[Asterisk-bsd] Zap on hold problem
Diego Valencia
dvalencia at powervt.com.ar
Mon Mar 6 12:47:49 MST 2006
Hi, anybody knows if is normal the "Ignoring this INVITE request"?:
The call is incoming from zap channel, this invite is when I put the call on
hold, and the UA does not get a response.
Thanks
Diego
<-- 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
----- Original Message -----
From: "dvalencia" <dvalencia at powervt.com.ar>
To: <asterisk-bsd at lists.digium.com>
Sent: Sunday, March 05, 2006 2:12 PM
Subject: [Asterisk-bsd] Zap on hold problem
> 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
> 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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