[Asterisk-Users] BROADVOICE fails to register
Francisco Perez-Landaeta
fplandae at hotmail.com
Thu Nov 4 06:50:44 MST 2004
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
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-request at lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-owner at lists.digium.com
>
> 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
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20041103/d471c6ed/attachment-0001.html
>
> ------------------------------
>
> 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...
> URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20041103/34b4f9c5/attachment-0001.html
>
> ------------------------------
>
> 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
> *********************************************
>
More information about the asterisk-users
mailing list