[asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped
Stuart Sheldon
stu at actusa.net
Tue Mar 20 20:47:00 MST 2007
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Are you using Answer() before VoiceMailMain()?
Stu
Timothy McKee wrote:
> I've been running the 8/1/2004 Head release up until a little over a
> week ago. I was forced to due to a card failure to upgrade to 1.2.16
> without any advance preparation or testing (most of my connections are
> via satellite to all corners of the globe with high latency).
>
> Up until the upgrade I was running with very few issues. Since the
> upgrade I have been experiencing strange issues with my Polycom SP-601
> phones. My customers attempt to get their voicemail and Asterisk drops
> their connection ~15 seconds after they dial VM. I have captured a SIP
> debug and included it (somewhat sanitized). I'm not a SIP guru, but I
> can see the 15 second timer being set and I see repeated INVITEs being
> sent without any acks. OPTIONs are being sent and acked. The remote
> SIP phone is 'eden-1000a' and the voicemail extension is 9990. *This
> worked just fine up until the upgrade.*
>
> Does this ring a bell with anyone out there???
>
> Tim McKee
> <tmckee at sdnglobal dot com>
> SDN Global
>
> ==============================================
>
> pbx*CLI> sip debug peer eden-1000a
> SIP Debugging Enabled for IP: 10.253.4.50:5060
> pbx*CLI>
> <-- SIP read from 10.253.4.50:5060:
> INVITE sip:9990 at hostname.company.domain;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>
> CSeq: 1 INVITE
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> Contact: <sip:eden-1000a at 10.253.4.50>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
> Supported: 100rel,replaces
> Allow-Events: talk,hold,conference
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 245
>
> v=0
> o=- 978307756 978307756 IN IP4 10.253.4.50
> s=Polycom IP Phone
> c=IN IP4 10.253.4.50
> t=0 0
> m=audio 2228 RTP/AVP 0 18 8 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
>
> --- (14 headers 11 lines) ---
> Using INVITE request as basis request -
> a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> Sending to 10.253.4.50 : 5060 (NAT)
> Reliably Transmitting (no NAT) to 10.253.4.50:5060:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as7f808f0f
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
> nonce="2584558d"
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call 'a857d7ac-36f29d46-4d6ef889 at 10.253.4.50'
> in 15000 ms
> Found user 'eden-1000a'
> pbx*CLI>
> <-- SIP read from 10.253.4.50:5060:
> INVITE sip:9990 at hostname.company.domain;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>
> CSeq: 1 INVITE
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> Contact: <sip:eden-1000a at 10.253.4.50>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
> Supported: 100rel,replaces
> Allow-Events: talk,hold,conference
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 245
>
> v=0
> o=- 978307756 978307756 IN IP4 10.253.4.50
> s=Polycom IP Phone
> c=IN IP4 10.253.4.50
> t=0 0
> m=audio 2228 RTP/AVP 0 18 8 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
>
> --- (14 headers 11 lines) ---
> Ignoring this INVITE request
> pbx*CLI>
> <-- SIP read from 10.253.4.50:5060:
> ACK sip:9990 at hostname.company.domain SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as7f808f0f
> CSeq: 1 ACK
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> Contact: <sip:eden-1000a at 10.253.4.50>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
> Max-Forwards: 70
> Content-Length: 0
>
>
> --- (11 headers 0 lines) ---
> pbx*CLI>
> <-- SIP read from 10.253.4.50:5060:
> INVITE sip:9990 at hostname.company.domain;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>
> CSeq: 2 INVITE
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> Contact: <sip:eden-1000a at 10.253.4.50>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
> Supported: 100rel,replaces
> Allow-Events: talk,hold,conference
> Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
> nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
> response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 245
>
> v=0
> o=- 978307756 978307756 IN IP4 10.253.4.50
> s=Polycom IP Phone
> c=IN IP4 10.253.4.50
> t=0 0
> m=audio 2228 RTP/AVP 0 18 8 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
>
> --- (15 headers 11 lines) ---
> Using INVITE request as basis request -
> a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> Sending to 10.253.4.50 : 5060 (non-NAT)
> Found user 'eden-1000a'
> Found RTP audio format 0
> Found RTP audio format 18
> Found RTP audio format 8
> Found RTP audio format 101
> Peer audio RTP is at port 10.253.4.50:2228
> Peer video RTP is at port 10.253.4.50:65535
> Found description format PCMU
> Found description format G729
> Found description format PCMA
> Found description format telephone-event
> Capabilities: us - 0x100 (g729), peer - audio=0x10c
> (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729)
> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
> Looking for 9990 in eden-dialout (domain
> hostname.company.domain;user=phone)
> list_route: hop: <sip:eden-1000a at 10.253.4.50>
> Transmitting (no NAT) to 10.253.4.50:5060:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Length: 0
