[Asterisk-Users] BROADVOICE fails to register
Bruce Komito
brucek at bagel.com
Thu Nov 4 07:02:29 MST 2004
Try two things (one at a time, to see which one helps you) in sip.conf:
1. After the IP address on the register statement, append "/954XXXXXX"
2. In the [Broadvoice] section, change NAT to no.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Thu, 4 Nov 2004, Francisco Perez-Landaeta wrote:
> I have the following registration information for broadvoice, but it fails
> to register. Can anyone tell me what is the problem ? I am not behind a
> firewall and I have added this below the [general]
>
> Thanks,
>
> My number and password is hidded. I received from broadvoice the following :
>
> primary_dns_ip: 147.135.0.6
> secondary_dns_ip: 147.138.8.6
> proxy_ip: proxy.broadvoice.com
> proxy_port: 5060
> registrar_ip: sip.broadvoice.com
> registrar_port: 5060
> phone_number: 954XXXXXXX
> auth_id: 954XXXXXXX
> auth_password: password
> tftp_ip: config.broadvoice.com
> ntp_ip: ntp.nasa.gov
> DTMF: InBand
> registration_time_out: 10_seconds
> voice_mail_key: *86
>
>
> BELOW IS WHAT I TOOK FROM THE WIKI, EVEN IF I CHANGE THE IP ADDRESSES IT
> FAILS TO REGISTER.
>
> sip.conf
> Under the general context:
> ;externip=<IP or Dynamic DNS Name> ; Needed if behind NAT
> context=incoming
> dtmfmode=inband
> register => 954XXXXXXX:password at 147.135.0.129
>
> [Broadvoice]
> type=peer
> username=954XXXXXXX
> fromuser=954XXXXXXX
> secret=password
> host=147.135.0.129
> context=incoming
> fromdomain=sip.broadvoice.com
> nat=yes
> canreinvite=no
> dtmfmode=inband
>
> extensions.conf For incoming calls
> [incoming]
> ;This extension line will ring SIP
> ;extension 2001 for 60 seconds then hang up. Modify as necessary to fit your
> dialplan
> exten => s,1,Dial(SIP/2001,60,tr)
> exten => s,2,hangup
>
> extensions.conf For outgoing calls:
> [toll-trunks]
> ;Pattern match for local call plan, use appropriate pattern if on nationwide
> plan.
> exten => _9NXXXXXX,1,Dial(SIP/${EXTEN:1}@broadvoice)
> exten => _9NXXXXXX,2,Congestion
>
>
> ;
> ----- Original Message -----
> From: <asterisk-users-request at lists.digium.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Wednesday, November 03, 2004 7:01 PM
> Subject: Asterisk-Users Digest, Vol 4, Issue 52
>
>
> > Send Asterisk-Users mailing list submissions to
> > asterisk-users at lists.digium.com
> >
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> > or, via email, send a message with subject or body 'help' to
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> >
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> >
> > When replying, please edit your Subject line so it is more specific
> > than "Re: Contents of Asterisk-Users digest..."
> >
> >
> > Today's Topics:
> >
> > 1. Re: ISDN Dialplan (Paulo Adriano)
> > 2. Re: Sip clients not longer registering (David Filion)
> > 3. RE: Installing X100P Asterisk - Unable to createchannel of
> > type 'Zap' (Vikas Deolaliker)
> > 4. MusicOnhold on Bridged calls (Paulo Adriano)
> > 5. Re: Automatically restart asterisk if not running
> > (Brancaleoni Matteo)
> > 6. What do I need to ask my T1 supplier? (Scott Nelson)
> > 7. getting cid from spa3k pstn to * (Randy Bush)
> > 8. How change default law for T100P (Manuel Marin)
> > 9. Re: What do I need to ask my T1 supplier? (niles at atheos.net)
> > 10. Re: What do I need to ask my T1 supplier? (TC)
> > 11. Cisco 79XX - Using built-in 3way conference (Matthew Boehm)
> >
> >
> > ----------------------------------------------------------------------
> >
> > Message: 1
> > Date: Wed, 03 Nov 2004 22:06:10 +0000
> > From: "Paulo Adriano" <pauloadriano at wavelis.pt>
> > Subject: Re: [Asterisk-Users] ISDN Dialplan
> > To: <asterisk-users at lists.digium.com>, <psvasterisk at psv.nu>
> > Cc: mje at posix.co.za
> > Message-ID: <s1895669.024 at wavelis>
> > Content-Type: text/plain; charset="us-ascii"
> >
> > It*s solved now. You are right it was a matter of formating the
> > outgoing number.
