[asterisk-users] Re: [A*UG] What does 'trunk' mean in outgoing and incoming?

Tue Sep 12 14:23:58 MST 2006

Wayne Walker wrote:
> Sorry, been out of touch for a while.  Run asterisk -r , then
> 
> set debug 30
> set verbose 30
> sip debug
> 
> Now call the phone line that will cause a ring on the Sipura.  Send us
> the sip and normal asterisk debug out put that occurs during  the ringing.
> 
> 
> On Fri, Sep 01, 2006 at 07:42:08PM -0500, Larry Alkoff wrote:
>> I'm configuring a Sipura SPA-3000 to go with my existing and working 
>> Asterisk 1.2.5 setup.
>>
>> The Sipura configuration files give an extension context [201] in 
>> sip.conf with the instruction "This goes into the Incoming settings for 
>> your Trunk".
>>
>> It also gives a extension context of [pstn-spa3k] in sip.conf with the 
>> instruction "This section goes into the Outgoing Settings for your Trunk".
>>
>> What does 'Incoming' and 'Outgoing' settings for your trunk mean?
>> Where do trunks live and what are they meant to do?
>>
>> In my setup I have in sip.conf a [telasip-gw] context that references a 
>> context=telasip-in in extensions.conf.
>>
>> In extensions I have a [telasip-in] and [telasip-out] context.
>>
>> Which if any of these are 'trunks'?
>>
>> The Future of Telephony doesn't say much about trunks.
>>
>> Larry

Attached is my sip.conf and extensions.conf - zip to conserve bw.
Below is the result of a call with the debug instructions you sent on.

It may start with the tail end of a call I aborted.  For some reason, 
Asterisk keeps trying to destroy the call over and over.

tillie*CLI>
Sep 12 16:06:41 NOTICE[3271]: chan_sip.c:5242 sip_reregister:    -- 
Re-registration for  lalkoff at gw3.telasip.com
REGISTER 13 headers, 0 lines
REGISTER attempt 1 to lalkoff at gw3.telasip.com
Reliably Transmitting (NAT) to 4.79.19.58:5060:
REGISTER sip:gw3.telasip.com SIP/2.0
Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK512c64ef;rport
From: <sip:lalkoff at gw3.telasip.com>;tag=as2575269c
To: <sip:lalkoff at gw3.telasip.com>
Call-ID: 1e2987044f31872c0c286c2b07378f0e at 192.168.0.22
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="lalkoff", realm="telasip.com", 
algorithm=MD5, uri="sip:gw3.telasip.com", nonce="508f794e", 
response="87133d4c6c53b72710140cc534c05d00", opaque=""
Expires: 120
Contact: <sip:5128796776 at 69.91.84.176>
Event: registration
Content-Length: 0

---
tillie*CLI>
<-- SIP read from 4.79.19.58:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
69.91.84.176:5060;branch=z9hG4bK512c64ef;received=69.91.84.176;rport=5060
From: <sip:lalkoff at gw3.telasip.com>;tag=as2575269c
To: <sip:lalkoff at gw3.telasip.com>
Call-ID: 1e2987044f31872c0c286c2b07378f0e at 192.168.0.22
CSeq: 106 REGISTER
User-Agent: Telasip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:lalkoff at 4.79.19.58>
Content-Length: 0

--- (10 headers 0 lines)---
tillie*CLI>
<-- SIP read from 4.79.19.58:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
69.91.84.176:5060;branch=z9hG4bK512c64ef;received=69.91.84.176;rport=5060
From: <sip:lalkoff at gw3.telasip.com>;tag=as2575269c
To: <sip:lalkoff at gw3.telasip.com>;tag=as53cb6b63
Call-ID: 1e2987044f31872c0c286c2b07378f0e at 192.168.0.22
CSeq: 106 REGISTER
User-Agent: Telasip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:lalkoff at 4.79.19.58>
WWW-Authenticate: Digest realm="telasip.com", nonce="13fd35f9"
Content-Length: 0

