[Asterisk-Users] X100p + cell socket no callerid
[EMAIL PROTECTED] root]# cat /proc/zaptel/1 Span 1: WCFXO/0 Wildcard X101P Board 1 1 WCFXO/0/0 FXSKS (In use) Asterisk CVS-HEAD-02/13/05-00:32:03, Copyright (C) 1999 - 2005 Digium. Feb 15 22:33:48 NOTICE[3002]: callerid.c:307 callerid_feed: Caller*ID failed checksum Feb 15 22:33:51 ERROR[3002]: callerid.c:261 callerid_feed: fsk_serie made mylen 0 (-6) Feb 15 22:33:51 WARNING[3002]: chan_zap.c:5613 ss_thread: CallerID feed failed: Success Feb 15 22:33:51 WARNING[3002]: chan_zap.c:5657 ss_thread: CallerID returned with error on channel 'Zap/1-1' I have verified I get callerid from a phone connected to the cell socket device. I experimented with txgain and rxgain but got similar results. Anyone have any ideas? Is it possible to disregard the checksum or see the value it is getting for the callerid. Thanks, Will ___ 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
[Asterisk-Users] Asterisk as ISDN simulator?
Anyone ever try it? is it possible? I am studying for my CCIE and ISDN simulators are very expensive. Thanks. Will
RE: [Asterisk-Users] Asterisk as ISDN simulator?
Take a data call in on one BRI and shoot it out on another. Sorry if I was not clear. Would look like this [cisco router with bri][asterisk w 2 bri cards]---[cisco router with bri] I am not to familiar with ISDN so I dunno if I could do this since I know pots has FXO/FXS and you can't go fxo to fxo. Thanks, Will -Original Message- From: Martin List-Petersen [mailto:[EMAIL PROTECTED] Sent: Sunday, March 28, 2004 10:36 PM To: [EMAIL PROTECTED] Cc: william carlson Subject: Re: [Asterisk-Users] Asterisk as ISDN simulator? Citat william carlson [EMAIL PROTECTED]: Anyone ever try it? is it possible? I am studying for my CCIE and ISDN simulators are very expensive. ISDN simulator in what way ? CCIE is far away for me yet, but you can definatly simulate a lot with hfc based cards. /Martin -- BOFH excuse #133: It's not plugged in. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone Unprovisioned Message in IP 7940 ?
proxy1_address: `129.82.44.223 it that ' really there? that could be it Thanks, Will - Original Message - From: Sascha Knific [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 08, 2003 6:07 AM Subject: AW: [Asterisk-Users] Phone Unprovisioned Message in IP 7940 ? Hi Tony The configuration looks fine to me. Did you check the log of your tftp server? Do the phone config files get loaded correctly? Do check also the Settings/Status/Status Messages of your phone for any errors. Sascha --- Sascha Knific K Systems Design Tel. +49-8151-773260 Wittelsbacherstr. 6a Fax. +49-8151-773262 82319 Starnberg, Germany Leo +49-8151-773261 WGS84: N57°59,875' E011°20,568' [EMAIL PROTECTED] http://www.k-sysdes.net -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von tony banks Gesendet: Montag, 8. Dezember 2003 03:21 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Phone Unprovisioned Message in IP 7940 ? Hello all, I am newbie to Telephony world (IP and PSTN). Please excuse me if you find my questions very dumb. I am trying to configure my IP 7940 with the Asterisk, when phone boots up it only shows the message Phone Unprovisioned on the LCD panel. Under Settings--SIP Configuration--Line 1 Settings I noticed that Proxy Address is set the UNPROVISIONED, I am not sure why it is showing that though I did set proxy1_address: `129.82.44.223 in SIPDefault.conf, which is my Astersik server. Following SIP image is installed on the IP 7940. Application Load ID POS30203 My sip.conf has following lines added for the the Phone [810] type=friend secret=pass host=dynamic callerid=JOSE 810 defaultip=129.82.44.205 In my SIPmac.conf file I have made following entries # Line 1 appearance line1_name: 810 # Line 1 Registration Authentication line1_authname: 810 # Line 1 Registration Password line1_password: pass Do you see any problem here, Please let me know if I should give any more information. Regards Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco 7960 intercom
How would I go about setting this up. I have a few 7960's with an extension set to autoanswer. How do I let all extensions answer and be active? Thanks, Will
Re: [Asterisk-Users] Skinny (SCCP) help
This is where I got the ringtones.
http://www.loligo.com/asterisk/sounds/
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 08, 2003 9:25 AM
Subject: Re: [Asterisk-Users] Skinny (SCCP) help
On Thu, Nov 06, 2003 at 12:57:49AM -0500, William Carlson wrote:
Ok I see the confusion. I actually do have a TFTP server running on the
asterisk machine but it does not have any Skinny stuff just ringtones
and
logos for my SIP 7960's. The id is found under settings then model
information just add SEP in front of the MAC address.