> >
> > Thanks
> >
> > Paulo
> >
> > >>>psvasterisk at psv.nu 11/03 1:34 pm >>>
> >
> > On Wed, 3 Nov 2004, Paulo Adriano wrote:
> >
> >
> > >After trying that syntax I still have the same problem, one thing is
> >
> > >very strange is the number that Asterisk reports as the incoming. My
> >
> > >ESN*s numbers are 219898334 and 219898335 but on the console I see
> >
> > >219898334,1 and 219898335,1
> >
> >
> > Are you sure it is not trying to tell you extension 219898334 and
> > priority
> >
> > 1 in the dialplan?
> >
> >
> > [snip}
> >
> >
> > >THIS IS AN OUTGOING CALL TRY TO NUMBER 213570150 MY DIALPLAN
> > REQUIRES 9 to go outside
> >
> > >
> >
> > >*CLI> -- Executing Dial(SIP/21-9da8, Modem/g1/9213570150||tr) in
> > new stack
> >
> > >Nov 3 18:37:35 WARNING[1110502320]: chan_modem.c:191 modem_call:
> > Destination g1/9213570150 requres a real destination
> > (device:destination)
> >
> > > -- Couldn't call g1/9213570150
> >
> > > -- Hungup 'Modem[i4l]/ttyI1'
> >
> > > == Everyone is busy/congested at this time
> >
> > >Nov 3 18:37:45 WARNING[1110502320]: pbx.c:1933 ast_pbx_run: Timeout,
> > but no rule 't' in context 'local-access'
> >
> > > -- Saved useragent SJLabs-SJphone/1.30.248 for peer 21
> >
> >
> > The source for chan_modem suggests that the dial string should be
> >
> > formatted like
> >
> > Dial(Modem/g1:9213570150||tr)
> >
> >
> > Peter
> >
> >
> >
> >
> >
> > Asterisk-Users mailing list
> >
> > Asterisk-Users at 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
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> >
> > ------------------------------
> >
> > Message: 2
> > Date: Wed, 03 Nov 2004 17:11:39 -0500
> > From: David Filion <dfilion at dotality.com>
> > Subject: Re: [Asterisk-Users] Sip clients not longer registering
> > To: asterisk-users at lists.digium.com
> > Message-ID: <4189579B.4030804 at dotality.com>
> > Content-Type: text/plain; charset=us-ascii; format=flowed
> >
> >
> >
> > Message: 8
> > Date: Wed, 03 Nov 2004 22:32:46 +0100
> > From: "Olle E. Johansson" <oej at edvina.net>
> > Subject: Re: [Asterisk-Users] Sip clients not longer registering
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > <asterisk-users at lists.digium.com>
> > Message-ID: <41894E7E.1060006 at edvina.net>
> > Content-Type: text/plain; charset=us-ascii; format=flowed
> >
> > David Filion wrote:
> >
> >
> >
> > >> Hi,
> > >>
> > >> We have been using Asterisk since version 0.9x with little or no
> > >> problems. However, for an unknow reasons, our sip clients can nolonger
> > >> register. We updated to Asterisk 1.0.2 hoping that would solve the
> > >> problem, but no luck.
> > >>
> > >
> > >
> > There is a chance that our change NAT logic is problematic in your
> network.
> > (rport support)
> >
> > check the sample sip.conf for various options to nat= and try them.
> > It's in configs/sip.conf.sample
> >
> > /O
> >
> >
> >
> > Thanks, I'll give them a try.
> >
> > David
> >
> >
> >
> >
> > ------------------------------
> >
> > Message: 3
> > Date: Wed, 3 Nov 2004 14:15:49 -0800
> > From: "Vikas Deolaliker" <vikasd at yahoo.com>
> > Subject: RE: [Asterisk-Users] Installing X100P Asterisk - Unable to
> > createchannel of type 'Zap'
> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> > <asterisk-users at lists.digium.com>
> > Message-ID: <20041103221544.2A0F82FE4ED at lists.digium.com>
> > Content-Type: text/plain; charset="us-ascii"
> >
> >
> > Look at your logs in /var/logs/asterisk/. I am pretty certain it is a
> fault
> > in your /etc/asterisk/Zapata.conf file.
