 headers, 8 lines
Reliably Transmitting:
INVITE sip:3052675760 at 192.168.0.158:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK5ec63c82;rport
From:
<sip:2601 at 192.168.0.157;user=phone;phone-context=local>;tag=as251b0dd8
To: "3052675760" <sip:3052675760 at 192.168.0.158>
Contact: <sip:2601 at 192.168.0.157>
Call-ID: ED798706-19CD11CC-84B9A654-50489A8A at 192.168.0.158
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 158
v=0
o=root 2730 2733 IN IP4 192.168.0.18
s=session
c=IN IP4 192.168.0.18
t=0 0
m=audio 19800 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
(no NAT) to 192.168.0.158:5060
office*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK5ec63c82;rport
From:
<sip:2601 at 192.168.0.157;user=phone;phone-context=local>;tag=as251b0dd8
To: "3052675760" <sip:3052675760 at 192.168.0.158>
Date: Sun, 07 Mar 1993 04:50:06 GMT
Call-ID: ED798706-19CD11CC-84B9A654-50489A8A at 192.168.0.158
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Contact: <sip:3052675760 at 192.168.0.158:5060;user=phone>
CSeq: 104 INVITE
Content-Type: application/sdp
Content-Length: 135
v=0
o=CiscoSystemsSIP-GW-UserAgent 6124 503 IN IP4 192.168.0.158
s=SIP Call
c=IN IP4 192.168.0.158
t=0 0
m=audio 20546 RTP/AVP 0
��11 headers, 6 lines
��Found RTP audio format 0
��Peer RTP is at port 192.168.0.158:0
��Capabilities: us - 0x4(ULAW), peer -
audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
��Non-codec capabilities: us - 0x1(G723), peer -
0x0(EMPTY), combined - 0x0(EMPTY)
��set_destination: Parsing
<sip:3052675760 at 192.168.0.158:5060;user=phone> for address/port to send
to
��set_destination: set destination to 192.168.0.158, port
5060
��Transmitting:
ACK sip:3052675760 at 192.168.0.158:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK3c17f66d;rport
From:
<sip:2601 at 192.168.0.157;user=phone;phone-context=local>;tag=as251b0dd8
To: "3052675760" <sip:3052675760 at 192.168.0.158>
Contact: <sip:2601 at 192.168.0.157>
Call-ID: ED798706-19CD11CC-84B9A654-50489A8A at 192.168.0.158
CSeq: 104 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.0.158:5060
set_destination: Parsing <sip:2601 at 192.168.0.18:60599;user=phone> for
address/port to send to
��set_destination: set destination to 192.168.0.18, port
60599
��We're at 192.168.0.157 port 11450
��Answering/Requesting with root capability 4
��11 headers, 8 lines
��Reliably Transmitting:
INVITE sip:2601 at 192.168.0.18:60599 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK2b67b774;rport
From: "3052675760" <sip:3052675760 at 192.168.0.157>;tag=as1ffa5810
To: <sip:2601 at 192.168.0.18:60599>;tag=e837deb6b1f056dd
Contact: <sip:3052675760 at 192.168.0.157>
Call-ID: 2b6451f52fe83fbb2017f86e408442ce at 192.168.0.157
CSeq: 106 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 160
v=0
o=root 2730 2735 IN IP4 192.168.0.158
s=session
c=IN IP4 192.168.0.158
t=0 0
m=audio 20546 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
(no NAT) to 192.168.0.18:60599
office*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK2b67b774;rport
From: "3052675760" <sip:3052675760 at 192.168.0.157>;tag=as1ffa5810
To: <sip:2601 at 192.168.0.18:60599>;tag=e837deb6b1f056dd
Call-ID: 2b6451f52fe83fbb2017f86e408442ce at 192.168.0.157
CSeq: 106 INVITE
User-Agent: Grandstream BT100 1.0.4.71
Contact: <sip:2601 at 192.168.0.18:60599;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 144
v=0
o=2601 8000 8000 IN IP4 192.168.0.18
s=SIP Call
c=IN IP4 192.168.0.18
t=0 0
m=audio 19800 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
��11 headers, 8 lines
��Found RTP audio format 0
��Peer RTP is at port 192.168.0.18:0
��Found description format PCMU
��Capabilities: us - 0x4(ULAW), peer -
audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
��Non-codec capabilities: us - 0x1(G723), peer -
0x0(EMPTY), combined - 0x0(EMPTY)
��set_destination: Parsing
<sip:2601 at 192.168.0.18:60599;user=phone> for address/port to send to
��set_destination: set destination to 192.168.0.18, port
60599
��Transmitting:
ACK sip:2601 at 192.168.0.18:60599 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK58148e67;rport
From: "3052675760" <sip:3052675760 at 192.168.0.157>;tag=as1ffa5810
To: <sip:2601 at 192.168.0.18:60599>;tag=e837deb6b1f056dd
Contact: <sip:3052675760 at 192.168.0.157>
Call-ID: 2b6451f52fe83fbb2017f86e408442ce at 192.168.0.157
CSeq: 106 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.0.18:60599
Destroying call '4d31adcd0f493d32 at 192.168.0.18'
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