[Asterisk-Dev] Bug with toll free between 2 asterisk servers (special toll free signaling?)

Fri Sep 23 09:11:32 MST 2005

Thanks for the suggestion. This isn't the problem because the "2000"
extension was only setup for the purpose of documenting the problem for
email. The "real world" setup involves only valid 20 digit callerids.

I also don't think the toll-free system relies on caller id otherwise it
would be trivial to spoof (not that it would save you any money since
the call is free anyhow).

John Lange

On Mon, 2005-09-19 at 19:00 -0400, Joseph Benden wrote:
> This may or may not be a solution to your problem; however, I had 
> problems sending toll-free calls to my CSX, if the calling number wasn't 
> present.  This was due to certain toll-free numbers requiring the 
> calling number for billing purposes.  Are you sure the calling 
> information is making it (ie: 2000 isn't a valid phone number)?
> 
> -Joe
> 
> 
> John Lange wrote:
> 
> >I believe I have discovered a bug with Asterisk.
> >
> >Situation: When a call is placed to a toll free number from a sip phone
> >that passes through 2 Asterisk servers before being terminated it
> >doesn't work. Asterisk seems to ultimately send the wrong progress
> >information to the SIP phone.
> >
> >Network looks like:
> >
> >PRI <-> Cisco Switch <-sip-> Astrsk 2 <-sip-> Astrsk 1 <-sip-> Sip Phone
> >
> >This only happens with Toll Free calls! Long distance and other calls
> >work just fine.
> >
> >If you remove Asterisk 1 from the equation and connect the Sip Phone
> >directly to Asterisk 2, then toll free works great. This problem only
> >happens when the SIP has to pass through 2 Asterisk servers.
> >
> >Here is what you see in the consoles (Verbose = 3) (detailed sip debug
> >below):
> >
> >Asterisk 1 (closest to the phone) console:
> >-- Executing Dial("SIP/2000-a7e0", "SIP/18665270123 at openit") in new stack
> >-- Called 18665270123 at openit
> >-- SIP/openit-c42e is making progress passing it to SIP/2000-a7e0
> >
> >Asterisk 2 (closest to the PRI) console:
> >-- Executing Dial("SIP/s-6e98", "SIP/18665270123 at idyia_gw") in new stack
> >-- Called 18665270123 at idyia_gw
> >-- SIP/idyia_gw-cca3 is making progress passing it to SIP/s-6e98
> >
> >On the phone you just hear a fast busy signal.
> >
> >Please don't be fooled into thinking this is a NAT issue. This test was
> >done from behind NAT for the purpose of this email only. Normally this
> >is done from two Asterisk servers with real-world (non NAT) IP
> >addresses. Also, remember it works perfectly fine for all calls EXCEPT
> >toll free calls.
> >
> >Here are the configs on the two machines:
> >
> >---------
> ># Asterisk 1 (closest to the phone)
> >
> ># sip.conf
> >
> >register => user:X at sol.bighostbox.com
> >
> >[openit]
> >type=peer
> >secret=X
> >username=user
> >host=XXX.XXX.XXX.XXX
> >insecure=very
> >nat=no
> >context=openit
> >dtmfmode=rfc2833
> >canreinvite=no
> >qualify=yes
> >disallow=all
> >allow=ulaw
> >
> ># extensions.conf
> >exten => _18XXNXXXXXX,1,Dial(SIP/${EXTEN}@openit)
> >
> >----------
> >
> >
> >---------
> ># Asterisk 2 (closest to the PRI)
> >
> ># sip.conf
> >
> >[darkcore]
> >type=peer
> >secret=X
> >username=user
> >host=dynamic
> >insecure=very
> >nat=yes
> >context=darkcore_in
> >dtmfmode=rfc2833
> >canreinvite=no
> >qualify=yes
> >disallow=all
> >allow=ulaw
> >
> ># extensions.conf
> >exten => _18XXNXXXXXX,1,Dial(SIP/${EXTEN}@gw)