> >
> > Vikas
> >
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Seth
> Remington
> > Sent: Wednesday, November 03, 2004 1:01 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Installing X100P Asterisk - Unable to
> > createchannel of type 'Zap'
> >
> > On Wed, 2004-11-03 at 13:02, Frank Kostin wrote:
> > > Hello list,
> > > I am trying to install a Digium X100P into a Redhat Asterisk.
> > > Kernel seems to be OK, card OK.
> > > Zaptel Configuration seems to be OK.
> > > # ztcfg -vv
> > > Channel map:
> > > Channel 01: FXS Kewlstart (Default) (Slaves: 01)
> > > 1 channels configured.
> > >
> > > Asterisk works fine with IP SIP but not with X100P
> > > I get the error on Asterisk CLI>
> > > .... channel.c:1919 ast_request: "No channel type registered for
> > > 'Zap'"
> > > and than ....app_dial.c:763 dial_exec "Unable to create channel of
> > > type 'Zap'".
> > >
> > > Does anyone know what might be the problem ?
> > > Thanks for any help
> >
> > Did you:
> >
> > modprobe zaptel
> > modprobe wcfxo
> >
> > Also make sure that you compiled and installed zaptel *before* you
> > installed asterisk. If you did it afterwords simply re-compile asterisk
> > and you should be good.
> >
> > -Seth
> >
> > --
> > Seth Remington
> > SaberLogic, LLC
> > 661-B Weber Drive
> > Wadsworth, Ohio 44281
> > Phone: (330)335-6442
> > Fax: (330)336-8559
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at 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
> >
> >
> >
> >
> > ------------------------------
> >
> > Message: 4
> > Date: Wed, 03 Nov 2004 22:27:23 +0000
> > From: "Paulo Adriano" <pauloadriano at wavelis.pt>
> > Subject: [Asterisk-Users] MusicOnhold on Bridged calls
> > To: <asterisk-users at lists.digium.com>
> > Message-ID: <s1895b59.027 at wavelis>
> > Content-Type: text/plain; charset="us-ascii"
> >
> > Now that my bridged calls are working fine with ISDN I have a question ?
> >
> >
> > When my customers call in and my ext is not available the call is routed
> > out to my mobile.
> >
> > Everything works but I would like to know if there is a way of having
> > the calling sign (tone) always on . With my current config the origin
> > caller phone hear a calling beeping tone only until the call is passed
> > on a bridged call. Then and until the end user (in this case , myself in
> > the cellular phone) answers the call , the origin has a big silence.
> > This may cause him to think that something append during the call.
> >
> > How can I configure * in order to avoid the calling tone from
> > desapearing on a bridged call. Even music would be acceptable as the
> > call is being tranfered to a pstn number.
> >
> > Thanks in advance
> > Paulo
> >
> > Francisco Paulo Adriano
> > WaveLIS LDA
> > Mobile +351 91 870 87 98
> > Office + 351 21 989 83 34
> > Fax +351 21 989 83 35
> > E-mail : pauloadriano at wavelis.pt
> >
> >
> >
> >
> >
> > -------------- next part --------------
> > An HTML attachment was scrubbed...
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> >
> > ------------------------------
> >
> > Message: 5
> > Date: Wed, 03 Nov 2004 23:28:04 +0100
> > From: Brancaleoni Matteo <mbrancaleoni at espia.it>
> > Subject: Re: [Asterisk-Users] Automatically restart asterisk if not
> > running
> > To: Matthew Marlowe <matthew.marlowe at gmail.com>, Asterisk Users
> > Mailing List - Non-Commercial Discussion
> > <asterisk-users at lists.digium.com>
> > Cc: asterisk-users at lists.digium.com
> > Message-ID: <1099520884.11249.2.camel at athlon64>
> > Content-Type: text/plain
> >
> > Hi
> > Il mer, 2004-11-03 alle 21:41, Matthew Marlowe ha scritto:
> > > I once found a script, I think it was on the mailing list that would
> > > run in crontab and restart asterisk if not running... Does anyone
> > > happen to have a copy of that?
> >
> > I suggest you to use something like a superdaemon,
> > ie a process (normally spawned by init) that
> > checks other processes.