--- (11 headers 0 lines)---
Responding to challenge, registration to domain/host name gw3.telasip.com
REGISTER 13 headers, 0 lines
REGISTER attempt 2 to lalkoff at gw3.telasip.com
Reliably Transmitting (NAT) to 4.79.19.58:5060:
REGISTER sip:gw3.telasip.com SIP/2.0
Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK3074f777;rport
From: <sip:lalkoff at gw3.telasip.com>;tag=as14bc6108
To: <sip:lalkoff at gw3.telasip.com>
Call-ID: 1e2987044f31872c0c286c2b07378f0e at 192.168.0.22
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="lalkoff", realm="telasip.com", 
algorithm=MD5, uri="sip:gw3.telasip.com", nonce="13fd35f9", 
response="c9d9c0d070e2a039ffa5b60e1699d72a", opaque=""
Expires: 120
Contact: <sip:5128796776 at 69.91.84.176>
Event: registration
Content-Length: 0

---
tillie*CLI>
<-- SIP read from 4.79.19.58:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
69.91.84.176:5060;branch=z9hG4bK3074f777;received=69.91.84.176;rport=5060
From: <sip:lalkoff at gw3.telasip.com>;tag=as14bc6108
To: <sip:lalkoff at gw3.telasip.com>
Call-ID: 1e2987044f31872c0c286c2b07378f0e at 192.168.0.22
CSeq: 107 REGISTER
User-Agent: Telasip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:lalkoff at 4.79.19.58>
Content-Length: 0

--- (10 headers 0 lines)---
tillie*CLI>
<-- SIP read from 4.79.19.58:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
69.91.84.176:5060;branch=z9hG4bK3074f777;received=69.91.84.176;rport=5060
From: <sip:lalkoff at gw3.telasip.com>;tag=as14bc6108
To: <sip:lalkoff at gw3.telasip.com>;tag=as53cb6b63
Call-ID: 1e2987044f31872c0c286c2b07378f0e at 192.168.0.22
CSeq: 107 REGISTER
User-Agent: Telasip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 120
Contact: <sip:5128796776 at 69.91.84.176>;expires=120
Date: Tue, 12 Sep 2006 21:12:01 GMT
Content-Length: 0

--- (12 headers 0 lines)---
Scheduling destruction of call 
'1e2987044f31872c0c286c2b07378f0e at 192.168.0.22' in 32000 ms
Sep 12 16:06:42 NOTICE[3271]: chan_sip.c:9669 handle_response_register: 
Outbound Registration: Expiry for gw3.telasip.com is 120 sec (Scheduling 
reregistration in 105 s)
tillie*CLI>
<-- SIP read from 192.168.0.41:5061:
INVITE sip:SIP/120 at 192.168.0.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.41:5061;branch=z9hG4bK-e858ec4
From: ALKOFF LAWRENCE <sip:5123017666 at 192.168.0.22>;tag=4eff7f365f272bf4o1
To: <sip:SIP/120 at 192.168.0.22>
Call-ID: b3c549de-7a7a16c at 192.168.0.41
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PSTN-IN <sip:5123017666 at 192.168.0.41:5061>
Expires: 240
User-Agent: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 420
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 89907 89907 IN IP4 192.168.0.41
s=-
c=IN IP4 192.168.0.41
t=0 0
m=audio 16446 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (14 headers 19 lines)---
Using INVITE request as basis request - b3c549de-7a7a16c at 192.168.0.41
Sending to 192.168.0.41 : 5061 (NAT)
Found peer 'pstn-spa3k'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.41:16446
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x1f07ff 
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), 
peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), 
combined - 0x51d (g723|ulaw|alaw|g726|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Looking for SIP/120 in home (domain 192.168.0.22)
Reliably Transmitting (NAT) to 192.168.0.41:5061:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 
192.168.0.41:5061;branch=z9hG4bK-e858ec4;received=192.168.0.41
From: ALKOFF LAWRENCE <sip:5123017666 at 192.168.0.22>;tag=4eff7f365f272bf4o1
To: <sip:SIP/120 at 192.168.0.22>;tag=as57fd29f4
Call-ID: b3c549de-7a7a16c at 192.168.0.41
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:SIP/120 at 192.168.0.22>
Content-Length: 0