Thanks,
Will
Where do you get the 7960 ring tones and logos?
sorry to cut in like this; very new to * and skinny phones;
do you mean, all i need to install is *; no need to activate linux's
tftp daemon?
I have mine working w/o the tftp server running on my * machine. I just
set
the dhcpd option for the tftp server to the ip addr of my * machine.
also, is the device name something i make up or burned in the phone's
rom ; is so, where can i find the device name?
This is just `SEP` . mac address
[general]
dateFormat = M-D-Y ; M,D,Y in any order (5 chars max)
keepAlive = 120
I had to put this in to get the voice to go from the 7960 to *:
bindaddr = 192.168.254.179 ; Address to bind to
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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Re: [Asterisk-Users] Skinny (SCCP) help
woops I ment
http://www.loligo.com/asterisk/Cisco/79xx/current/
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 08, 2003 9:25 AM
Subject: Re: [Asterisk-Users] Skinny (SCCP) help
On Thu, Nov 06, 2003 at 12:57:49AM -0500, William Carlson wrote:
Ok I see the confusion. I actually do have a TFTP server running on the
asterisk machine but it does not have any Skinny stuff just ringtones
and
logos for my SIP 7960's. The id is found under settings then model
information just add SEP in front of the MAC address.
Thanks,
Will
Where do you get the 7960 ring tones and logos?
sorry to cut in like this; very new to * and skinny phones;
do you mean, all i need to install is *; no need to activate linux's
tftp daemon?
I have mine working w/o the tftp server running on my * machine. I just
set
the dhcpd option for the tftp server to the ip addr of my * machine.
also, is the device name something i make up or burned in the phone's
rom ; is so, where can i find the device name?
This is just `SEP` . mac address
[general]
dateFormat = M-D-Y ; M,D,Y in any order (5 chars max)
keepAlive = 120
I had to put this in to get the voice to go from the 7960 to *:
bindaddr = 192.168.254.179 ; Address to bind to
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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Re: [Asterisk-Users] 6.0 image for Cisco 7960's?
Nice this image lets my flakey 7960 run the SIP software :) Thanks, Will - Original Message - From: Paul Mahler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 08, 2003 10:09 AM Subject: RE: [Asterisk-Users] 6.0 image for Cisco 7960's? The 6.0 image is available for download from Cisco TAC. The 6.0 image does support auto answer (Intercom.) Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Thursday, November 06, 2003 1:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 6.0 image for Cisco 7960's? Has anyone managed to get their hands on a 6.0 image for their 7960's yet? Or is it still in beta? Rumor (official rumor, from Cisco) is that it supports paging and intercom. I'm anxious to start working with those features, if they've been implemented sanely. What would be just as nice would be NOTIFY messages for pushing XML URL's to the phones, but sadly that feature request has gone uncommented-upon by Cisco, so I will assume the worst... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream ntp
after awhile? I have had mine running for the past week or so with no problems. Although my NTP server is a cisco not the asterisk box. Thanks, Will - Original Message - From: Sean Rodger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 07, 2003 8:40 AM Subject: [Asterisk-Users] grandstream ntp I am running ntpd on the same machine as asterisk in order for the grandstream phones to display the time. After a while the time display fails until the phone is re-booted. Has anyone run into this problem before? Is it simply a bug in the GS firmware? Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream problem
Does everything work fine now? I am still having
problems with SayUnixTime. Voicemailmain2 works
though. The one simple AGI script I wrote doesn't do anything. Asterisk starts
playing and the grandstream just rings. Both work fine on other channels/sip
phones.
Thanks,
Will
- Original Message -
From:
Wim
Venneman
To: [EMAIL PROTECTED]
Sent: Friday, November 07, 2003 1:46
PM
Subject: Re: [Asterisk-Users] Grandstream
problem
Thanks William,
Works fine now.