> > I suggest you monit (http://www.tildeslash.com/monit/),
> > it can monitor processes in various ways and other
> > vital system informations. I'm pretty happy
> > with it (using it in all my * installations)
> >
> > Matteo
> > --
> > Brancaleoni Matteo <mbrancaleoni at espia.it>
> > Espia Srl
> >
> >
> >
> > ------------------------------
> >
> > Message: 6
> > Date: Wed, 3 Nov 2004 17:32:27 -0500
> > From: Scott Nelson <asterisk at nelson.saint-louis.mo.us>
> > Subject: [Asterisk-Users] What do I need to ask my T1 supplier?
> > To: asterisk-users at lists.digium.com
> > Message-ID: <200411031632.27499.asterisk at nelson.saint-louis.mo.us>
> > Content-Type: text/plain; charset="us-ascii"
> >
> > My employer is switching to a new T1 supplier (it was AT&T, we are now
> going
> > with XO), and sometime in the future we want to replace our PBX with an
> > Asterisk system.
> >
> > What do I need to know to make sure the T1 line is "provisioned" (is that
> the
> > right term?) correctly for a Digium T100P/TE410P/TE405P?
> >
> > They will split the T1 line into 10 channels of voice and 14 channels of
> data.
> > >From what I understand, they will terminate the T1 into a channel bank,
> and
> > then from that give is 10 POTS phone jacks and one data port (to go to an
> > Adtran router for our Internet access).
> >
> > Any comments and/or suggestions?
> >
> > Scott
> >
> >
> > ------------------------------
> >
> > Message: 7
> > Date: Wed, 3 Nov 2004 14:40:29 -0800
> > From: Randy Bush <randy at psg.com>
> > Subject: [Asterisk-Users] getting cid from spa3k pstn to *
> > To: splatters <asterisk-users at lists.digium.com>
> > Message-ID: <16777.24157.228132.850877 at ran.psg.com>
> > Content-Type: text/plain; charset=us-ascii
> >
> > i am still going crazy with this one. i can not get callerid from a call
> > received on the spa3k pstn to asterisk. THIS USED TO WORK!
> >
> > in order to get the cid from the spa3k to *, i need to turn on
> > PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES
> >
> > the sip.conf entry looks like
> > [spa3k]
> > type=friend
> > host=dynamic
> > port=5061
> > auth=md5
> > secret=hidden
> > qualify=1000
> > dtmfmode=rfc2833
> > canreinvite=yes
> > context=spa3k-ext
> >
> > this produces a sip exchange as follows:
> >
> > No. Time Source Destination
> Protocol Info
> > 1 0.000000 spa3k asterisk.foo.edu SIP/SDP Request: INVITE
> sip:105 at asterisk.foo.edu, with session description
> >
> > Frame 1 (1095 bytes on wire, 1095 bytes captured)
> > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
> > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu
> (asterisk-ip-ad)
> > User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
> > Session Initiation Protocol
> > Request-Line: INVITE sip:105 at asterisk.foo.edu SIP/2.0
> > Message Header
> > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-f9456447
> > From: CallerName
> <sip:2065551212 at asterisk.foo.edu>;tag=54e649b356424567o1
> > SIP Display info: CallerName
> > SIP from address: sip:2065551212 at asterisk.foo.edu
> > SIP tag: 54e649b356424567o1
> > To: <sip:105 at asterisk.foo.edu>
> > SIP to address: sip:105 at asterisk.foo.edu
> > Remote-Party-ID: CallerName
> <sip:2065551212 at asterisk.foo.edu>;screen=yes;party=calling
> > Call-ID: 51efe8a3-2d73b337 at spa3k
> > CSeq: 101 INVITE
> > Max-Forwards: 70
> > Contact: biwa 0431 <sip:biwaa1-in at spa3k:5061>
> > Expires: 240
> > User-Agent: Sipura/SPA3000-2.0.11(GWa)
> > Content-Length: 430
> > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> > Supported: x-sipura
> > Content-Type: application/sdp
> > Message body
> > Session Description Protocol
> >
> > No. Time Source Destination
> Protocol Info
> > 2 0.000514 asterisk.foo.edu spa3k SIP Status: 407 Proxy
> Authentication Required
> >
> > Frame 2 (520 bytes on wire, 520 bytes captured)
> > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a
> > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), Dst
> Addr: spa3k (spa3k)
> > User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5061 (5061)
> > Session Initiation Protocol
> > Status-Line: SIP/2.0 407 Proxy Authentication Required
> > Message Header
> > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-f9456447
> > From: CallerName
> <sip:2065551212 at asterisk.foo.edu>;tag=54e649b356424567o1
> > SIP Display info: CallerName
> > SIP from address: sip:2065551212 at asterisk.foo.edu
> > SIP tag: 54e649b356424567o1
> > To: <sip:105 at asterisk.foo.edu>;tag=as741941ff
> > SIP to address: sip:105 at asterisk.foo.edu
> > SIP tag: as741941ff
> > Call-ID: 51efe8a3-2d73b337 at spa3k
> > CSeq: 101 INVITE
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > Contact: <sip:105 at asterisk-ip-ad>
> > Proxy-Authenticate: Digest realm="asterisk", nonce="263c07e5"
> > Content-Length: 0
> >
> > No. Time Source Destination
> Protocol Info
> > 3 0.090441 spa3k asterisk.foo.edu SIP Request: ACK
> sip:105 at asterisk.foo.edu
> >
> > Frame 3 (453 bytes on wire, 453 bytes captured)
> > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
> > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu
> (asterisk-ip-ad)
> > User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
> > Session Initiation Protocol
> > Request-Line: ACK sip:105 at asterisk.foo.edu SIP/2.0
> > Message Header
> > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-f9456447
> > From: CallerName
> <sip:2065551212 at asterisk.foo.edu>;tag=54e649b356424567o1
> > SIP Display info: CallerName
> > SIP from address: sip:2065551212 at asterisk.foo.edu
> > SIP tag: 54e649b356424567o1
> > To: <sip:105 at asterisk.foo.edu>;tag=as741941ff
> > SIP to address: sip:105 at asterisk.foo.edu
> > SIP tag: as741941ff
> > Call-ID: 51efe8a3-2d73b337 at spa3k
> > CSeq: 101 ACK
> > Max-Forwards: 70
> > Contact: biwa 0431 <sip:biwaa1-in at spa3k:5061>
> > User-Agent: Sipura/SPA3000-2.0.11(GWa)
> > Content-Length: 0
> >
> > No. Time Source Destination
> Protocol Info
> > 4 0.135913 spa3k asterisk.foo.edu SIP/SDP Request: INVITE
> sip:105 at asterisk.foo.edu, with session description
> >
> > Frame 4 (1265 bytes on wire, 1265 bytes captured)
> > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
> > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu
> (asterisk-ip-ad)
> > User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
> > Session Initiation Protocol
> > Request-Line: INVITE sip:105 at asterisk.foo.edu SIP/2.0
> > Message Header
> > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-e2744867
> > From: CallerName
> <sip:2065551212 at asterisk.foo.edu>;tag=54e649b356424567o1
> > SIP Display info: CallerName
> > SIP from address: sip:2065551212 at asterisk.foo.edu
> > SIP tag: 54e649b356424567o1
> > To: <sip:105 at asterisk.foo.edu>
> > SIP to address: sip:105 at asterisk.foo.edu
> > Remote-Party-ID: CallerName
> <sip:2065551212 at asterisk.foo.edu>;screen=yes;party=calling
> > Call-ID: 51efe8a3-2d73b337 at spa3k
> > CSeq: 102 INVITE
> > Max-Forwards: 70
> > Proxy-Authorization: Digest
> username="biwaa1-in",realm="asterisk",nonce="263c07e5",uri="sip:105 at asterisk
> foo.edu",algorithm=MD5,response="f8e02292686b3b5cb2117186b1474ba9"
> > Contact: biwa 0431 <sip:biwaa1-in at spa3k:5061>
> > Expires: 240
> > User-Agent: Sipura/SPA3000-2.0.11(GWa)
> > Content-Length: 430
> > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> > Supported: x-sipura
> > Content-Type: application/sdp
> > Message body
> > Session Description Protocol
> >
> > No. Time Source Destination
> Protocol Info
> > 5 0.136261 asterisk.foo.edu spa3k SIP Status: 403
> Forbidden
> >
> > Frame 5 (437 bytes on wire, 437 bytes captured)
> > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a
> > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), Dst
> Addr: spa3k (spa3k)
> > User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5061 (5061)
> > Session Initiation Protocol
> > Status-Line: SIP/2.0 403 Forbidden
> > Message Header
> > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-e2744867
> > From: CallerName
> <sip:2065551212 at asterisk.foo.edu>;tag=54e649b356424567o1
> > SIP Display info: CallerName
> > SIP from address: sip:2065551212 at asterisk.foo.edu
> > SIP tag: 54e649b356424567o1
> > To: <sip:105 at asterisk.foo.edu>;tag=as741941ff
> > SIP to address: sip:105 at asterisk.foo.edu
> > SIP tag: as741941ff
> > Call-ID: 51efe8a3-2d73b337 at spa3k
> > CSeq: 102 INVITE
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > Contact: <sip:105 at asterisk-ip-ad>
> > Content-Length: 0
> >
> > No. Time Source Destination
> Protocol Info
> > 6 0.383761 spa3k asterisk.foo.edu SIP Request: ACK
> sip:105 at asterisk.foo.edu
> >
> > Frame 6 (623 bytes on wire, 623 bytes captured)
> > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
> > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu
> (asterisk-ip-ad)
> > User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
> > Session Initiation Protocol
> > Request-Line: ACK sip:105 at asterisk.foo.edu SIP/2.0
> > Message Header
> > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-e2744867
> > From: CallerName
> <sip:2065551212 at asterisk.foo.edu>;tag=54e649b356424567o1
> > SIP Display info: CallerName
> > SIP from address: sip:2065551212 at asterisk.foo.edu
> > SIP tag: 54e649b356424567o1
> > To: <sip:105 at asterisk.foo.edu>;tag=as741941ff
> > SIP to address: sip:105 at asterisk.foo.edu
> > SIP tag: as741941ff
> > Call-ID: 51efe8a3-2d73b337 at spa3k
> > CSeq: 102 ACK
> > Max-Forwards: 70
> > Proxy-Authorization: Digest
> username="biwaa1-in",realm="asterisk",nonce="263c07e5",uri="sip:105 at asterisk
> foo.edu",algorithm=MD5,response="c33e3a4bab8eef38ca12b9ddf192b796"
> > Contact: biwa 0431 <sip:biwaa1-in at spa3k:5061>
> > User-Agent: Sipura/SPA3000-2.0.11(GWa)
> > Content-Length: 0
> >
> > No. Time Source Destination
> Protocol Info
> > 7 7.079655 asterisk.foo.edu spa3k SIP Request: OPTIONS
> sip:spa3k
> >
> > Frame 7 (463 bytes on wire, 463 bytes captured)
> > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a
> > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), Dst
> Addr: spa3k (spa3k)
> > User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
> > Session Initiation Protocol
> > Request-Line: OPTIONS sip:spa3k SIP/2.0
> > Message Header
> > Via: SIP/2.0/UDP asterisk-ip-ad:5060;branch=z9hG4bK1b040c15
> > From: "Unknown" <sip:Unknown at asterisk-ip-ad>;tag=as3f547347
> > SIP Display info: "Unknown"
> > SIP from address: sip:Unknown at asterisk-ip-ad
> > SIP tag: as3f547347
> > To: <sip:spa3k>
> > SIP to address: sip:spa3k
> > Contact: <sip:Unknown at asterisk-ip-ad>
> > Call-ID: 159ec16b69ac62e334905b487158eeed at asterisk-ip-ad
> > CSeq: 102 OPTIONS
> > User-Agent: Asterisk PBX
> > Date: Mon, 01 Nov 2004 17:34:49 GMT
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > Content-Length: 0
> >
> > No. Time Source Destination
> Protocol Info
> > 8 7.079766 asterisk.foo.edu spa3k SIP Request: OPTIONS
> sip:spa3k:5061
> >
> > Frame 8 (473 bytes on wire, 473 bytes captured)
> > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a
> > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), Dst
> Addr: spa3k (spa3k)
> > User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5061 (5061)
> > Session Initiation Protocol
> > Request-Line: OPTIONS sip:spa3k:5061 SIP/2.0
> > Message Header
> > Via: SIP/2.0/UDP asterisk-ip-ad:5060;branch=z9hG4bK29909a71
> > From: "Unknown" <sip:Unknown at asterisk-ip-ad>;tag=as67500153
> > SIP Display info: "Unknown"
