[Home]

Summary:ASTERISK-00032: Calls into Asterisk to SIP phone get dropped when put on hold
Reporter:keithtaylor (keithtaylor)Labels:
Date Opened:2003-08-01 04:43:43Date Closed:2004-09-25 02:23:01
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Using CVS from CVS-07/30/03-11:43:15 (cvs co -D "a fortnight ago" on 30/07/03)

I can make calls through asterisk between SIP phones (7940), put a call on hold, music plays and then resume the call.  But when making a call from the outside through the ZAP (E100P) card to a SIP phone and put a call on hold, it disconnects (sip debug is in additional info).

You cannot transfer a call in the same situation - but I guess it is a symptom of the same thing.


****** ADDITIONAL INFORMATION ******


pbx*CLI> sip debug
SIP Debugging Enabled
   -- Executing Macro("Zap/1-1", "common1|1571|sip/1@192.168.1.148") in new stack
   -- Executing Wait("Zap/1-1", "2") in new stack
   -- Accepting call from '1512982860' to '1571' on channel 1, span 1
   -- Executing Answer("Zap/1-1", "") in new stack
   -- Executing Playback("Zap/1-1", "transfer|skip") in new stack
   -- Playing 'transfer'
   -- Executing AGI("Zap/1-1", "CallerId.agi") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/CallerId.agi
   -- AGI Script CallerId.agi completed, returning 0
   -- Executing Dial("Zap/1-1", "sip/1@192.168.1.148|20") in new stack
We're at 192.168.1.212 port 11726
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
10 headers, 11 lines
Reliably Transmitting:
INVITE sip:1@192.168.1.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK5a3c80a9
From: "1512982860" <sip:1512982860@192.168.1.212>;tag=as7be7ba3d
To: <sip:1@192.168.1.148>
Contact: <sip:1512982860@192.168.1.212>
Call-ID: 6e82671553ffbbb715ba5a1d0040422e@192.168.1.212
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 1205 1205 IN IP4 192.168.1.212
s=session
c=IN IP4 192.168.1.212
t=0 0
m=audio 11726 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
(no NAT) to 192.168.1.148:5060
   -- Called 1@192.168.1.148
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK5a3c80a9
From: "1512982860" <sip:1512982860@192.168.1.212>;tag=as7be7ba3d
To: <sip:1@192.168.1.148>
Call-ID: 6e82671553ffbbb715ba5a1d0040422e@192.168.1.212
CSeq: 102 INVITE
Server: CSCO/5
Contact: <sip:1@192.168.1.148:5060>
Content-Length: 0


9 headers, 0 lines
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK5a3c80a9
From: "1512982860" <sip:1512982860@192.168.1.212>;tag=as7be7ba3d
To: <sip:1@192.168.1.148>;tag=000a8aa2315800b758182509-34b67116
Call-ID: 6e82671553ffbbb715ba5a1d0040422e@192.168.1.212
CSeq: 102 INVITE
Server: CSCO/5
Contact: <sip:1@192.168.1.148:5060>
Content-Length: 0


9 headers, 0 lines
   -- SIP/192.168.1.148-7750 is ringing
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK5a3c80a9
From: "1512982860" <sip:1512982860@192.168.1.212>;tag=as7be7ba3d
To: <sip:1@192.168.1.148>;tag=000a8aa2315800b758182509-34b67116
Call-ID: 6e82671553ffbbb715ba5a1d0040422e@192.168.1.212
CSeq: 102 INVITE
Server: CSCO/5
Contact: <sip:1@192.168.1.148:5060>
Content-Type: application/sdp
Content-Length: 197

v=0
o=Cisco-SIPUA 2441 8344 IN IP4 192.168.1.148
s=SIP Call
c=IN IP4 192.168.1.148
t=0 0
m=audio 17166 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

