Strange SIP Problem - no audio to asterisk

Julisade · September 12, 2006, 6:38pm

I’ve been trying to figure this one out for a couple days without a solution from any of the forums I could find. I have Asterisk installed on CentOS, and have created two SIP extensions. One happens to be a Cisco 7960 IP Phone, and the other a Polycom IP 501.

Anyway, I can call the extension of either phone and Asterisk does a native bridge and everything works OK. The issue is that any calls made into Asterisk (voicemail, etc) just plain do nothing. It says it’s playing the audio (-- Playing ‘vm-login’ (language ‘en’)", but I hear nothing from either extension, and it just stalls there until I hangup.

The phones are NOT behind a NAT, they are plugged into the same switch as the Asterisk box and on the same IP range. (I have tried nat=no and nat=never in sip.conf)

Both of these phones work on my company’s production Asterisk system, which is an Asterisk@Home box. I’ve even copied sip.conf for the two phones without any luck. Earlier, I started from scratch completely formatting the hard drive and installing Linux and then Asterisk and still have the same problem.

The Asterisk version is 1.2.9.1 svn rev 34876.

Any ideas are appreciated. Thanks in advance.

IronHelix · September 12, 2006, 8:22pm

so you have (I am assuming):
(internet) – NAT router – Asterisk server – IP phones.
The phones can call each other but not Asterisk…

First I would suggest try canreinvite=no in sip.conf. Also make sure externip= and localnet= are filled out, that can sometimes help. Lastly, for the IP phones make sure nat=no, and the phones are not using STUN or anything like that. If that doesnt help, post a sip debug of a working call and a non working call…

Julisade · September 12, 2006, 9:08pm

[code]asterisk1*CLI>
<-- SIP read from 192.168.13.140:50682:
INVITE sip:102@192.168.13.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.140:5060;branch=z9hG4bK005c62c9
From: “003094c450fc” sip:003094c450fc@192.168.13.11;tag=003094c450fc00e324d4cf1a-0695eeeb
To: sip:102@192.168.13.11
Call-ID: 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
CSeq: 101 INVITE
User-Agent: CSCO/6
Contact: sip:003094c450fc@192.168.13.140:5060
Expires: 180
Content-Type: application/sdp
Content-Length: 248
Accept: application/sdp

v=0
o=Cisco-SIPUA 10651 44 IN IP4 192.168.13.140
s=SIP Call
c=IN IP4 192.168.13.140
t=0 0
m=audio 25030 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

— (12 headers 11 lines)—
Using INVITE request as basis request - 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
Sending to 192.168.13.140 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.13.140:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.13.140:5060;branch=z9hG4bK005c62c9;received=192.168.13.140
From: “003094c450fc” sip:003094c450fc@192.168.13.11;tag=003094c450fc00e324d4cf1a-0695eeeb
To: sip:102@192.168.13.11;tag=as3174240b
Call-ID: 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:102@192.168.13.11
Proxy-Authenticate: Digest realm=“asterisk”, nonce="17205aaf"
Content-Length: 0

—erisk1CLI>
Scheduling destruction of call ‘003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140’ in 15000 ms
Found user '003094c450fc’
asterisk1CLI>
<-- SIP read from 192.168.13.140:50896:
ACK sip:102@192.168.13.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.140:5060;branch=z9hG4bK005c62c9
From: “003094c450fc” sip:003094c450fc@192.168.13.11;tag=003094c450fc00e324d4cf1a-0695eeeb
To: sip:102@192.168.13.11;tag=as3174240b
Call-ID: 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
CSeq: 101 ACK
Content-Length: 0

— (7 headers 0 lines)—
asterisk1*CLI>
<-- SIP read from 192.168.13.140:50682:
INVITE sip:102@192.168.13.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.140:5060;branch=z9hG4bK3dba8eb7
From: “003094c450fc” sip:003094c450fc@192.168.13.11;tag=003094c450fc00e324d4cf1a-0695eeeb
To: sip:102@192.168.13.11
Call-ID: 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
CSeq: 102 INVITE
User-Agent: CSCO/6
Contact: sip:003094c450fc@192.168.13.140:5060
Proxy-Authorization: Digest username=“003094c450fc”,realm=“asterisk”,uri=“sip:192.168.13.11”,response=“56ff1b328e0c6ba5e82cebf11c5408a4”,nonce=“17205aaf”,algorithm=md5
Expires: 180
Content-Type: application/sdp
Content-Length: 248

v=0
o=Cisco-SIPUA 10651 44 IN IP4 192.168.13.140
s=SIP Call
c=IN IP4 192.168.13.140
t=0 0
m=audio 25030 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