> > SIP from address: sip:Unknown at asterisk-ip-ad
> > SIP tag: as67500153
> > To: <sip:spa3k:5061>
> > SIP to address: sip:spa3k:5061
> > Contact: <sip:Unknown at asterisk-ip-ad>
> > Call-ID: 2b80a2980a32bf7809b8648328ced971 at asterisk-ip-ad
> > CSeq: 102 OPTIONS
> > User-Agent: Asterisk PBX
> > Date: Mon, 01 Nov 2004 17:34:49 GMT
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > Content-Length: 0
> >
> > No. Time Source Destination Protoc
> ol Info
> > 9 7.173099 spa3k asterisk.foo.edu SIP Status: 404 Not
> Found
> >
> > Frame 9 (361 bytes on wire, 361 bytes captured)
> > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
> > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu
> (asterisk-ip-ad)
> > User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
> > Session Initiation Protocol
> > Status-Line: SIP/2.0 404 Not Found
> > Message Header
> > To: <sip:spa3k>;tag=828c8dcf8cd9e760i0
> > SIP to address: sip:spa3k
> > SIP tag: 828c8dcf8cd9e760i0
> > From: "Unknown" <sip:Unknown at asterisk-ip-ad>;tag=as3f547347
> > SIP Display info: "Unknown"
> > SIP from address: sip:Unknown at asterisk-ip-ad
> > SIP tag: as3f547347
> > Call-ID: 159ec16b69ac62e334905b487158eeed at asterisk-ip-ad
> > CSeq: 102 OPTIONS
> > Via: SIP/2.0/UDP asterisk-ip-ad:5060;branch=z9hG4bK1b040c15
> > Server: Sipura/SPA3000-2.0.11(GWa)
> > Content-Length: 0
> >
> > note that the From: has the cid, as does the Remote-Party-ID:. and the
> > Contact: has the spa3k's id and display name. and
> > the From: and/or Remote-Party-ID: cause asterisk respond with 407 Proxy
> > Authentication Required, and things do not improve from there
> >
> > if i set the spa3k config to have
> > PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = NO
> >
> > Frame 1 (1072 bytes on wire, 1072 bytes captured)
> > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
> > Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr:
> 666.42.7.11 (666.42.7.11)
> > User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
> > Session Initiation Protocol
> > Request-Line: INVITE sip:105 at my.asterisk.su SIP/2.0
> > Method: INVITE
> > Resent Packet: False
> > Message Header
> > Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-f5998d8a
> > From: spa3k pstn <sip:spa3k at my.asterisk.su>;tag=8fc58211a0dc60f2o1
> > To: <sip:105 at my.asterisk.su>
> > Remote-Party-ID: spa3k pstn
> <sip:spa3k at my.asterisk.su>;screen=yes;party=calling
> > Call-ID: daed83bd-b2b66b36 at 42.666.11.7
> > CSeq: 101 INVITE
> > Max-Forwards: 70
> > Contact: spa3k pstn <sip:biwaa1 at 42.666.11.7:5061>
> > Expires: 240
> > User-Agent: Sipura/SPA3000-2.0.11(GWa)
> > Content-Length: 430
> > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> > Supported: x-sipura
> > Content-Type: application/sdp
> > Message body
> > Session Description Protocol
> >
> > the connection completes, but asterisk does not have the pstn caller id.
> >
> > randy
> >
> >
> >
> > ------------------------------
> >
> > Message: 8
> > Date: Wed, 3 Nov 2004 15:40:33 -0700 (MST)
> > From: Manuel Marin <mmg at transtelco.com.mx>
> > Subject: [Asterisk-Users] How change default law for T100P
> > To: asterisk-users at lists.digium.com
> > Message-ID:
> > <653967.1099521633455.SLOX.WebMail.wwwrun at iGrup.transtelco.com.mx>
> > Content-Type: text/plain; charset=us-ascii
> >
> > I would like to know if there is a way to change default ulaw for a T1
> > card. I see in the zap show channel X that is working as ulaw. How do I
> > change it in zapata.conf or zaptel.conf to alaw. Iam interconnecting a
> > Meridian PBX but I need to configure it as alaw.
> >
> >
> >
> >
> > ------------------------------
> >
> > Message: 9
> > Date: Wed, 3 Nov 2004 17:40:55 -0500
> > From: niles at atheos.net
> > Subject: Re: [Asterisk-Users] What do I need to ask my T1 supplier?