---
tillie*CLI>
<-- SIP read from 192.168.0.41:5061:
ACK sip:SIP/120 at 192.168.0.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.41:5061;branch=z9hG4bK-e858ec4
From: ALKOFF LAWRENCE <sip:5123017666 at 192.168.0.22>;tag=4eff7f365f272bf4o1
To: <sip:SIP/120 at 192.168.0.22>;tag=as57fd29f4
Call-ID: b3c549de-7a7a16c at 192.168.0.41
CSeq: 101 ACK
Max-Forwards: 70
Contact: PSTN-IN <sip:5123017666 at 192.168.0.41:5061>
User-Agent: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0

--- (10 headers 0 lines)---
Destroying call 'b3c549de-7a7a16c at 192.168.0.41'
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.124:5060:
OPTIONS sip:124 at 192.168.0.124:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK5d031b29;rport
From: "asterisk" <sip:asterisk at 192.168.0.22>;tag=as3a38c760
To: <sip:124 at 192.168.0.124:5060>
Contact: <sip:asterisk at 192.168.0.22>
Call-ID: 525910d626731f3c78673c915e2af396 at 192.168.0.22
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Sep 2006 21:07:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.112:5060:
OPTIONS sip:112 at 192.168.0.112:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK0bfe5594;rport
From: "asterisk" <sip:asterisk at 192.168.0.22>;tag=as5e5d0064
To: <sip:112 at 192.168.0.112:5060>
Contact: <sip:asterisk at 192.168.0.22>
Call-ID: 430e20cd2d16c7d62aa551ac0793fab2 at 192.168.0.22
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Sep 2006 21:07:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.110:5060:
OPTIONS sip:110 at 192.168.0.110:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK646607ef;rport
From: "asterisk" <sip:asterisk at 192.168.0.22>;tag=as3ce95fc5
To: <sip:110 at 192.168.0.110:5060;user=phone>
Contact: <sip:asterisk at 192.168.0.22>
Call-ID: 3ce4853d2ab7337d7926e1634e1e0d55 at 192.168.0.22
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Sep 2006 21:07:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.120:5060:
OPTIONS sip:120 at 192.168.0.120:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK089ba80a;rport
From: "asterisk" <sip:asterisk at 192.168.0.22>;tag=as513614bf
To: <sip:120 at 192.168.0.120:5060>
Contact: <sip:asterisk at 192.168.0.22>
Call-ID: 675ef5e1282dce1e53c921b668f9b775 at 192.168.0.22
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Sep 2006 21:07:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
tillie*CLI>
<-- SIP read from 192.168.0.124:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK5d031b29;rport
From: "asterisk" <sip:asterisk at 192.168.0.22>;tag=as3a38c760
To: <sip:124 at 192.168.0.124:5060>;tag=e14e575b1eaf58a0
Call-ID: 525910d626731f3c78673c915e2af396 at 192.168.0.22
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2000 1.1.0.16
Contact: <sip:124 at 192.168.0.124:5060>
Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0