— (12 headers 11 lines)—
Using INVITE request as basis request - 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
Sending to 192.168.13.140 : 5060 (non-NAT)
Found user '003094c450fc’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.13.140:25030
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 102 in from-internal (domain 192.168.13.11)
list_route: hop: sip:003094c450fc@192.168.13.140:5060
Transmitting (no NAT) to 192.168.13.140:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.13.140:5060;branch=z9hG4bK3dba8eb7;received=192.168.13.140
From: “003094c450fc” sip:003094c450fc@192.168.13.11;tag=003094c450fc00e324d4cf1a-0695eeeb
To: sip:102@192.168.13.11
Call-ID: 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:102@192.168.13.11
Content-Length: 0

-- Executing AGI("SIP/003094c450fc-ad3c", "simicomm/from-internal.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/simicomm/from-internal.php
-- AGI Script Executing Application: (Dial) Options: (SIP/0004F204AB54|15|tr)

We’re at 192.168.13.11 port 19570
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.13.142:5060:
INVITE sip:0004f204ab54@192.168.13.142 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK65aef169;rport
From: “Test2” sip:102@192.168.13.11;tag=as18db8e63
To: sip:0004f204ab54@192.168.13.142
Contact: sip:102@192.168.13.11
Call-ID: 7297930405a0cfc02718a2242f0a405f@192.168.13.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Sep 2006 21:06:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 28932 28932 IN IP4 192.168.13.11
s=session
c=IN IP4 192.168.13.11
t=0 0
m=audio 19570 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

-- Called 0004F204AB54

Transmitting (no NAT) to 192.168.13.140:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.13.140:5060;branch=z9hG4bK3dba8eb7;received=192.168.13.140
From: “003094c450fc” sip:003094c450fc@192.168.13.11;tag=003094c450fc00e324d4cf1a-0695eeeb
To: sip:102@192.168.13.11;tag=as53823535
Call-ID: 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:102@192.168.13.11
Content-Length: 0

asterisk1*CLI>
<-- SIP read from 192.168.13.142:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK65aef169;rport
From: “Test2” sip:102@192.168.13.11;tag=as18db8e63
To: sip:0004f204ab54@192.168.13.142;tag=BC6A9877-2C4FBC92
CSeq: 102 INVITE
Call-ID: 7297930405a0cfc02718a2242f0a405f@192.168.13.11
Contact: sip:0004f204ab54@192.168.13.142
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Content-Length: 0

— (9 headers 0 lines)—
asterisk1*CLI>
<-- SIP read from 192.168.13.142:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK65aef169;rport
From: “Test2” sip:102@192.168.13.11;tag=as18db8e63
To: sip:0004f204ab54@192.168.13.142;tag=BC6A9877-2C4FBC92
CSeq: 102 INVITE
Call-ID: 7297930405a0cfc02718a2242f0a405f@192.168.13.11
Contact: sip:0004f204ab54@192.168.13.142
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Allow-Events: talk,hold,conference
Content-Length: 0

— (10 headers 0 lines)—
– SIP/0004F204AB54-057e is ringing
asterisk1*CLI>
<-- SIP read from 192.168.13.142:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK65aef169;rport
From: “Test2” sip:102@192.168.13.11;tag=as18db8e63
To: sip:0004f204ab54@192.168.13.142;tag=BC6A9877-2C4FBC92
CSeq: 102 INVITE
Call-ID: 7297930405a0cfc02718a2242f0a405f@192.168.13.11
Contact: sip:0004f204ab54@192.168.13.142
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Content-Type: application/sdp
Content-Length: 191

v=0
o=- 1158094931 1158094931 IN IP4 192.168.13.142
s=Polycom IP Phone
c=IN IP4 192.168.13.142
t=0 0
m=audio 2242 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

— (11 headers 8 lines)—
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.13.142:2242
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: sip:0004f204ab54@192.168.13.142
set_destination: Parsing sip:0004f204ab54@192.168.13.142 for address/port to send to
set_destination: set destination to 192.168.13.142, port 5060
Transmitting (no NAT) to 192.168.13.142:5060:
ACK sip:0004f204ab54@192.168.13.142 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK4a5bf949;rport
From: “Test2” sip:102@192.168.13.11;tag=as18db8e63
To: sip:0004f204ab54@192.168.13.142;tag=BC6A9877-2C4FBC92
Contact: sip:102@192.168.13.11
Call-ID: 7297930405a0cfc02718a2242f0a405f@192.168.13.11
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

