Anyone know if it is possible to load your own XML scripts on to the phone, bypassing the Cisco CallManager? I am still waiting for my phone to arrive, but I have been playing with Cisco's phone services emulator, and that doesn't seem to like anything I pass to it. If it is possible, anyone want to share any sample scripts they have. --Mike
What mode are you running the Phone in? SIP, MCGP, or SCCP (Skinny) You mentioned Call Manager so I will assume SCCP. If that is the case I do not know. However if you are running it in SIP, All you have to do is set # XML URLs services_url: "" ; URL for external Phone Services directory_url: "" ; URL for external Directory location logo_url: "" ; URL for branding logo to be used on phone display These in you configuration and point it to a webserver. The xml format can be found here. http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/bxtml.htm Hope this helps! Stephen On Wed, 2003-03-12 at 18:59, Mike Reiling wrote:> Anyone know if it is possible to load your own XML scripts on to the > phone, bypassing the Cisco CallManager? I am still waiting for my > phone to arrive, but I have been playing with Cisco's phone services > emulator, and that doesn't seem to like anything I pass to it. > > If it is possible, anyone want to share any sample scripts they have. > > --Mike > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
I send example XML files a week or so ago, as well as an impotent line for you dhcp server. Mike Reiling wrote:> Anyone know if it is possible to load your own XML scripts on to the > phone, bypassing the Cisco CallManager? I am still waiting for my > phone to arrive, but I have been playing with Cisco's phone services > emulator, and that doesn't seem to like anything I pass to it. > > If it is possible, anyone want to share any sample scripts they have. > > --Mike > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
Has anyone used the MGCP firmware image? I have only seen talk about the SIP image. Just curious! Stephen
You need a CCO. Then you can go to the Software Center has a MGCP image for the 7960. Also you can poke around in the Cisco Engineering FTP site and you might get lucky. Jeremy McNamara Stephen Webb wrote:>Has anyone used the MGCP firmware image? I have only seen talk about the >SIP image. > >Just curious! > >Stephen > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users > >
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 reboot or 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
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William,
I am running 7960/7940's with 5.1 (Asterisk SIP) without problems although I
did have some issues (too numerous to mention) with new phones that had never
been operated on a CallManager network first. It seems the firmware must be
upgraded to support SIP and this can only be done with CallManager (apparently).
The only way I managed to figure everything out was with a packet analyser, I
don't suppose you have the possibility of doing that ?
Rgds, Adam
-----Original Message-----
From: William Carlson [mailto:wcarlson@ici.net]
Sent: 17 July 2003 13:40
To: asterisk-users@lists.digium.com
Subject: 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 <mailto:wcarlson@ici.net>
To: asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com>
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 reboot or 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
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I need some help with upgrading a 7960. Any of you guys familiar with that ? I friend of mine have a couple of 7960 , and would like to get 'em to work. /Mike
This one helped me a lot : http://www.loligo.com/asterisk/Cisco/79xx/ kind regards Michael Devenijn www.dkma.be ________________________________ Van: Micke Andersson [mailto:micke@party.pp.se] Verzonden: vr 24/10/2003 9:33 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] Cisco 7960 I need some help with upgrading a 7960. Any of you guys familiar with that ? I friend of mine have a couple of 7960 , and would like to get 'em to work. /Mike _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4344 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20031024/c28163bf/attachment.bin
Does anybody know or have good examples of using all functions in a 7960 (SIP) /Mike
On Tue, 2004-03-02 at 06:35, Micke Andersson wrote:> Does anybody know or have good examples of using all functions in a 7960 > (SIP)http://voip-info.org/wiki-Asterisk+phone+cisco+79xx F
Hi all, i am new to this list and i dot not know, if anybody had already the same problem. I have two cisco 7960 which i want to upgrade to sip. Has somebody already taken the upgrade-process for special hints and suggestions? I have already visited the cisco-page and i have read the proposal for the migration. Is there a special order of firmware-upgrades? Thanks a lot Thomas
Thomas, The definitive guide of what versions can be upgraded to what is at: http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guides09186a008022a968.html In particular, look at tables 2 and 3. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Thomas Trepper wrote:> Hi all, > > i am new to this list and i dot not know, if anybody had already the > same problem. I have two cisco 7960 which i want to upgrade to sip. Has > somebody already taken the upgrade-process for special hints and > suggestions? I have already visited the cisco-page and i have read the > proposal for the migration. Is there a special order of firmware-upgrades? > > Thanks a lot > > Thomas > > _______________________________________________ > 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 > >