Wim
- Original Message -
From:
William
Carlson
To: [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 9:43
PM
Subject: Re: [Asterisk-Users]
Grandstream problem
try
disallow=all
allow=ulaw
under the general section of
sip.conf
that half fixes it for me calls between phones
work but talking to asterisk has some problems.
- Original Message -
From:
Wim
Venneman
To: [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003
2:29 PM
Subject: [Asterisk-Users] Grandstream
problem
Hi,
I installed Asterisk an all works
fineexept for Grandstream.
When I call with a softphone (ex X-ten) to a
Grandstream (BudgetTone-100), I can make a conversation. = ok
WhenI call to a softphone with a
Grandstream I can pich up the call with the softphone but the Grandstream
keeps ringing like on the other site you didn't pick up the phone.(even if
you do so)
It's the same when I call between two
Grandstream phone's. Call from phone1 to phone 2, I pick up phone2 and
afther 3 seconds I get congestion tone from both phone's.
Info from command *CLI
-- Executing Dial("SIP/phone2-a030a",
"sip/phone1") in new stack
-- Called phone1
-- SIP/phone1-663a is ringing
-- SIP/phone1-663a answered
SIP/phone2-a030a
-- Attempting native bridge of
SIP/phone2-a030a and SIP/phone1-663a
== Spawn extension (sip, 1,1) exited
non-zero on 'SIP/phone2-a030a'
and I get congestion
Can anyone give me a direction to solve my
problem?
Thanks in advance,
Wim
Re: [Asterisk-Users] To SIP or Not?
I have 2 7960's one sip one skinny. I am using the skinny channel never used the sccp. I am not getting callerid with the skinny version but other than that(it's a big one) they both work. I dunno if I just have it configed wroung or if caller id is just not supported. Thanks Will - Original Message - From: David Stubbs [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 06, 2003 12:13 PM Subject: [Asterisk-Users] To SIP or Not? Hi all, we have go a bunch of cisco 7940 phones, i currently wondering wether to use the sccp channel of sip. Could some one educate me on the features / advantages of each, as I'm unsure of witch one to use? Thanks David, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream problem
try
disallow=all
allow=ulaw
under the general section of sip.conf
that half fixes it for me calls between phones work
but talking to asterisk has some problems.
- Original Message -
From:
Wim
Venneman
To: [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 2:29
PM
Subject: [Asterisk-Users] Grandstream
problem
Hi,
I installed Asterisk an all works fineexept
for Grandstream.
When I call with a softphone (ex X-ten) to a
Grandstream (BudgetTone-100), I can make a conversation. = ok
WhenI call to a softphone with a
Grandstream I can pich up the call with the softphone but the Grandstream
keeps ringing like on the other site you didn't pick up the phone.(even if you
do so)
It's the same when I call between two Grandstream
phone's. Call from phone1 to phone 2, I pick up phone2 and afther 3 seconds I
get congestion tone from both phone's.
Info from command *CLI
-- Executing Dial("SIP/phone2-a030a",
"sip/phone1") in new stack
-- Called phone1
-- SIP/phone1-663a is ringing
-- SIP/phone1-663a answered
SIP/phone2-a030a
-- Attempting native bridge of SIP/phone2-a030a
and SIP/phone1-663a
== Spawn extension (sip, 1,1) exited
non-zero on 'SIP/phone2-a030a'
and I get congestion
Can anyone give me a direction to solve my
problem?
Thanks in advance,
Wim
[Asterisk-Users] SIP broken for budgtone.
I just downloaded the newest version from CVS([EMAIL PROTECTED]) and I am getting an error whenever
I call the asterisk box. I cannot here any audio on the budgtone. This works
fine with my pingtel phone and my sip 7960. Also if I call my Skinny 7960 it
rings but I get that same error when I pick up. When the skinny phone calls the
Budgtone it works fine. I have 2 budgtone phones and it does this on both of
them. This worked fine before I installed the newest version of
asterisk.
