[Asterisk-bsd] Zap on hold problem
dvalencia
dvalencia at powervt.com.ar
Sun Mar 5 10:12:21 MST 2006
Hi everyone, I have a serious problem with calls incoming from a pstn, I can´t transfer it. Asterisk version is 1.2.4 running on Freebsd 6, with zaptel drivers v. 0.11
The call flow:
-- Starting simple switch on 'Zap/2-1'
Mar 5 13:46:41 NOTICE[3477]: chan_zap.c:6063 ss_thread: Got event 2 (Ring/Answered)...
-- Executing Wait("Zap/2-1", "1") in new stack
-- Executing Answer("Zap/2-1", "") in new stack
-- Executing Set("Zap/2-1", "TIMEOUT(digit)=5") in new stack
-- Digit timeout set to 5
-- Executing Set("Zap/2-1", "TIMEOUT(response)=3") in new stack
-- Response timeout set to 3
-- Executing Set("Zap/2-1", "LANGUAGE()=es") in new stack
-- Executing BackGround("Zap/2-1", "welcome") in new stack
-- Playing 'welcome' (language 'es')
== CDR updated on Zap/2-1
-- Executing Macro("Zap/2-1", "stdexten|233|SIP/233") in new stack (233 is eyebeam)
-- Executing Dial("Zap/2-1", "SIP/233|30|t") in new stack
-- Called 233
-- SIP/233-7aa8 is ringing
-- SIP/233-7aa8 answered Zap/2-1 -------------> I press "line 2" button on eyebeam to call to other extension
-- Started music on hold, class 'default', on Zap/2-1 ---------> MOH on ZAP
In this point the callee (PSTN) is on MOH, but I can't return to call 1 to transfer it. After a few minutes the eyebeam says "Failed to place call on hold"
I made test to transfer calls between sip uas and it works fine, the problem is with incoming calls from zap channel.
When I try it with a sipura instead eyebeam, I see the same behaviour, but the zap channel hung up when I try to return to call 1.
On the sip debug I don´t see an asterisk response to the moh invite from eyebeam, is it correct? I attach the sip log
Could anybody help me?
Thanks!
Diego
-------------- next part --------------
SIP read from 192.168.1.99:6900:
--- (0 headers 0 lines) Nat keepalive ---
Mar 5 13:55:25 NOTICE[3477]: chan_zap.c:6063 ss_thread: Got event 2 (Ring/Answered)...
-- Executing Wait("Zap/2-1", "1") in new stack
-- Executing Answer("Zap/2-1", "") in new stack
-- Executing Set("Zap/2-1", "TIMEOUT(digit)=5") in new stack
-- Digit timeout set to 5
-- Executing Set("Zap/2-1", "TIMEOUT(response)=3") in new stack
-- Response timeout set to 3
-- Executing Set("Zap/2-1", "LANGUAGE()=es") in new stack
-- Executing BackGround("Zap/2-1", "welcome") in new stack
-- Playing 'welcome' (language 'es')
== CDR updated on Zap/2-1
-- Executing Macro("Zap/2-1", "stdexten|233|SIP/233") in new stack
-- Executing Dial("Zap/2-1", "SIP/233|30|t") in new stack
We're at ip.of.asterisk port 10152
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (no NAT) to ip.of.UA:6738:
INVITE sip:233 at ip.of.UA:6738 SIP/2.0
Via: SIP/2.0/UDP ip.of.asterisk:5060;branch=z9hG4bK24acf382;rport
From: "asterisk" <sip:asterisk at ip.of.asterisk>;tag=as4e5acd84
To: <sip:233 at ip.of.UA:6738>
Contact: <sip:asterisk at ip.of.asterisk>