>
>
> ---
> -- Executing Answer("SIP/eden-1000a-4150cc98", "") in new stack
> We're at 172.30.42.5 port 29816
> Video is at 172.30.42.5 port 29214
> Adding codec 0x100 (g729) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> ontent-Length: 235
>
> v=0
> o=root 5641 5641 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> -- Executing VoiceMailMain("SIP/eden-1000a-4150cc98", "1000 at eden")
> in new stack
> -- Playing 'vm-password' (language 'en')
> pbx*CLI>
> <-- SIP read from 10.253.4.50:5060:
> INVITE sip:9990 at hostname.company.domain;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>
> CSeq: 2 INVITE
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> Contact: <sip:eden-1000a at 10.253.4.50>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
> Supported: 100rel,replaces
> Allow-Events: talk,hold,conference
> Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
> nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
> response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 245
>
> v=0
> o=- 978307756 978307756 IN IP4 10.253.4.50
> s=Polycom IP Phone
> c=IN IP4 10.253.4.50
> t=0 0
> m=audio 2228 RTP/AVP 0 18 8 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
>
> --- (15 headers 11 lines) ---
> Ignoring this INVITE request
> We're at 172.30.42.5 port 29816
> Video is at 172.30.42.5 port 29214
> Adding codec 0x100 (g729) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> ontent-Length: 235
>
> v=0
> o=root 5641 5642 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> pbx*CLI>
> <-- SIP read from 10.253.4.50:5060:
> INVITE sip:9990 at hostname.company.domain;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>
> CSeq: 2 INVITE
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> Contact: <sip:eden-1000a at 10.253.4.50>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
> Supported: 100rel,replaces
> Allow-Events: talk,hold,conference
> Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
> nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
> response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 245
>
> v=0
> o=- 978307756 978307756 IN IP4 10.253.4.50
> s=Polycom IP Phone
> c=IN IP4 10.253.4.50
> t=0 0
> m=audio 2228 RTP/AVP 0 18 8 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
>
> --- (15 headers 11 lines) ---
> Ignoring this INVITE request
> We're at 172.30.42.5 port 29816
> Video is at 172.30.42.5 port 29214
> Adding codec 0x100 (g729) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> ontent-Length: 235
>
> v=0
> o=root 5641 5643 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> pbx*CLI>
> <-- SIP read from 10.253.4.50:5060:
> ACK sip:9990 at 172.30.42.5 SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK1674aeae5EA3A4B
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> CSeq: 2 ACK
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> Contact: <sip:eden-1000a at 10.253.4.50>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
> Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
> nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
> response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5
> Max-Forwards: 70
> Content-Length: 0
>
>
> --- (12 headers 0 lines) ---
> Retransmitting #1 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5641 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> Retransmitting #1 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5642 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> Retransmitting #2 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5641 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> Retransmitting #2 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5642 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> -- Playing 'vm-youhave' (language 'en')
> -- Playing 'digits/1' (language 'en')
> Retransmitting #3 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5641 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> Retransmitting #3 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5642 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> -- Playing 'vm-Old' (language 'en')
> -- Playing 'vm-message' (language 'en')
> -- Playing 'vm-onefor' (language 'en')
> -- Playing 'digits/7' (language 'en')
> -- Playing 'vm-Old' (language 'en')
> -- Playing 'vm-first' (language 'en')
> Retransmitting #4 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5641 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> -- Playing 'vm-message' (language 'en')
> Retransmitting #4 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5642 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> == Parsing '/var/spool/asterisk/voicemail/eden/1000/Old/msg0000.txt':
> Found
> -- Playing 'vm-received' (language 'en')
> -- Playing 'digits/at' (language 'en')
> -- Playing 'digits/17' (language 'en')
> -- Playing 'digits/hundred' (language 'en')
> Retransmitting #5 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5641 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> -- SIP/cp-0821a7d8 is making progress passing it to IAX2/acppbx-102
> Retransmitting #5 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5642 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> -- Playing 'digits/50' (language 'en')
> -- Playing 'digits/5' (language 'en')