-- SIP/0004F204AB54-057e answered SIP/003094c450fc-ad3c

We’re at 192.168.13.11 port 13196
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.13.140:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.13.140:5060;branch=z9hG4bK3dba8eb7;received=192.168.13.140
From: “003094c450fc” sip:003094c450fc@192.168.13.11;tag=003094c450fc00e324d4cf1a-0695eeeb
To: sip:102@192.168.13.11;tag=as53823535
Call-ID: 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:102@192.168.13.11
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 28932 28932 IN IP4 192.168.13.11
s=session
c=IN IP4 192.168.13.11
t=0 0
m=audio 13196 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

-- Attempting native bridge of SIP/003094c450fc-ad3c and SIP/0004F204AB54-057e

asterisk1*CLI>
<-- SIP read from 192.168.13.140:50682:
ACK sip:102@192.168.13.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.140:5060;branch=z9hG4bK4a3533a1
From: “003094c450fc” sip:003094c450fc@192.168.13.11;tag=003094c450fc00e324d4cf1a-0695eeeb
To: sip:102@192.168.13.11;tag=as53823535
Call-ID: 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
CSeq: 102 ACK
User-Agent: CSCO/6
Content-Length: 0

— (8 headers 0 lines)—
asterisk1*CLI>
<-- SIP read from 192.168.13.142:5060:
BYE sip:102@192.168.13.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.142;branch=z9hG4bKfb9ff6a0C864EFDD
From: sip:0004f204ab54@192.168.13.142;tag=BC6A9877-2C4FBC92
To: “Test2” sip:102@192.168.13.11;tag=as18db8e63
CSeq: 1 BYE
Call-ID: 7297930405a0cfc02718a2242f0a405f@192.168.13.11
Contact: sip:0004f204ab54@192.168.13.142
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0

— (10 headers 0 lines)—
Sending to 192.168.13.142 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.13.142:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.13.142;branch=z9hG4bKfb9ff6a0C864EFDD;received=192.168.13.142
From: sip:0004f204ab54@192.168.13.142;tag=BC6A9877-2C4FBC92
To: “Test2” sip:102@192.168.13.11;tag=as18db8e63
Call-ID: 7297930405a0cfc02718a2242f0a405f@192.168.13.11
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:102@192.168.13.11
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing

-- AGI Script simicomm/from-internal.php completed, returning 0

== Auto fallthrough, channel ‘SIP/003094c450fc-ad3c’ status is ‘ANSWER’
– Executing NoOp(“SIP/003094c450fc-ad3c”, “”) in new stack
set_destination: Parsing sip:003094c450fc@192.168.13.140:5060 for address/port to send to
set_destination: set destination to 192.168.13.140, port 5060
Reliably Transmitting (no NAT) to 192.168.13.140:5060:
BYE sip:003094c450fc@192.168.13.140:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK5db11695;rport
From: sip:102@192.168.13.11;tag=as53823535
To: “003094c450fc” sip:003094c450fc@192.168.13.11;tag=003094c450fc00e324d4cf1a-0695eeeb
Contact: sip:102@192.168.13.11
Call-ID: 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.13.140:5060:
OPTIONS sip:003094c450fc@192.168.13.140:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK2cd4f995;rport
From: “Unknown” sip:Unknown@192.168.13.11;tag=as03fbaec7
To: sip:003094c450fc@192.168.13.140:5060
Contact: sip:Unknown@192.168.13.11
Call-ID: 74785b9855f5fe660639984d3631e892@192.168.13.11
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Sep 2006 21:06:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

Destroying call '7297930405a0cfc02718a2242f0a405f@192.168.13.11’
asterisk1*CLI>
<-- SIP read from 192.168.13.140:50682:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK5db11695;rport
From: sip:102@192.168.13.11;tag=as53823535
To: “003094c450fc” sip:003094c450fc@192.168.13.11;tag=003094c450fc00e324d4cf1a-0695eeeb
Call-ID: 003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140
CSeq: 102 BYE
Server: CSCO/6
Content-Length: 0