-- Executing
Playback("SIP/budgtone-7ee9", "carried-away-by-monkeys") in new
stack -- Playing 'carried-away-by-monkeys' (language
'en') -- Executing Playback("SIP/budgtone-7ee9",
"lots-o-monkeys") in new stack -- Playing 'lots-o-monkeys'
(language 'en')WARNING[40966]: File chan_sip.c, Line 456 (retrans_pkt):
Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1735 (Response)
With sip debug
Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
192.168.1.223 From: "William Carlson"
sip:[EMAIL PROTECTED];tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To:
sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 62159 INVITE User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp Content-Length: 263 v=0 o=budgtone 0 0 IN
IP4 192.168.1.223 s=- c=IN IP4 192.168.1.223 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18
2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000
12 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.223 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format ULAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 285/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Reliably Transmitting (no NAT):SIP/2.0 407 Proxy Authentication
Required Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson"
sip:[EMAIL PROTECTED];tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To:
sip:[EMAIL PROTECTED];tag=as67b6f854 Call-ID: [EMAIL PROTECTED]
CSeq: 62159 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk",
nonce="6c3e5732" Content-Length: 0 to 192.168.1.223:5060
Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
192.168.1.223 From: "William Carlson"
sip:[EMAIL PROTECTED];tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To:
sip:[EMAIL PROTECTED];tag=as67b6f854 Contact:
sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED]
CSeq: 62159 ACK User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow:
INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Length: 0
11 headers, 0 lines
Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
192.168.1.223 From: "William Carlson"
sip:[EMAIL PROTECTED];tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]
Proxy-Authorization: DIGEST username="budgtone", realm="asterisk",
algorithm=MD5, uri="sip:[EMAIL PROTECTED]", nonce="6c3e5732",
response="4e90c985822b15d83f297e8c4fe80372" Call-ID: [EMAIL PROTECTED]
CSeq: 62160 INVITE User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp Content-Length: 263 v=0 o=budgtone 0 0 IN
IP4 192.168.1.223 s=- c=IN IP4 192.168.1.223 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18
2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000
13 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.223 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format ULAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 285/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 9998 in default
list_route: hop: sip:[EMAIL PROTECTED]
Transmitting (no NAT):SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.223
From: "William Carlson"
sip:[EMAIL PROTECTED];tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
sip:[EMAIL PROTECTED];tag=as5481a27e Call-ID: [EMAIL PROTECTED]
CSeq
Re: [Asterisk-Users] Skinny (SCCP) help
I just got mine working. All I did was create a skinny.conf and point the phone to the asterisk server for tftp. the phone then boots and says useing TFTP as CM and works. I have no SEP.cnf's on my tftp server. my skinny.conf is [general] dateFormat = M-D-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 [will] device=SEP000750834016 context=default callerid=William carlson linelabel= mailbox= line = - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 1:24 PM Subject: [Asterisk-Users] Skinny (SCCP) help I have a cisco 7910 phone, I'm trying to get it to connect to asterisk, But it seems like it needs either a SEPDefault.cnf file or a SEPMACADDR.cnf file to Continue, I created empty ones but it's still sitting there saying opening Does anyone have examples of the SEPDefault.cnf file? Kevin, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] To anyone with a grandstream budgetone...
I logged a bug I wanted to see if anyone else is
having this problem or if it's just me.
http://bugs.digium.com./bug_view_page.php?bug_id=486
I just downloaded the newest version from CVS([EMAIL PROTECTED]) and I am
getting an error whenever I call the asterisk box. I cannot here any audio on
the budgtone. This works fine with my pingtel phone and my sip 7960. Also if I
call my Skinny 7960 it rings but I get that same error when I pick up. When the
skinny phone calls the Budgtone it works fine. I have 2 budgtone phones and it
does this on both of them. This worked fine before I installed the newest
version of asterisk. -- Executing Playback("SIP/budgtone-7ee9",
"carried-away-by-monkeys") in new stack -- Playing 'carried-away-by-monkeys'
(language 'en') -- Executing Playback("SIP/budgtone-7ee9", "lots-o-monkeys")
in new stack -- Playing 'lots-o-monkeys' (language 'en') WARNING[40966]:
File chan_sip.c, Line 456 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno
1735 (Response)
Thanks,
Will
Re: [Asterisk-Users] Skinny (SCCP) help