--- (11 headers 0 lines)---
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.122:5060:
OPTIONS sip:122 at 192.168.0.122:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK244c9e3e;rport
From: "asterisk" <sip:asterisk at 192.168.0.22>;tag=as4bd4d56a
To: <sip:122 at 192.168.0.122:5060;user=phone>
Contact: <sip:asterisk at 192.168.0.22>
Call-ID: 49508eb65c016b010176919d45134a41 at 192.168.0.22
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Sep 2006 21:07:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
Destroying call '525910d626731f3c78673c915e2af396 at 192.168.0.22'
tillie*CLI>
<-- SIP read from 192.168.0.112:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK0bfe5594;rport
From: "asterisk" <sip:asterisk at 192.168.0.22>;tag=as5e5d0064
To: <sip:112 at 192.168.0.112:5060>;tag=eaedb459128a878d
Call-ID: 430e20cd2d16c7d62aa551ac0793fab2 at 192.168.0.22
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2000 1.1.0.16
Contact: <sip:112 at 192.168.0.112:5060>
Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0

--- (11 headers 0 lines)---
Destroying call '430e20cd2d16c7d62aa551ac0793fab2 at 192.168.0.22'
tillie*CLI>
<-- SIP read from 192.168.0.110:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK646607ef;rport
From: "asterisk" <sip:asterisk at 192.168.0.22>;tag=as3ce95fc5
To: <sip:110 at 192.168.0.110:5060;user=phone>;tag=b2da6304d37a8958
Call-ID: 3ce4853d2ab7337d7926e1634e1e0d55 at 192.168.0.22
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2000 1.1.0.16
Contact: <sip:110 at 192.168.0.110:5060;user=phone>
Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0

--- (11 headers 0 lines)---
Destroying call '3ce4853d2ab7337d7926e1634e1e0d55 at 192.168.0.22'
tillie*CLI>
<-- SIP read from 192.168.0.120:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK089ba80a;rport
From: "asterisk" <sip:asterisk at 192.168.0.22>;tag=as513614bf
To: <sip:120 at 192.168.0.120:5060>;tag=69d7179758c7edd4
Call-ID: 675ef5e1282dce1e53c921b668f9b775 at 192.168.0.22
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2000 1.1.0.16
Contact: <sip:120 at 192.168.0.120:5060>
Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0

--- (11 headers 0 lines)---
Destroying call '675ef5e1282dce1e53c921b668f9b775 at 192.168.0.22'
tillie*CLI>
<-- SIP read from 192.168.0.122:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK244c9e3e;rport
From: "asterisk" <sip:asterisk at 192.168.0.22>;tag=as4bd4d56a
To: <sip:122 at 192.168.0.122:5060;user=phone>;tag=c8cd12ea2205d29c
Call-ID: 49508eb65c016b010176919d45134a41 at 192.168.0.22
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2000 1.1.0.16
Contact: <sip:122 at 192.168.0.122:5060;user=phone>
Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0

--- (11 headers 0 lines)---
Destroying call '49508eb65c016b010176919d45134a41 at 192.168.0.22'
Destroying call '1e2987044f31872c0c286c2b07378f0e at 192.168.0.22'
12 headers, 0 lines
Reliably Transmitting (NAT) to 4.79.19.58:5060:
OPTIONS sip:gw3.telasip.com SIP/2.0
Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK4feee17c;rport
From: "asterisk" <sip:asterisk at 69.91.84.176>;tag=as0011d202
To: <sip:gw3.telasip.com>
Contact: <sip:asterisk at 69.91.84.176>
Call-ID: 695e4c876c421661051b5a143f542174 at 69.91.84.176
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Sep 2006 21:07:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
tillie*CLI>
<-- SIP read from 4.79.19.58:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK4feee17c
From: "asterisk" <sip:asterisk at 69.91.84.176>;tag=as0011d202
To: <sip:gw3.telasip.com>;tag=as0dbf1bab
Call-ID: 695e4c876c421661051b5a143f542174 at 69.91.84.176
CSeq: 102 OPTIONS
User-Agent: Telasip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4.79.19.58>
Accept: application/sdp
Content-Length: 0

--- (11 headers 0 lines)---
Destroying call '695e4c876c421661051b5a143f542174 at 69.91.84.176'
tillie*CLI>

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