10 headers, 9 lines
Found audio format 0
Found audio format 101
Found description format PCMU
Found description format telephone-event
Capabilities: us - 524302, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
list_route: hop: <sip:1@192.168.1.148:5060>
set_destination: Parsing <sip:1@192.168.1.148:5060> for address/port to send to
set_destination: set destination to 192.168.1.148, port 5060
Transmitting:
ACK sip:1@192.168.1.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK5a3c80a9
From: "1512982860" <sip:1512982860@192.168.1.212>;tag=as7be7ba3d
To: <sip:1@192.168.1.148>;tag=000a8aa2315800b758182509-34b67116
Contact: <sip:1512982860@192.168.1.212>
Call-ID: 6e82671553ffbbb715ba5a1d0040422e@192.168.1.212
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 192.168.1.148:5060
   -- SIP/192.168.1.148-7750 answered Zap/1-1
Sip read:
INVITE sip:1512982860@192.168.1.212:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.148:5060
From: <sip:1@192.168.1.148>;tag=000a8aa2315800b758182509-34b67116
To: "1512982860" <sip:1512982860@192.168.1.212>;tag=as7be7ba3d
Call-ID: 6e82671553ffbbb715ba5a1d0040422e@192.168.1.212
CSeq: 101 INVITE
User-Agent: CSCO/5
Contact: <sip:1@192.168.1.148:5060>
Content-Type: application/sdp
Content-Length: 192

v=0
o=Cisco-SIPUA 1107 26059 IN IP4 192.168.1.148
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 17166 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

10 headers, 9 lines
Using latest request as basis request
Sending to 192.168.1.148 : 5060 (non-NAT)
Found audio format 0
Found audio format 101
Found description format PCMU
Found description format telephone-event
Capabilities: us - 524302, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
   -- Started music on hold, class 'default', on Zap/1-1
We're at 192.168.1.212 port 11726
Answering with capability 4
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.148:5060
From: <sip:1@192.168.1.148>;tag=000a8aa2315800b758182509-34b67116
To: "1512982860" <sip:1512982860@192.168.1.212>;tag=as7be7ba3d
Call-ID: 6e82671553ffbbb715ba5a1d0040422e@192.168.1.212
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Contact: <sip:1512982860@192.168.1.212>
Content-Type: application/sdp
Content-Length: 189

v=0
o=root 1205 1205 IN IP4 192.168.1.212
s=session
c=IN IP4 192.168.1.212
t=0 0
m=audio 11726 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

to 192.168.1.148:5060
set_destination: Parsing <sip:1@192.168.1.148:5060> for address/port to send to
set_destination: set destination to 192.168.1.148, port 5060
Reliably Transmitting:
BYE sip:1@192.168.1.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK5a3c80a9
From: "1512982860" <sip:1512982860@192.168.1.212>;tag=as7be7ba3d
To: <sip:1@192.168.1.148>;tag=000a8aa2315800b758182509-34b67116
Contact: <sip:1512982860@192.168.1.212>
Call-ID: 6e82671553ffbbb715ba5a1d0040422e@192.168.1.212
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 192.168.1.148:5060
 == Spawn extension (macro-common1, s, 5) exited non-zero on 'Zap/1-1' in macro 'common1'
 == Spawn extension (default, 1571, 1) exited non-zero on 'Zap/1-1'
   -- Stopped music on hold on Zap/1-1
   -- Hungup 'Zap/1-1'
Sip read:
ACK sip:1512982860@192.168.1.212:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.148:5060
From: <sip:1@192.168.1.148>;tag=000a8aa2315800b758182509-34b67116
To: "1512982860" <sip:1512982860@192.168.1.212>;tag=as7be7ba3d
Call-ID: 6e82671553ffbbb715ba5a1d0040422e@192.168.1.212
CSeq: 101 ACK
User-Agent: CSCO/5
Content-Length: 0


8 headers, 0 lines
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK5a3c80a9
From: "1512982860" <sip:1512982860@192.168.1.212>;tag=as7be7ba3d
To: <sip:1@192.168.1.148>;tag=000a8aa2315800b758182509-34b67116
Call-ID: 6e82671553ffbbb715ba5a1d0040422e@192.168.1.212
CSeq: 103 BYE
Server: CSCO/5
Content-Length: 0


8 headers, 0 lines
Message is BYE
pbx*CLI> exit
[root@pbx cdr-csv]#
Comments:By: keithtaylor (keithtaylor) 2003-08-01 05:47:00

please resolve this - it is fixed in this morning's CVS
By: Mark Spencer (markster) 2003-08-02 15:04:50

Fixed in CVS