Actually I consider myself unlucky a this phone does not work with the sip load. Phone works fine with a skinny and mgcp load though. I am familiar with the config for the SIP and MGCP config files is there an example anywhere. Ciscos site is less then helpful on the Skinny documentation. Thanks, Will - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 06, 2003 12:02 AM Subject: Re: [Asterisk-Users] Skinny (SCCP) help Kevin wrote: Interesting, you must have a newer firmware than what is on my 7910 Beause mine just keeps saying opening with the little spinning line, It seems like it needs these sep*.cnf files to get some configuration settings Because it keeps trying to fetch these sep*.cnf files. Even though when I goto settings it says that call manager 1 is set to my * server. I created chan_skinny using a Call Manager 3.1 version. This way you get an XML config file. -Original Message- I just got mine working. All I did was create a skinny.conf and point the phone to the asterisk server for tftp. the phone then boots and says useing TFTP as CM and works. I have no SEP.cnf's on my tftp server. Consider yourself lucky. I use XMLDefault.cnf.xml. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skinny (SCCP) help
Ok I see the confusion. I actually do have a TFTP server running on the asterisk machine but it does not have any Skinny stuff just ringtones and logos for my SIP 7960's. The id is found under settings then model information just add SEP in front of the MAC address. Thanks, Will - Original Message - From: hkirrc.patrick [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 11:12 PM Subject: Re: [Asterisk-Users] Skinny (SCCP) help sorry to cut in like this; very new to * and skinny phones; do you mean, all i need to install is *; no need to activate linux's tftp daemon? also, is the device name something i make up or burned in the phone's rom ; is so, where can i find the device name? William Carlson wrote: I just got mine working. All I did was create a skinny.conf and point the phone to the asterisk server for tftp. the phone then boots and says useing TFTP as CM and works. I have no SEP.cnf's on my tftp server. my skinny.conf is [general] dateFormat = M-D-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 [will] device=SEP000750834016 context=default callerid=William carlson linelabel= mailbox= line = - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 1:24 PM Subject: [Asterisk-Users] Skinny (SCCP) help I have a cisco 7910 phone, I'm trying to get it to connect to asterisk, But it seems like it needs either a SEPDefault.cnf file or a SEPMACADDR.cnf file to Continue, I created empty ones but it's still sitting there saying opening Does anyone have examples of the SEPDefault.cnf file? Kevin, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk nightmare from hell!
Perhaps something in /var/log/asterisk/cdr-csv/Master.csv can help. That will at least tell you what channel the call came in on and where it went. Hope it helps, Will - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 8:28 PM Subject: [Asterisk-Users] asterisk nightmare from hell! Ok for those of you all up in a tizzy over my subject line, please don't take it literally because I'm certainly not saying that asterisk is the problem here. I just got a little nightmare problem that I need a bit of help figuring out. I installed an asterisk system a few months ago for a client, it has run almost flawlessly with the exception of a few small glitches. However, I got a call from my onsite liason today stating that they had encountered a very strange problem with the telephone system. The problem was very simple, when 2 callers today reported that when calling the firms main number (the asterisk server I set up) they received a menu for another business (this is my nightmare) which happened to be a pretty mature type of business. The one caller simply hungup, the other caller actually pressed one of the options and it actually took him to the voicemailbox of one of the actual persons of the actual business he was intending to call. Now normally I would blame this on perhaps a cross line with my incoming Verizon PRIs or something; however what made the story even more strange is that the menu that he got for the mature business is a business that I also service and have installed an asterisk server in. The 2 asterisk servers are in no way connected to the other. Also, me and members of my staff called the intended business all afternoon and got their normal menu as we were supposed to so therefore I was unable to recreate / emulate the problem. Now there is a remote possibility that in going through converting wav files to gsm I actually gave one the wrong name or something accidentally since I have the same lady do all my voice over work. However, I doubt this is the case because when I call the intended business and step through each of their menus / submenus that I created it plays the correct messages. My question is how can I go back and try to figure out exactly what happened? I have a very close approximation as to what time one of the calls came in so if there were some way I could go back and pull a record that looks something like the CLI, I could probably see exactly what happened and correct it. But everywhere I've looked I see no file where I'm able to do this and nothing gives me any clues. I've checked both /var/log/asterisk/messages as well as /var/log/asterisk/event_log. The messages log shows some things from today's date but it looks mostly like actual app stuff. The event_log has multiple entries but the last entry there is when I upgraded the asterisk server about 2 weeks ago. As for a little background, I'm running a pretty straight configuration with an incoming PRI and a 24 channelbank on the inside. All analog handsets. The only exception I sometimes play around with some IAX clients on the box but this is very limited. Thanks alot for any suggestions, sorry if it sounds stupid. It's just my nightmare. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