— (8 headers 0 lines)—
Destroying call '003094c4-50fc0030-5b67d794-0432ec99@192.168.13.140’
asterisk1*CLI>
<-- SIP read from 192.168.13.140:50897:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK2cd4f995;rport
From: “Unknown” sip:Unknown@192.168.13.11;tag=as03fbaec7
To: sip:003094c450fc@192.168.13.140:5060;tag=003094c450fc00e46782dac0-125cec5b
Call-ID: 74785b9855f5fe660639984d3631e892@192.168.13.11
CSeq: 102 OPTIONS
Server: CSCO/6
Content-Type: application/sdp
Content-Length: 239
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER

v=0
o=Cisco-SIPUA 0 0 IN IP4 192.168.13.140
s=SIP Call
c=IN IP4 192.168.13.140
t=0 0
m=audio 1 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

— (10 headers 11 lines)—
Destroying call '74785b9855f5fe660639984d3631e892@192.168.13.11’
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.13.142:5060:
OPTIONS sip:0004f204ab54@192.168.13.142 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK1704a528;rport
From: “Unknown” sip:Unknown@192.168.13.11;tag=as7d7a0f35
To: sip:0004f204ab54@192.168.13.142
Contact: sip:Unknown@192.168.13.11
Call-ID: 42d5e1d47a0f15ea7d6f882126ac3876@192.168.13.11
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Sep 2006 21:06:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

asterisk1*CLI>
<-- SIP read from 192.168.13.142:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK1704a528;rport
From: “Unknown” sip:Unknown@192.168.13.11;tag=as7d7a0f35
To: sip:0004f204ab54@192.168.13.142;tag=70D8B5EC-E5C8B13
CSeq: 102 OPTIONS
Call-ID: 42d5e1d47a0f15ea7d6f882126ac3876@192.168.13.11
Contact: sip:0004f204ab54@192.168.13.142
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Content-Length: 0

— (10 headers 0 lines)—
Destroying call ‘42d5e1d47a0f15ea7d6f882126ac3876@192.168.13.11’
[/code]

Julisade · September 12, 2006, 9:09pm

[code]asterisk1*CLI>
<-- SIP read from 192.168.13.142:5060:
INVITE sip:*97@192.168.13.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.142;branch=z9hG4bK528eef7833B1B695
From: “0004f204ab54” sip:0004f204ab54@192.168.13.11;tag=2129B0C2-B905B4D1
To: sip:*97@192.168.13.11
CSeq: 1 INVITE
Call-ID: 67ba5156-2dc2cf44-9a91e80b@192.168.13.142
Contact: sip:0004f204ab54@192.168.13.142
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 253

v=0
o=- 1158095034 1158095034 IN IP4 192.168.13.142
s=Polycom IP Phone
c=IN IP4 192.168.13.142
t=0 0
a=sendrecv
m=audio 2244 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

— (14 headers 11 lines)—
Using INVITE request as basis request - 67ba5156-2dc2cf44-9a91e80b@192.168.13.142
Sending to 192.168.13.142 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.13.142:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.13.142;branch=z9hG4bK528eef7833B1B695;received=192.168.13.142
From: “0004f204ab54” sip:0004f204ab54@192.168.13.11;tag=2129B0C2-B905B4D1
To: sip:*97@192.168.13.11;tag=as340c66d8
Call-ID: 67ba5156-2dc2cf44-9a91e80b@192.168.13.142
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:*97@192.168.13.11
Proxy-Authenticate: Digest realm=“asterisk”, nonce="71b19e43"
Content-Length: 0

Scheduling destruction of call ‘67ba5156-2dc2cf44-9a91e80b@192.168.13.142’ in 15000 ms
Found user '0004f204ab54’
asterisk1*CLI>
<-- SIP read from 192.168.13.142:5060:
ACK sip:*97@192.168.13.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.142;branch=z9hG4bK528eef7833B1B695
From: “0004f204ab54” sip:0004f204ab54@192.168.13.11;tag=2129B0C2-B905B4D1
To: sip:*97@192.168.13.11;tag=as340c66d8
CSeq: 1 ACK
Call-ID: 67ba5156-2dc2cf44-9a91e80b@192.168.13.142
Contact: sip:0004f204ab54@192.168.13.142
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0

