[Asterisk-Dev] Bug with toll free between 2 asterisk servers
(special toll free signaling?)
Joseph Benden
joe at thrallingpenguin.com
Mon Sep 19 16:00:27 MST 2005
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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