> >
> >----------
> >SIP DEBUG ASTERISK 2 (closest to the PRI)
> >----------
> >
> >Sip read:
> >INVITE sip:18665270123 at XXX.XXX.XXX.XXX SIP/2.0
> >Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK71e1c90a
> >From: "2000" <sip:2000 at 192.168.1.50>;tag=as04c5d8a7
> >To: <sip:18665270123 at XXX.XXX.XXX.XXX>
> >Contact: <sip:2000 at 192.168.1.50>
> >Call-ID: 7091774a7745747f71bb9a95648ce7b4 at 192.168.1.50
> >CSeq: 102 INVITE
> >User-Agent: Asterisk PBX
> >Date: Mon, 19 Sep 2005 21:08:21 GMT
> >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >Content-Type: application/sdp
> >Content-Length: 267
> >
> >v=0
> >o=root 29843 29843 IN IP4 192.168.1.50
> >s=session
> >c=IN IP4 192.168.1.50
> >t=0 0
> >m=audio 10980 RTP/AVP 0 8 2 101
> >a=rtpmap:0 PCMU/8000
> >a=rtpmap:8 PCMA/8000
> >a=rtpmap:2 G726-32/8000
> >a=rtpmap:101 telephone-event/8000
> >a=fmtp:101 0-16
> >a=silenceSupp:off - - - -
> >
> >12 headers, 12 lines
> >Using latest request as basis request
> >Found peer 'darkcore'
> >Found RTP audio format 0
> >Found RTP audio format 8
> >Found RTP audio format 2
> >Found RTP audio format 101
> >Peer audio RTP is at port 192.168.1.50:10980
> >Found description format PCMU
> >Found description format PCMA
> >Found description format G726-32
> >Found description format telephone-event
> >Capabilities: us - 0x1c (ulaw|alaw|g726), peer - audio=0x1c (ulaw|alaw|g726)/video=0x0 (nothing), combined - 0x1c (ulaw|alaw|g726)
> >Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
> >Looking for 18665270123 in darkcore_in
> >list_route: hop: <sip:2000 at 192.168.1.50>
> >Transmitting (NAT):
> >SIP/2.0 100 Trying
> >Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK71e1c90a;received=YYY.YYY.YYY.YYY;rport=1027
> >From: "2000" <sip:2000 at 192.168.1.50>;tag=as04c5d8a7
> >To: <sip:18665270123 at XXX.XXX.XXX.XXX>
> >Call-ID: 7091774a7745747f71bb9a95648ce7b4 at 192.168.1.50
> >CSeq: 102 INVITE
> >User-Agent: Asterisk PBX
> >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >Contact: <sip:18665270123 at XXX.XXX.XXX.XXX>
> >Content-Length: 0
> >
> >
> > to YYY.YYY.YYY.YYY:1027
> >    -- Executing Dial("SIP/s-66cf", "SIP/18665270123 at idyia_gw") in new stack
> >    -- Called 18665270123 at idyia_gw
> >    -- SIP/idyia_gw-1306 is making progress passing it to SIP/s-66cf
> >We're at XXX.XXX.XXX.XXX port 17630
> >Answering with preferred capability 0x4 (ulaw)
> >Answering with preferred capability 0x8 (alaw)
> >Answering with capability 0x10 (g726)
> >Answering with non-codec capability 0x1 (telephone-event)
> >Transmitting (NAT):
> >SIP/2.0 183 Session Progress
> >Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK71e1c90a;received=YYY.YYY.YYY.YYY;rport=1027
> >From: "2000" <sip:2000 at 192.168.1.50>;tag=as04c5d8a7
> >To: <sip:18665270123 at XXX.XXX.XXX.XXX>;tag=as7fe769dc
> >Call-ID: 7091774a7745747f71bb9a95648ce7b4 at 192.168.1.50
> >CSeq: 102 INVITE
> >User-Agent: Asterisk PBX
> >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >Contact: <sip:18665270123 at XXX.XXX.XXX.XXX>
> >Content-Type: application/sdp
> >Content-Length: 271
> >
> >v=0
> >o=root 17825 17825 IN IP4 XXX.XXX.XXX.XXX
> >s=session
> >c=IN IP4 XXX.XXX.XXX.XXX
> >t=0 0
> >m=audio 17630 RTP/AVP 0 8 2 101
> >a=rtpmap:0 PCMU/8000
> >a=rtpmap:8 PCMA/8000
> >a=rtpmap:2 G726-32/8000
> >a=rtpmap:101 telephone-event/8000
> >a=fmtp:101 0-16
> >a=silenceSupp:off - - - -
> >