— (11 headers 0 lines)—
asterisk1*CLI>
<-- SIP read from 192.168.13.142:5060:
INVITE sip:*97@192.168.13.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.142;branch=z9hG4bK72688a67E278C150
From: “0004f204ab54” sip:0004f204ab54@192.168.13.11;tag=2129B0C2-B905B4D1
To: sip:*97@192.168.13.11
CSeq: 2 INVITE
Call-ID: 67ba5156-2dc2cf44-9a91e80b@192.168.13.142
Contact: sip:0004f204ab54@192.168.13.142
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username=“0004f204ab54”, realm=“asterisk”, nonce=“71b19e43”, uri=“sip:*97@192.168.13.11:5060”, response=“bea32f2f91c8ead2be0a6b1c7cfca7e0”, algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 253

v=0
o=- 1158095034 1158095034 IN IP4 192.168.13.142
s=Polycom IP Phone
c=IN IP4 192.168.13.142
t=0 0
a=sendrecv
m=audio 2244 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

— (15 headers 11 lines)—
Using INVITE request as basis request - 67ba5156-2dc2cf44-9a91e80b@192.168.13.142
Sending to 192.168.13.142 : 5060 (non-NAT)
Found user '0004f204ab54’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.13.142:2244
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for *97 in from-internal (domain 192.168.13.11)
list_route: hop: sip:0004f204ab54@192.168.13.142
Transmitting (no NAT) to 192.168.13.142:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.13.142;branch=z9hG4bK72688a67E278C150;received=192.168.13.142
From: “0004f204ab54” sip:0004f204ab54@192.168.13.11;tag=2129B0C2-B905B4D1
To: sip:*97@192.168.13.11
Call-ID: 67ba5156-2dc2cf44-9a91e80b@192.168.13.142
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:*97@192.168.13.11
Content-Length: 0

-- Executing AGI("SIP/0004f204ab54-e61c", "simicomm/from-internal.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/simicomm/from-internal.php

We’re at 192.168.13.11 port 18110
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.13.142:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.13.142;branch=z9hG4bK72688a67E278C150;received=192.168.13.142
From: “0004f204ab54” sip:0004f204ab54@192.168.13.11;tag=2129B0C2-B905B4D1
To: sip:*97@192.168.13.11;tag=as1d8a2126
Call-ID: 67ba5156-2dc2cf44-9a91e80b@192.168.13.142
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:*97@192.168.13.11
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 28932 28932 IN IP4 192.168.13.11
s=session
c=IN IP4 192.168.13.11
t=0 0
m=audio 18110 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

-- AGI Script Executing Application: (VoiceMailMain) Options: ((null))
-- Playing 'vm-login' (language 'en')

asterisk1*CLI>
<-- SIP read from 192.168.13.142:5060:
ACK sip:*97@192.168.13.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.142;branch=z9hG4bK4fc5719cAD3DB309
From: “0004f204ab54” sip:0004f204ab54@192.168.13.11;tag=2129B0C2-B905B4D1
To: sip:*97@192.168.13.11;tag=as1d8a2126
CSeq: 2 ACK
Call-ID: 67ba5156-2dc2cf44-9a91e80b@192.168.13.142
Contact: sip:0004f204ab54@192.168.13.142
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0

— (11 headers 0 lines)—
asterisk1*CLI>
<-- SIP read from 192.168.13.142:5060:
BYE sip:*97@192.168.13.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.142;branch=z9hG4bK5e331c5a2D1A22DF
From: “0004f204ab54” sip:0004f204ab54@192.168.13.11;tag=2129B0C2-B905B4D1
To: sip:*97@192.168.13.11;tag=as1d8a2126
CSeq: 3 BYE
Call-ID: 67ba5156-2dc2cf44-9a91e80b@192.168.13.142
Contact: sip:0004f204ab54@192.168.13.142
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Proxy-Authorization: Digest username=“0004f204ab54”, realm=“asterisk”, nonce=“71b19e43”, uri=“sip:*97@192.168.13.11:5060”, response=“bea32f2f91c8ead2be0a6b1c7cfca7e0”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0

— (11 headers 0 lines)—
Sending to 192.168.13.142 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.13.142:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.13.142;branch=z9hG4bK5e331c5a2D1A22DF;received=192.168.13.142
From: “0004f204ab54” sip:0004f204ab54@192.168.13.11;tag=2129B0C2-B905B4D1
To: sip:*97@192.168.13.11;tag=as1d8a2126
Call-ID: 67ba5156-2dc2cf44-9a91e80b@192.168.13.142
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:*97@192.168.13.11
Content-Length: 0