> -- Playing 'hours' (language 'en')
> -- Playing '/var/spool/asterisk/voicemail/eden/1000/Old/msg0000'
> (language 'en')
> 12 headers, 0 lines
> Reliably Transmitting (no NAT) to 10.253.4.50:5060:
> OPTIONS sip:eden-1000a at 10.253.4.50 SIP/2.0
> Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
> From: "asterisk" <sip:asterisk at 172.30.42.5>;tag=as021e29c4
> To: <sip:eden-1000a at 10.253.4.50>
> Contact: <sip:asterisk at 172.30.42.5>
> Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4 at 172.30.42.5
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Tue, 20 Mar 2007 23:01:40 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> Retransmitting #6 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5641 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> Retransmitting #6 (no NAT) to 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990 at 172.30.42.5>
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 5641 5642 IN IP4 172.30.42.5
> s=session
> c=IN IP4 172.30.42.5
> t=0 0
> m=audio 29816 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> Retransmitting #1 (no NAT) to 10.253.4.50:5060:
> OPTIONS sip:eden-1000a at 10.253.4.50 SIP/2.0
> Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
> From: "asterisk" <sip:asterisk at 172.30.42.5>;tag=as021e29c4
> To: <sip:eden-1000a at 10.253.4.50>
> Contact: <sip:asterisk at 172.30.42.5>
> Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4 at 172.30.42.5
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Tue, 20 Mar 2007 23:01:40 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ontent-Length: 0
>
>
> ---
> pbx*CLI> exit
> <-- SIP read from 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
> From: "asterisk" <sip:asterisk at 172.30.42.5>;tag=as021e29c4
> To: <sip:eden-1000a at 10.253.4.50>;tag=9E3B7462-6F180925
> CSeq: 102 OPTIONS
> Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4 at 172.30.42.5
> Contact: <sip:eden-1000a at 10.253.4.50>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
> Content-Length: 0
>
>
> --- (10 headers 0 lines) ---
> Destroying call '2a1ab9c42b63a0305f6de14715f4f8f4 at 172.30.42.5'
> pbx*CLI> exit
> <-- SIP read from 10.253.4.50:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
> From: "asterisk" <sip:asterisk at 172.30.42.5>;tag=as021e29c4
> To: <sip:eden-1000a at 10.253.4.50>;tag=9E3B7462-6F180925
> CSeq: 102 OPTIONS
> Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4 at 172.30.42.5
> Contact: <sip:eden-1000a at 10.253.4.50>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
> Content-Length: 0
>
>
> --- (10 headers 0 lines) ---
> Mar 20 18:01:44 WARNING[2770]: chan_sip.c:1228 retrans_pkt: Maximum
> retries exceeded on transmission a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> for seqno 2 (Critical Response)
> Mar 20 18:01:44 WARNING[2770]: chan_sip.c:1245 retrans_pkt: Hanging up
> call a857d7ac-36f29d46-4d6ef889 at 10.253.4.50 - no reply to our critical
> packet.
> == Spawn extension (eden-dialout, 9990, 2) exited non-zero on
> 'SIP/eden-1000a-4150cc98'
> Mar 20 18:01:45 WARNING[2770]: chan_sip.c:1228 retrans_pkt: Maximum
> retries exceeded on transmission a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> for seqno 2 (Non-critical Response)
> -- SIP/cp-0821a7d8 answered IAX2/acppbx-102
> Destroying call 'a857d7ac-36f29d46-4d6ef889 at 10.253.4.50'
> pbx*CLI> exit
> <-- SIP read from 10.253.4.50:5060:
> BYE sip:9990 at 172.30.42.5 SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> CSeq: 3 BYE
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> Contact: <sip:eden-1000a at 10.253.4.50>
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
> Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
> nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
> response="32687f30de53796b3ad2c3283d199984", algorithm=MD5
> Max-Forwards: 70
> Content-Length: 0
>
>
> --- (11 headers 0 lines) ---
> Transmitting (NAT) to 10.253.4.50:5060:
> SIP/2.0 481 Call leg/transaction does not exist
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK449f277f6319767C;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 3 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> pbx*CLI> exit
> <-- SIP read from 10.253.4.50:5060:
> BYE sip:9990 at 172.30.42.5 SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
> CSeq: 3 BYE
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> Contact: <sip:eden-1000a at 10.253.4.50>
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
> Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
> nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
> response="32687f30de53796b3ad2c3283d199984", algorithm=MD5
> Max-Forwards: 70
> Content-Length: 0
>
>
> --- (11 headers 0 lines) ---
> Transmitting (NAT) to 10.253.4.50:5060:
> SIP/2.0 481 Call leg/transaction does not exist
> Via: SIP/2.0/UDP
> 10.253.4.50;branch=z9hG4bK449f277f6319767C;received=10.253.4.50
> From: "eden-1000a"
> <sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990 at 111.111.111.111;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
> CSeq: 3 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> pbx*CLI> exit
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
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>
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