Paul P. Pongco
2005-Mar-14 02:40 UTC
[Asterisk-Users] weird outbound problem through broadvoice (new)
Hello,
Have a weird problem when using asterisk (1.0.6). There are certain
numbers I cannot dial when using asterisk with my broadvoice account.
No problems with inbound. With outbound calls, I can call some numbers
(for example broadvoice customer support number) and unsuccessfully with
some. However, when I configure my account directly on x-lite, I dont
see these outbound problems.
Here is a snapshot of my sip.conf
register =>
UUUUUUUUUU@sip.broadvoice.com:PPPPPPPPPP:UUUUUUUUUU@sip.broadvoice.com
[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromuser=UUUUUUUUUU
fromdomain=sip.broadvoice.com
secret=PPPPPPPPPP
username=UUUUUUUUUU
port=5060
dtmfmode=inband
dtmf=inband
insecure=very
context=incoming
authname=UUUUUUUUUU
canreinvite=no
qualify=no
nat=no
extensions.conf
[outgoing]
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()
A portion of sip debug during successful calls (calling broadvoice
support)
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a
From: "1001" <sip:UUUUUUUUUU@sip.broadvoice.com>;tag=as65b65920
To: <sip:19784187300@sip.broadvoice.com>
Call-ID: 2007fca97e36e72b54818caa377e6dcc@sip.broadvoice.com
CSeq: 103 INVITE
6 headers, 0 lines
CLI>
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a
From: "1001" <sip:UUUUUUUUUU@sip.broadvoice.com>;tag=as65b65920
To:
<sip:19784187300@sip.broadvoice.com>;tag=SD58a8499-104694000-1110784950009
Call-ID: 2007fca97e36e72b54818caa377e6dcc@sip.broadvoice.com
CSeq: 103 INVITE
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported: 100rel,timer
Contact:
<sip:19784187300@147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp>
Remote-Party-ID: "Auto Attendant
PrimaryAttendant"<sip:9784187395@147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;screen=yes;party=called;privacy=off;id-type=subscriber
Content-Length: 0
A portion of sip debug during unsuccessful calls, where TTTTTTTTT is the
target phone number
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18
From: "1001" <sip:UUUUUUUUUU@sip.broadvoice.com>;tag=as6f6dba69
To: <sip:1TTTTTTTTTT@sip.broadvoice.com>
Call-ID: 095981b26d97329e4155ccd529617e5c@sip.broadvoice.com
CSeq: 103 INVITE
6 headers, 0 lines
Reliably Transmitting:
CANCEL sip:1TTTTTTTTTT@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18
From: "1001" <sip:UUUUUUUUUU@sip.broadvoice.com>;tag=as6f6dba69
To: <sip:1TTTTTTTTTT@sip.broadvoice.com>
Contact: <sip:UUUUUUUUUU@x.x.x.x>
Call-ID: 095981b26d97329e4155ccd529617e5c@sip.broadvoice.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="UUUUUUUUUU",
realm="BroadWorks",
algorithm=MD5,
uri="sip:1TTTTTTTTTT@sip.broadvoice.com",
nonce="1110785211206",
response="f68a31735aec843b9ef68b7909fcf178", opaque=""
Content-Length: 0
(no NAT) to 147.135.8.128:5060
Scheduling destruction of call
'095981b26d97329e4155ccd529617e5c@sip.broadvoice.com' in 15000 ms
Transmitting (no NAT):
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK01853115f3033a3c
From: <sip:1001@x.x.x.x>;tag=9d9e03fd7b4508e9
To: <sip:1TTTTTTTTTT@x.x.x.x>;tag=as79fd7936
Call-ID: 3512f0bb5f5ebf20@x.x.x.x
CSeq: 7327 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1TTTTTTTTT@x.x.x.x>
Content-Length: 0
to x.x.x.x:5060
Asterisk box not behind firewall. No iptables filters either. It seems
that asterisk is sending CANCEL due to call timeout after the 2nd 100
Trying during INVITE message flow. I am not sure what is causing the
timeout. Anyone experienced this before? Tried using ethereal to debug
the problem deeply, but I can only see the same flow as the sip debug.
Hoping for your assistance. Thanks.