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > <asterisk-users at lists.digium.com>
> > Message-ID: <6C40DB70-2DE9-11D9-B4EE-000A957899C8 at atheos.net>
> > Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
> >
> >
> > On Nov 3, 2004, at 5:32 PM, Scott Nelson wrote:
> >
> > > My employer is switching to a new T1 supplier (it was AT&T, we are now
> > > going
> > > with XO), and sometime in the future we want to replace our PBX with an
> > > Asterisk system.
> > >
> > > What do I need to know to make sure the T1 line is "provisioned" (is
> > > that the
> > > right term?) correctly for a Digium T100P/TE410P/TE405P?
> > >
> > > They will split the T1 line into 10 channels of voice and 14 channels
> > > of data.
> > > From what I understand, they will terminate the T1 into a channel
> > > bank, and
> > > then from that give is 10 POTS phone jacks and one data port (to go to
> > > an
> > > Adtran router for our Internet access).
> > >
> > > Any comments and/or suggestions?
> > >
> > > Scott
> > > _______________________________________________
> >
> > Scott,
> >
> > you can skip the channel bank & router, and use asterisk with a T100P to
> > serve your data & voice. You can find all the info you need on the Wiki
> > http://www.voip-info.org/tiki-index.php?
> > page=Asterisk%20Data%20Configuration
> >
> > I use this setup for 11 voice channels and 256K of data from Nuvox.
> > Niles
> >
> >
> >
> >
> > ------------------------------
> >
> > Message: 10
> > Date: Wed, 03 Nov 2004 14:47:09 -0800
> > From: TC <trclark at shaw.ca>
> > Subject: Re: [Asterisk-Users] What do I need to ask my T1 supplier?
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > <asterisk-users at lists.digium.com>
> > Message-ID: <032601c4c1f7$0dd0ff20$c901a8c0 at w2ktopcat>
> > Content-Type: text/plain; charset=iso-8859-1
> >
> >
> >
> > >They will split the T1 line into 10 channels of voice and 14 channels of
> > data.
> > >From what I understand, they will terminate the T1 into a channel bank,
> and
> > >then from that give is 10 POTS phone jacks and one data port (to go to an
> > >Adtran router for our Internet access).
> >
> > >Any comments and/or suggestions?
> >
> > what would be realy nice from them is to present those 10 voice channels
> > as not POTS but as the first 10 channels of a pri t1 net interface ie a
> > fractional t1 voice
> > and skip the a/d nonsense I know an adit 600 with a router & t1 cards can
> do
> > that for you
> >
> >
> >
> >
> > ------------------------------
> >
> > Message: 11
> > Date: Wed, 3 Nov 2004 17:01:11 -0600
> > From: "Matthew Boehm" <mboehm at cytelcom.com>
> > Subject: [Asterisk-Users] Cisco 79XX - Using built-in 3way conference
> > To: <asterisk-users at lists.digium.com>
> > Message-ID: <016b01c4c1f9$1508c410$8100000a at cytelcom.com>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Hey guys,
> > This has worked before but for some reason isn't anymore and I have no
> clue
> > what to check.
> > Here are the steps I follow:
> >
> > 1. Place call to PSTN number. They answer and we talk.
> > 2. I press 'Conference' button on Cisco phone.
> > 3. Line 1 is now on hold and I get a new dial tone.
> > 4. Place call 2 to another PSTN. They answer and we talk.
> > 5. I press 'Join' on the Cisco phone. Caller 1 gets dropped and I get the
> > following
> > message in * console:
> >
> > Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from
> > 10.0.0.122
> >
> > Now, 10.0.0.122 is the IP of my Cisco phone. * has 2 NICs, 1 is 10.0.3.10
> > and the other is external public IP. I can make/recieve calls all day
> long.
> > But recently this conference stopped working.
> >
> > Any ideas on what to check? The error doesn't make sense since the 2 calls
> > are present. Right before I press join, I can put caller 2 on hold and
> > resume caller 1 and vice versa. It isn't until I press 'Join' that call 1
> is
> > dropped.
> >
> > This works fine if caller 1 and 2 are both other phones in the office or
> > caller 1 is a phone in office and 2 is PSTN. Just doesn't work when both
> > PSTN. Worked before..
> >
> > THanks,
> > Matthew
> >
> >
> >
> > ------------------------------
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > End of Asterisk-Users Digest, Vol 4, Issue 52
> > *********************************************
> >
> _______________________________________________
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