Call-ID: 5396c39e4f04e4f14a52b2be7b54fe10 at ip.of.asterisk
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 05 Mar 2006 16:55:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 3477 3477 IN IP4 ip.of.asterisk
s=session
c=IN IP4 ip.of.asterisk
t=0 0
m=audio 10152 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 233
<-- SIP read from ip.of.UA:6738:
SIP/2.0 180 Ringing
To: <sip:233 at ip.of.UA:6738>;tag=26682c38
From: "asterisk"<sip:asterisk at ip.of.asterisk>;tag=as4e5acd84
Via: SIP/2.0/UDP ip.of.asterisk:5060;branch=z9hG4bK24acf382;rport=5060;received=ip.of.asterisk
Call-ID: 5396c39e4f04e4f14a52b2be7b54fe10 at ip.of.asterisk
CSeq: 102 INVITE
Contact: <sip:233 at ip.of.UA:6738>
Content-Length: 0
--- (8 headers 0 lines)---
-- SIP/233-0bb2 is ringing
<-- SIP read from ip.of.UA:6738:
--- (0 headers 0 lines) Nat keepalive ---
<-- SIP read from ip.of.UA:6738:
INVITE sip:asterisk at ip.of.asterisk SIP/2.0
To: "asterisk"<sip:asterisk at ip.of.asterisk>;tag=as0a0f3604
From: <sip:233 at ip.of.UA:6738>;tag=fd79775f
Via: SIP/2.0/UDP ip.of.UA:6738;branch=z9hG4bK-d87543-12584724-1--d87543-;rport
Call-ID: 3465f442118733ee4362573f6d4e0ba8 at ip.of.asterisk
CSeq: 4 INVITE
Contact: <sip:233 at ip.of.UA:6738>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="233",realm="asterisk",nonce="5888fae2",uri="sip:asterisk at ip.of.asterisk",response="c111f002bc128b4da54ff69a662ebdfd",algorithm=MD5
User-Agent: eyeBeam release 3004t stamp 16741
Content-Length: 234
v=0
o=- 14115490 14635628 IN IP4 ip.of.UA
s=eyeBeam
c=IN IP4 0.0.0.0
t=0 0
m=audio 6740 RTP/AVP 18 0 8 101
a=alt:1 1 : 51B3EDC9 554BDE6A ip.of.UA 6740
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendonly
--- (13 headers 10 lines)---
Using INVITE request as basis request - 3465f442118733ee4362573f6d4e0ba8 at ip.of.asterisk
Sending to ip.of.UA : 6738 (NAT)
Reliably Transmitting (no NAT) to ip.of.UA:6738:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP ip.of.UA:6738;branch=z9hG4bK-d87543-12584724-1--d87543-;rport;received=ip.of.UA
From: <sip:233 at ip.of.UA:6738>;tag=fd79775f
To: "asterisk"<sip:asterisk at ip.of.asterisk>;tag=as0a0f3604
Call-ID: 3465f442118733ee4362573f6d4e0ba8 at ip.of.asterisk
CSeq: 4 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:asterisk at ip.of.asterisk>
Proxy-Authenticate: Digest realm="asterisk", nonce="534c6e42"
Content-Length: 0
---
Scheduling destruction of call '3465f442118733ee4362573f6d4e0ba8 at ip.of.asterisk' in 15000 ms
Found user '233'
<-- SIP read from ip.of.UA:6738:
SIP/2.0 200 OK
To: <sip:233 at ip.of.UA:6738>;tag=26682c38
From: "asterisk"<sip:asterisk at ip.of.asterisk>;tag=as4e5acd84
Via: SIP/2.0/UDP ip.of.asterisk:5060;branch=z9hG4bK24acf382;rport=5060;received=ip.of.asterisk
Call-ID: 5396c39e4f04e4f14a52b2be7b54fe10 at ip.of.asterisk
CSeq: 102 INVITE
Contact: <sip:233 at ip.of.UA:6738>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 241
v=0
o=- 14631840 14631866 IN IP4 ip.of.UA
s=eyeBeam
c=IN IP4 ip.of.UA