I cannot seem to get the software to work on my machine. I am multihomed running windows XP home. Perhaps the software is binding to the card not connected to asterisk. If I turn on debugging in asterisk I see no IAX stuff coming in from the IP. Thanks, Will - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 3:21 AM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Please provide your feedback about the application Only in that way it can be improoved. Thanks! Dan - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 12:13 AM Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform) Finaly, someone has started the IAX soft phone ball :) Thanks, Dan... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
I did set it up to register here is my iax.conf config. [blah] type=friend user=blah secret=blah context=default host=192.168.5.200 This is what I am seeing in asterisk. NOTICE[32773]: File chan_iax.c, Line 2708 (register_verify): Peer 'blah' is not dynamic (from 192.168.5.200) Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: REGREQ Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: REGREJ Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: REGREQ Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: ACK Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: REGREQ Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: REGREJ etc Ok I figured it out I need to change the host field to host=dynamic Thanks, Will - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 5:51 AM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Hi Will, - Original Message - From: William Carlson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 12:31 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) I cannot seem to get the software to work on my machine. I am multihomed running windows XP home. Perhaps the software is binding to the card not connected to asterisk. If I turn on debugging in asterisk I see no IAX stuff coming in from the IP. Thanks, Will In order to see something in the * console you must register first. Have you enter your credentials and * server IP address when asked? If not registered, nothing works and the application closes by himself. Please give me more details about this behaviour. It must work on a multihomed computer too if a correct route exists to the Asterisk server. If you can ping it, then it is only a registering problem. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960
I bought a 7960 it was running version 3.3 of the SIP software. It worked fine. Me being the idiot I am upgraded to 5.1. Now it downloads the configs and then reboots. if I unplug the ethernet it doesn't rebootor if I remove all the lines in the SIP config it won't reboot. Since this is used cisco won't give me any support. For now I am running the MGCP version but eh asterisk seems to have some issues with it. Thanks, Will
Re: [Asterisk-Users] Cisco 7960
lol well I probaly should ask a question lol. Any idea what could be causing this? Also I cannot call from my pingtel phone to the 7960 but I can call the other way around. any ideas on that? Thanks, Will - Original Message - From: William Carlson To: [EMAIL PROTECTED] Sent: Thursday, July 17, 2003 7:34 AM Subject: [Asterisk-Users] Cisco 7960 I bought a 7960 it was running version 3.3 of the SIP software. It worked fine. Me being the idiot I am upgraded to 5.1. Now it downloads the configs and then reboots. if I unplug the ethernet it doesn't rebootor if I remove all the lines in the SIP config it won't reboot. Since this is used cisco won't give me any support. For now I am running the MGCP version but eh asterisk seems to have some issues with it. Thanks, Will
Re: [Asterisk-Users] Cisco 7960
Cisco's website has some stuff on there website which seems to indicate if the 7960 cannot contact the call manager server it reboots. However to my knowledge this has never had call manager software before and cisco doesn't mention this "feature" with the SIP firmware. I downgraded to 5.0 unfortunately due to only being able to run Secure images now thats as far back as I can go. Thanks again cisco for this "feature". From what I can tell the phone never talked to the Asterisk box. If I turn on SIP debugging I do not see any traffic coming from the cisco box. Although I did have them on seperate subnets. Let me try putting them on the same subnet and see if that helps. Thanks, Will - Original Message - From: Matthew Hardeman To: [EMAIL PROTECTED] Sent: Thursday, July 17, 2003 12:58 PM Subject: RE: [Asterisk-Users] Cisco 7960 Ive run into this before, and its a pain to debug Be sure that your eth0 interface (primary, first interface) is set to your internal address space (of the same subnet that you assign to the phone). You can add an IP alias on eth0:1 if you need an external IP on that box as well, but you must have them in that order: internal = eth0, externals, others eth0:1+ Try that Matt PaperSoft -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William CarlsonSent: Thursday, July 17, 2003 6:35 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Cisco 7960 I bought a 7960 it was running version 3.3 of the SIP software. It worked fine. Me being the idiot I am upgraded to 5.1. Now it downloads the configs and then reboots. if I unplug the ethernet it doesn't rebootor if I remove all the lines in the SIP config it won't reboot. Since this is used cisco won't give me any support. For now I am running the MGCP version but eh asterisk seems to have some issues with it. Thanks, Will