Paul P. Pongco
2005-Mar-14 21:48 UTC
[Asterisk-Users] weird outbound problem through broadvoice (new)
Hello, I changed my asterisk to the recently posted software on CVS (Asterisk CVS-v1-0-03/15/05-12:11:02). Problem still persists. What is weird here is I can dial certain numbers (broadvoice support number works) but cant on others. Checked the SIP call flow via ethereal and I can see Im sending and receiving invites from the same broadvoice server (147.135.8.128) w/c is what I have mapped sip.broadvoice.com to at /etc/hosts. Any other way I can debug this? Thanks. On Mon, 2005-03-14 at 17:40, Paul P. Pongco wrote:> Hello, > > Have a weird problem when using asterisk (1.0.6). There are certain > numbers I cannot dial when using asterisk with my broadvoice account. > No problems with inbound. With outbound calls, I can call some numbers > (for example broadvoice customer support number) and unsuccessfully with > some. However, when I configure my account directly on x-lite, I dont > see these outbound problems. > Here is a snapshot of my sip.conf > > register => UUUUUUUUUU@sip.broadvoice.com:PPPPPPPPPP:UUUUUUUUUU@sip.broadvoice.com > > > [sip.broadvoice.com] > type=peer > host=sip.broadvoice.com > fromuser=UUUUUUUUUU > fromdomain=sip.broadvoice.com > secret=PPPPPPPPPP > username=UUUUUUUUUU > port=5060 > dtmfmode=inband > dtmf=inband > insecure=very > context=incoming > authname=UUUUUUUUUU > canreinvite=no > qualify=no > nat=no > > extensions.conf > [outgoing] > exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) > exten => _1NXXNXXXXXX, 2, congestion() > exten => _1NXXNXXXXXX, 102, busy() > > A portion of sip debug during successful calls (calling broadvoice > support) > > Sip read: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a > From: "1001" <sip:UUUUUUUUUU@sip.broadvoice.com>;tag=as65b65920 > To: <sip:19784187300@sip.broadvoice.com> > Call-ID: 2007fca97e36e72b54818caa377e6dcc@sip.broadvoice.com > CSeq: 103 INVITE > > 6 headers, 0 lines > CLI> > > Sip read: > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a > From: "1001" <sip:UUUUUUUUUU@sip.broadvoice.com>;tag=as65b65920 > To: > <sip:19784187300@sip.broadvoice.com>;tag=SD58a8499-104694000-1110784950009 > Call-ID: 2007fca97e36e72b54818caa377e6dcc@sip.broadvoice.com > CSeq: 103 INVITE > Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY > Supported: 100rel,timer > Contact: > <sip:19784187300@147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp> > Remote-Party-ID: "Auto Attendant > PrimaryAttendant"<sip:9784187395@147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;screen=yes;party=called;privacy=off;id-type=subscriber > Content-Length: 0 > > A portion of sip debug during unsuccessful calls, where TTTTTTTTT is the > target phone number > > Sip read: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18 > From: "1001" <sip:UUUUUUUUUU@sip.broadvoice.com>;tag=as6f6dba69 > To: <sip:1TTTTTTTTTT@sip.broadvoice.com> > Call-ID: 095981b26d97329e4155ccd529617e5c@sip.broadvoice.com > CSeq: 103 INVITE > > > 6 headers, 0 lines > Reliably Transmitting: > CANCEL sip:1TTTTTTTTTT@sip.broadvoice.com SIP/2.0 > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18 > From: "1001" <sip:UUUUUUUUUU@sip.broadvoice.com>;tag=as6f6dba69 > To: <sip:1TTTTTTTTTT@sip.broadvoice.com> > Contact: <sip:UUUUUUUUUU@x.x.x.x> > Call-ID: 095981b26d97329e4155ccd529617e5c@sip.broadvoice.com > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Proxy-Authorization: Digest username="UUUUUUUUUU", realm="BroadWorks", > algorithm=MD5, > uri="sip:1TTTTTTTTTT@sip.broadvoice.com", nonce="1110785211206", > response="f68a31735aec843b9ef68b7909fcf178", opaque="" > Content-Length: 0 > > (no NAT) to 147.135.8.128:5060 > Scheduling destruction of call > '095981b26d97329e4155ccd529617e5c@sip.broadvoice.com' in 15000 ms > Transmitting (no NAT): > SIP/2.0 503 Service Unavailable > Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK01853115f3033a3c > From: <sip:1001@x.x.x.x>;tag=9d9e03fd7b4508e9 > To: <sip:1TTTTTTTTTT@x.x.x.x>;tag=as79fd7936 > Call-ID: 3512f0bb5f5ebf20@x.x.x.x > CSeq: 7327 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:1TTTTTTTTT@x.x.x.x> > Content-Length: 0 > > to x.x.x.x:5060 > > Asterisk box not behind firewall. No iptables filters either. It seems > that asterisk is sending CANCEL due to call timeout after the 2nd 100 > Trying during INVITE message flow. I am not sure what is causing the > timeout. Anyone experienced this before? Tried using ethereal to debug > the problem deeply, but I can only see the same flow as the sip debug. > Hoping for your assistance. Thanks. > > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Cheers, Paul P. Pongco Mosaic Communications Inc.