-- Executing NoOp("SIP/0004f204ab54-e61c", "") in new stack

Destroying call '67ba5156-2dc2cf44-9a91e80b@192.168.13.142’
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.13.140:5060:
OPTIONS sip:003094c450fc@192.168.13.140:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK7090f729;rport
From: “Unknown” sip:Unknown@192.168.13.11;tag=as15d0f79b
To: sip:003094c450fc@192.168.13.140:5060
Contact: sip:Unknown@192.168.13.11
Call-ID: 4029119526d217137036aeb3202bb6d3@192.168.13.11
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Sep 2006 21:08:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

asterisk1*CLI>
<-- SIP read from 192.168.13.140:50899:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK7090f729;rport
From: “Unknown” sip:Unknown@192.168.13.11;tag=as15d0f79b
To: sip:003094c450fc@192.168.13.140:5060;tag=003094c450fc00e67e1ddd7e-6edb21e8
Call-ID: 4029119526d217137036aeb3202bb6d3@192.168.13.11
CSeq: 102 OPTIONS
Server: CSCO/6
Content-Type: application/sdp
Content-Length: 239
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER

v=0
o=Cisco-SIPUA 0 0 IN IP4 192.168.13.140
s=SIP Call
c=IN IP4 192.168.13.140
t=0 0
m=audio 1 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

— (10 headers 11 lines)—
Destroying call '4029119526d217137036aeb3202bb6d3@192.168.13.11’
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.13.142:5060:
OPTIONS sip:0004f204ab54@192.168.13.142 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK5c103788;rport
From: “Unknown” sip:Unknown@192.168.13.11;tag=as66d662a9
To: sip:0004f204ab54@192.168.13.142
Contact: sip:Unknown@192.168.13.11
Call-ID: 3865ae940e5bacfa149545474ac72dd9@192.168.13.11
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Sep 2006 21:08:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

asterisk1*CLI>
<-- SIP read from 192.168.13.142:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.13.11:5060;branch=z9hG4bK5c103788;rport
From: “Unknown” sip:Unknown@192.168.13.11;tag=as66d662a9
To: sip:0004f204ab54@192.168.13.142;tag=51220086-949C2B05
CSeq: 102 OPTIONS
Call-ID: 3865ae940e5bacfa149545474ac72dd9@192.168.13.11
Contact: sip:0004f204ab54@192.168.13.142
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Content-Length: 0

— (10 headers 0 lines)—
Destroying call ‘3865ae940e5bacfa149545474ac72dd9@192.168.13.11’
[/code]

IronHelix · September 12, 2006, 9:33pm

both of those traces look fine unless I am missing something.

Try making a very simple extension… just Answer() then Playback() something. Does that work?
Also try Answer() before the agi…

Julisade · September 12, 2006, 11:08pm

I tried what you suggested, adding this to extensions.conf:

exten => 123,1,Answer();
exten => 123,2,Playback(beep);

On the console, when I dial 123 I see the message “-- Playing ‘beep’ (language ‘en’)”, but hear nothing on the extension. Nothing else shows up on the console until I hangup, and then it executes NoOp because I have exten => h,1,NoOp(); in extensions.conf.

I tried answering before the AGI, but this is a script that I programmed which I know answers before it does anything else.

If you think it’d be helpful, I could post the debug again, I’m not an expert at SIP, but everything looked good to me too from what I know of how it works.

I thought possibly it was my AGI, so I tried the sample extensions.conf with no luck as well. Is there anything else I can do, short of give up on this hardware? I’m deploying Asterisk on a new Pentium 3.06 GHz system, maybe it’s just not compatible for some reason?

IronHelix · September 12, 2006, 11:41pm

just curious, is zaptel installed?

try running the zttest tool in zaptel source folder… what does that give you?

also try re download/compile/install asterisk and zaptel, see if that helps at all…

Julisade · September 12, 2006, 11:45pm

Yes, zaptel should be installed. I have a Digium Wildcard TE110P in the system that I haven’t configured yet. When I run zttest, it says “Opened pseudo zap interface, measuring accuracy…” and seems to stall.

I’ve read about people having issues with zaptel and smp, is this potentially a problem with that?

Julisade · September 13, 2006, 1:18am

If I unload the zaptel drivers and then restart asterisk it works. BUT, I have a T1 card in this system because that’s how it interfaces with the PSTN. So, how can I have zaptel AND sip working?