> Whilst I agree with Joe, has anybody actually been able to > sucessfuly get the Cisco 7940's/7960's to register into *?Yeah, I've been using a 7960 with * since November. Nabeel
> Date: Sun, 06 Mar 2005 20:03:52 +0100 > From: Thomas Trepper <thomas.trepper@microbyte.at> > Subject: [Asterisk-Users] Cisco 7960 > To: asterisk-users@lists.digium.com > Message-ID: <422B5418.4060607@microbyte.at> > Content-Type: text/plain; charset=us-ascii; format=flowed > > Hi all, > > i am new to this list and i dot not know, if anybody had already the > same problem. I have two cisco 7960 which i want to upgrade to sip. > Has somebody already taken the upgrade-process for special hints and > suggestions? I have already visited the cisco-page and i have read > the proposal for the migration. Is there a special order of firmware- > upgrades? > > Thanks a lot > > Thomas >Thomas The asterisk-wiki is the best place to start. It will tell you that it is a 3 stage process if your currently on call-manager. You will need to load the version 3, then 5 and then 7 SIP firmware. I tried to load the version 7 straight away and of course it wouldnt work. Please read the wiki and all will be revealed. Dont expect very much from the cisco website at all ! Regards..pete
Hi There I am currently having an issue with a Cisco 7960. The phone is using SIP firmware version 6.3. I have successfully got the phone to register with Asterisk and I can call the phone from other non Cisco handsets. However when I dial out from the 7960 I do not even see any output on the Asterisk console. Is there some sort of DTMF setting which I might have incorrectly set ? Following DTMF settings are in my SIPdefault.cnf file. dtmf_inband: 1 dtmf_outband: avt dtmf_db_level: 3 It seems to me like Asterisk is not detecting any key tones from the phone. I have followed a number of setup guides for this phone to no avail. Any help or suggestions are greatly appreciated. Regards Ed
Modify the dialplan.xml on your tftp server to this
<DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="4"
User="Phone"/>
</DIALTEMPLATE>
Jason
On Thu, 10 Mar 2005 10:31:34 -0000, Marshall, Ed
<ed.marshall@cetelem.co.uk> wrote:> Hi There
>
> I am currently having an issue with a Cisco 7960. The phone is using SIP
> firmware version 6.3. I have successfully got the phone to register with
> Asterisk and I can call the phone from other non Cisco handsets. However
> when I dial out from the 7960 I do not even see any output on the Asterisk
> console. Is there some sort of DTMF setting which I might have incorrectly
> set ? Following DTMF settings are in my SIPdefault.cnf file.
>
> dtmf_inband: 1
> dtmf_outband: avt
> dtmf_db_level: 3
>
> It seems to me like Asterisk is not detecting any key tones from the phone.
> I have followed a number of setup guides for this phone to no avail.
>
> Any help or suggestions are greatly appreciated.
>
> Regards
> Ed
>
> _______________________________________________
> 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
>
On Thu, 10 Mar 2005 10:31:34 -0000, Marshall, Ed <ed.marshall@cetelem.co.uk> wrote:> Hi There > > I am currently having an issue with a Cisco 7960. The phone is using SIP > firmware version 6.3. I have successfully got the phone to register with > Asterisk and I can call the phone from other non Cisco handsets. However > when I dial out from the 7960 I do not even see any output on the Asterisk > console. Is there some sort of DTMF setting which I might have incorrectly > set ? Following DTMF settings are in my SIPdefault.cnf file. > > dtmf_inband: 1 > dtmf_outband: avt > dtmf_db_level: 3A sip phone using the SIP protocol never sends actual DTMF when dialing, instead it uses an invite message to asterisk, which has the digits dialed in it. The tones you hear -when pushing the buttons on your phone- are actualy not sent to asterisk until you either press dial or #, or if your dialplan setup in diaplan.xml sends it. The inband and outband settings are only used when already talking on the phone (I might be wrong on this one, anybody out there if I am, please let me know). What does the phones display tell you when you dial?> It seems to me like Asterisk is not detecting any key tones from the phone. > I have followed a number of setup guides for this phone to no avail. > > Any help or suggestions are greatly appreciated. > > Regards > Ed > > _______________________________________________ > 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 >
does one know how to program so i can have 2 lines on one sip account on that phone ? im runnign my own asterisk do i need 2 local accounts ? one for each line ? that rebounds to same SIP forp VOIP provider ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060405/c38052d8/attachment.htm
On Wed, 2006-04-05 at 17:54 -0400, Jimmy Smith wrote:> does one know how to program so i can have 2 lines on one sip account > on that phone ? > > im runnign my own asterisk > > do i need 2 local accounts ? one for each line ? that rebounds to same > SIP forp VOIP provider ?Yes.
On Wed, 5 Apr 2006, Greg Oliver wrote:> On Wed, 2006-04-05 at 17:54 -0400, Jimmy Smith wrote: >> does one know how to program so i can have 2 lines on one sip account >> on that phone ? >> >> im runnign my own asterisk >> >> do i need 2 local accounts ? one for each line ? that rebounds to same >> SIP forp VOIP provider ? > > > Yes.The cisco phones can have multiple lines with the same registration... we had our phones set up like that until we decided to move to a one line call waiting type system. You just put the same account information in the configuration file for the second line as for the first line. -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198