> > to YYY.YYY.YYY.YYY:1027
> > 
> >----------
> >SIP DEBUG ASTERISK 1 (closest to the Phone)
> >----------
> >
> >    -- Executing Dial("SIP/2000-5afd", "SIP/18665270123 at openit") in new stack
> >We're at 192.168.1.50 port 14234
> >Answering/Requesting with root capability 0x4 (ulaw)
> >Answering with capability 0x8 (alaw)
> >Answering with capability 0x10 (g726)
> >Answering with non-codec capability 0x1 (telephone-event)
> >12 headers, 12 lines
> >Reliably Transmitting:
> >INVITE sip:18665270123 at XXX.XXX.XXX.XXX SIP/2.0
> >Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK3b250ec2
> >From: "2000" <sip:2000 at 192.168.1.50>;tag=as5f5675bb
> >To: <sip:18665270123 at XXX.XXX.XXX.XXX>
> >Contact: <sip:2000 at 192.168.1.50>
> >Call-ID: 4d41e9573b02ff97222113d10377154a at 192.168.1.50
> >CSeq: 102 INVITE
> >User-Agent: Asterisk PBX
> >Date: Mon, 19 Sep 2005 21:27:20 GMT
> >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >Content-Type: application/sdp
> >Content-Length: 267
> >
> >v=0
> >o=root 30653 30653 IN IP4 192.168.1.50
> >s=session
> >c=IN IP4 192.168.1.50
> >t=0 0
> >m=audio 14234 RTP/AVP 0 8 2 101
> >a=rtpmap:0 PCMU/8000
> >a=rtpmap:8 PCMA/8000
> >a=rtpmap:2 G726-32/8000
> >a=rtpmap:101 telephone-event/8000
> >a=fmtp:101 0-16
> >a=silenceSupp:off - - - -
> > (no NAT) to XXX.XXX.XXX.XXX:5060
> >    -- Called 18665270123 at openit
> >ws50*CLI>
> >
> >Sip read:
> >SIP/2.0 100 Trying
> >Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK3b250ec2;received=YYY.YYY.YYY.YYY;rport=1027
> >From: "2000" <sip:2000 at 192.168.1.50>;tag=as5f5675bb
> >To: <sip:18665270123 at XXX.XXX.XXX.XXX>
> >Call-ID: 4d41e9573b02ff97222113d10377154a at 192.168.1.50
> >CSeq: 102 INVITE
> >User-Agent: Asterisk PBX
> >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >Contact: <sip:18665270123 at XXX.XXX.XXX.XXX>
> >Content-Length: 0
> >
> >
> >10 headers, 0 lines
> >ws50*CLI>
> >
> >Sip read:
> >SIP/2.0 183 Session Progress
> >Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK3b250ec2;received=YYY.YYY.YYY.YYY;rport=1027
> >From: "2000" <sip:2000 at 192.168.1.50>;tag=as5f5675bb
> >To: <sip:18665270123 at XXX.XXX.XXX.XXX>;tag=as45f7543e
> >Call-ID: 4d41e9573b02ff97222113d10377154a at 192.168.1.50
> >CSeq: 102 INVITE
> >User-Agent: Asterisk PBX
> >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >Contact: <sip:18665270123 at XXX.XXX.XXX.XXX>
> >Content-Type: application/sdp
> >Content-Length: 271
> >
> >v=0
> >o=root 17825 17825 IN IP4 XXX.XXX.XXX.XXX
> >s=session
> >c=IN IP4 XXX.XXX.XXX.XXX
> >t=0 0
> >m=audio 18466 RTP/AVP 0 8 2 101
> >a=rtpmap:0 PCMU/8000
> >a=rtpmap:8 PCMA/8000
> >a=rtpmap:2 G726-32/8000
> >a=rtpmap:101 telephone-event/8000
> >a=fmtp:101 0-16
> >a=silenceSupp:off - - - -
> >
> >11 headers, 12 lines
> >Found RTP audio format 0
> >Found RTP audio format 8
> >Found RTP audio format 2
> >Found RTP audio format 101
> >Peer audio RTP is at port XXX.XXX.XXX.XXX:18466
> >Found description format PCMU
> >Found description format PCMA
> >Found description format G726-32
> >Found description format telephone-event
> >Capabilities: us - 0x1c (ulaw|alaw|g726), peer - audio=0x1c (ulaw|alaw|g726)/video=0x0 (nothing), combined - 0x1c (ulaw|alaw|g726)
> >Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
> >    -- SIP/openit-6444 is making progress passing it to SIP/2000-5afd
> >
> >
> >  
> >
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