t=0 0
m=audio 6744 RTP/AVP 18 0 8 101
a=alt:1 1 : 9EECBE2D 6FD6ECEA ip.of.UA 6744
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (10 headers 10 lines)---
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port ip.of.UA:6744
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:233 at ip.of.UA:6738>
set_destination: Parsing <sip:233 at ip.of.UA:6738> for address/port to send to
set_destination: set destination to ip.of.UA, port 6738
Transmitting (no NAT) to ip.of.UA:6738:
ACK sip:233 at ip.of.UA:6738 SIP/2.0
Via: SIP/2.0/UDP ip.of.asterisk:5060;branch=z9hG4bK1a762a45;rport
From: "asterisk" <sip:asterisk at ip.of.asterisk>;tag=as4e5acd84
To: <sip:233 at ip.of.UA:6738>;tag=26682c38
Contact: <sip:asterisk at ip.of.asterisk>
Call-ID: 5396c39e4f04e4f14a52b2be7b54fe10 at ip.of.asterisk
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/233-0bb2 answered Zap/2-1
<-- SIP read from ip.of.UA:6738:
ACK sip:asterisk at ip.of.asterisk SIP/2.0
To: "asterisk"<sip:asterisk at ip.of.asterisk>;tag=as0a0f3604
From: <sip:233 at ip.of.UA:6738>;tag=fd79775f
Via: SIP/2.0/UDP ip.of.UA:6738;branch=z9hG4bK-d87543-12584724-1--d87543-;rport
Call-ID: 3465f442118733ee4362573f6d4e0ba8 at ip.of.asterisk
CSeq: 4 ACK
Content-Length: 0
--- (7 headers 0 lines)---
<-- SIP read from ip.of.UA:6738:
INVITE sip:asterisk at ip.of.asterisk SIP/2.0
To: "asterisk"<sip:asterisk at ip.of.asterisk>;tag=as0a0f3604
From: <sip:233 at ip.of.UA:6738>;tag=fd79775f
Via: SIP/2.0/UDP ip.of.UA:6738;branch=z9hG4bK-d87543-867975881-1--d87543-;rport
Call-ID: 3465f442118733ee4362573f6d4e0ba8 at ip.of.asterisk
CSeq: 5 INVITE
Contact: <sip:233 at ip.of.UA:6738>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="233",realm="asterisk",nonce="534c6e42",uri="sip:asterisk at ip.of.asterisk",response="d43aaaeb35197c7ce2bb9c49bb797e87",algorithm=MD5
User-Agent: eyeBeam release 3004t stamp 16741
Content-Length: 234
v=0
o=- 14115490 14635628 IN IP4 ip.of.UA
s=eyeBeam
c=IN IP4 0.0.0.0
t=0 0
m=audio 6740 RTP/AVP 18 0 8 101
a=alt:1 1 : 51B3EDC9 554BDE6A ip.of.UA 6740
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendonly
--- (13 headers 10 lines)---
Using INVITE request as basis request - 3465f442118733ee4362573f6d4e0ba8 at ip.of.asterisk
Sending to ip.of.UA : 6738 (NAT)
Found user '233'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 0.0.0.0:6740
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for asterisk in nacionales (domain ip.of.asterisk)
Reliably Transmitting (no NAT) to ip.of.UA:6738:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP ip.of.UA:6738;branch=z9hG4bK-d87543-867975881-1--d87543-;rport;received=ip.of.UA
From: <sip:233 at ip.of.UA:6738>;tag=fd79775f
To: "asterisk"<sip:asterisk at ip.of.asterisk>;tag=as0a0f3604
Call-ID: 3465f442118733ee4362573f6d4e0ba8 at ip.of.asterisk
CSeq: 5 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:asterisk at ip.of.asterisk>
Content-Length: 0
<-- SIP read from ip.of.UA:6738:
ACK sip:asterisk at ip.of.asterisk SIP/2.0
To: "asterisk"<sip:asterisk at ip.of.asterisk>;tag=as0a0f3604
From: <sip:233 at ip.of.UA:6738>;tag=fd79775f
Via: SIP/2.0/UDP ip.of.UA:6738;branch=z9hG4bK-d87543-867975881-1--d87543-;rport
Call-ID: 3465f442118733ee4362573f6d4e0ba8 at ip.of.asterisk
CSeq: 5 ACK
Content-Length: 0
--- (7 headers 0 lines)---
Destroying call '3465f442118733ee4362573f6d4e0ba8 at ip.of.asterisk'
<-- SIP read from ip.of.UA:6738:
INVITE sip:asterisk at ip.of.asterisk SIP/2.0
To: "asterisk"<sip:asterisk at ip.of.asterisk>;tag=as4e5acd84
From: <sip:233 at ip.of.UA:6738>;tag=26682c38
Via: SIP/2.0/UDP ip.of.UA:6738;branch=z9hG4bK-d87543-1036132300-1--d87543-;rport
Call-ID: 5396c39e4f04e4f14a52b2be7b54fe10 at ip.of.asterisk
CSeq: 2 INVITE
Contact: <sip:233 at ip.of.UA:6738>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 3004t stamp 16741
Content-Length: 234
v=0
o=- 14631840 14636804 IN IP4 ip.of.UA
s=eyeBeam
c=IN IP4 0.0.0.0
t=0 0
m=audio 6744 RTP/AVP 18 0 8 101
a=alt:1 1 : 9EECBE2D 6FD6ECEA ip.of.UA 6744
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendonly
--- (12 headers 10 lines)---
Using INVITE request as basis request - 5396c39e4f04e4f14a52b2be7b54fe10 at ip.of.asterisk
Sending to ip.of.UA : 6738 (NAT)
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 0.0.0.0:6744
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
-- Started music on hold, class 'default', on Zap/2-1
<-- SIP read from ip.of.UA:6738:
INVITE sip:asterisk at ip.of.asterisk SIP/2.0
To: "asterisk"<sip:asterisk at ip.of.asterisk>;tag=as4e5acd84
From: <sip:233 at ip.of.UA:6738>;tag=26682c38
Via: SIP/2.0/UDP ip.of.UA:6738;branch=z9hG4bK-d87543-1036132300-1--d87543-;rport
Call-ID: 5396c39e4f04e4f14a52b2be7b54fe10 at ip.of.asterisk
CSeq: 2 INVITE
Contact: <sip:233 at ip.of.UA:6738>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 3004t stamp 16741
Content-Length: 234
v=0
o=- 14631840 14636804 IN IP4 ip.of.UA
s=eyeBeam
c=IN IP4 0.0.0.0
t=0 0
m=audio 6744 RTP/AVP 18 0 8 101
a=alt:1 1 : 9EECBE2D 6FD6ECEA ip.of.UA 6744
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendonly
--- (12 headers 10 lines)---
Ignoring this INVITE request
<-- SIP read from ip.of.UA:6738:
INVITE sip:asterisk at ip.of.asterisk SIP/2.0
To: "asterisk"<sip:asterisk at ip.of.asterisk>;tag=as4e5acd84
From: <sip:233 at ip.of.UA:6738>;tag=26682c38
Via: SIP/2.0/UDP ip.of.UA:6738;branch=z9hG4bK-d87543-1036132300-1--d87543-;rport
Call-ID: 5396c39e4f04e4f14a52b2be7b54fe10 at ip.of.asterisk
CSeq: 2 INVITE
Contact: <sip:233 at ip.of.UA:6738>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 3004t stamp 16741
Content-Length: 234
v=0
o=- 14631840 14636804 IN IP4 ip.of.UA
s=eyeBeam
c=IN IP4 0.0.0.0
t=0 0
m=audio 6744 RTP/AVP 18 0 8 101
a=alt:1 1 : 9EECBE2D 6FD6ECEA ip.of.UA 6744
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendonly
--- (12 headers 10 lines)---
Ignoring this INVITE request
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