[Asterisk-Dev] Bug in realtime SIP

Sat Aug 13 09:24:41 MST 2005

Well, for one, what you describe is NOT a bug in RealTime SIP. (damn I wish
people would stop doing that.)

Secondly, this does not belong on the developers list.

Since your UA has obviously registered, SIP RT has done its job.

Your problem now seems to be in your extensions. Do you have an extensions
line for 18005551212?

-Matthew

> From: "Chris A. Icide" <chris at netgeeks.net>
> Reply-To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Date: Fri, 12 Aug 2005 20:49:26 -0700
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Subject: [Asterisk-Dev] Bug in realtime SIP
> 
> Asterisk version:  CVS Head from 8/11/05
> 
> extconfig.conf
> 
> sipusers => mysql,zbx,sipusers
> sippeers => mysql,zbx,sippeers
> 
> 
> localhost*CLI> realtime mysql status
> Connected to zbx at localhost, port 3306 with username zbx for 27 minutes,
> 20 seconds.
> 
> 
> realtime load sippeers name 104 and realtime load sipusers 104 both show
> the sip device information on screen.
> 
> When the sip device registers (it's dynamic), sip show peers and sip
> show users show the following
> 
> sip show peers
> 104/104                    (Unspecified)    D          255.255.255.255
> 0        Unmonitored
> 
> sip show users
> 
> 
> No entry every shows up for users.  If I try to make a call with the
> device, I get 403 Forbidden
> 
> <-- SIP read from 192.168.254.16:5060:
> INVITE sip:18005551212 at 192.168.254.6 SIP/2.0
> Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c
> From: Sipura2000-2 <sip:104 at 192.168.254.6>;tag=ffbcad7047bef41co1
> To: <sip:18005551212 at 192.168.254.6>
> Call-ID: 11739700-af5297cc at 192.168.254.16
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: Sipura2000-2 <sip:104 at 192.168.254.16:5060>
> Expires: 240
> User-Agent: Sipura/SPA2000-2.0.10(c)
> Content-Length: 426
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
>                  
> 
> v=0
> o=- 266317 266317 IN IP4 192.168.254.16
> s=-
> c=IN IP4 192.168.254.16
> t=0 0
> m=audio 16384 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 - 11739700-af5297cc at 192.168.254.16
> Sending to 192.168.254.16 : 5060 (non-NAT)
> Found no matching peer or user for '192.168.254.16:5060'
> 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 192.168.254.16:16384
> 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 - 0x404 (ulaw|ilbc), peer - audio=0x51d
> (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
> (ulaw|ilbc)
> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
> Looking for 18005551212 in default
> list_route: hop: <sip:104 at 192.168.254.16:5060>
> Transmitting (no NAT) to 192.168.254.16:5060:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c
> From: Sipura2000-2 <sip:104 at 192.168.254.6>;tag=ffbcad7047bef41co1
> To: <sip:18005551212 at 192.168.254.6>
> Call-ID: 11739700-af5297cc at 192.168.254.16
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: <sip:18005551212 at 192.168.254.4>
> Content-Length: 0
>                  
> 
>                  
> 
> ---
>     -- Executing Hangup("SIP/192.168.254.6-08a17090", "") in new stack
>   == Spawn extension (default, 18005551212, 1) exited non-zero on
> 'SIP/192.168.254.6-08a17090'
> Reliably Transmitting (no NAT) to 192.168.254.16:5060:
> SIP/2.0 403 Forbidden
> Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c
> From: Sipura2000-2 <sip:104 at 192.168.254.6>;tag=ffbcad7047bef41co1
> To: <sip:18005551212 at 192.168.254.6>;tag=as31d7c52b
> Call-ID: 11739700-af5297cc at 192.168.254.16
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: <sip:18005551212 at 192.168.254.4>
> Content-Length: 0
> 
> 
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