Sorry, missed that you had a 601... yea, only legacy firmware (below 3.2).
I've not tested any legacy sip firmware above 3.1.3c. Mike On Tue, May 3, 2011 at 8:06 PM, Charles Chalekson <[email protected]>wrote: > Thanks so much. I thought the 601s only support up to SIP 3.1.7 and > BootROM 4.1.3? At least that's what they list on the polycom site? > > Charles > > > On 2011-05-03, at 4:55 PM, [email protected] wrote: > > Send sipx-users mailing list submissions to > > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://list.sipfoundry.org/mailman/listinfo/sipx-users > or, via email, send a message with subject or body 'help' to > [email protected] > > You can reach the person managing the list at > [email protected] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sipx-users digest..." > [please change the subject line when replying]Today's Topics: > > 1. Re: Multiple Public IP's (Joe Micciche) > 2. Re: Multiple Public IP's (Tony Graziano) > 3. Re: Multiple Public IP's (Joe Micciche) > 4. Audiocodes mp114fxo gateway (Wayne A. Green) > 5. Re: Bria 3.2 or 3.1 Doesn?t Work With (((( MWI )))) Message > waiting indicator (Yuri (( SipXecs ))) > 6. Re: Audiocodes mp114fxo gateway (Tony Graziano) > 7. Re: Polycom SIP application (Michael Picher) > > *From: *Joe Micciche <[email protected]> > *Date: *May 3, 2011 12:04:45 PM PDT > *To: *Tony Graziano <[email protected]> > *Cc: *Discussion list for users of sipXecs software < > [email protected]> > *Subject: **Re: [sipx-users] Multiple Public IP's* > *Reply-To: *[email protected], Discussion list for users of sipXecs > software <[email protected]> > > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 05/03/2011 09:14 AM, Tony Graziano wrote: > > I think Joegen is trying to point out that you don't populate this field > > with an external controller. You specify that you use an external > controller > > and route your sip calls via that. > > > Sorry for being dense: I take your comment to mean this field need not > be populated at all, in spite of the direction to do so? > > "When the server is deployed behind a NAT, the "Public IP address" field > must be set to the Internet-facing IP address of the NAT / firewall > device fronting the server." > > Our servers are behind NAT. > > - -- > ================================================================== > Joe Micciche [email protected] > Red Hat, Inc. http://www.redhat.com > Senior Communications Engineer X (81) 44554 > +1.919.754.4554 Key: 65F90FE1 > ================================================================== > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.11 (GNU/Linux) > Comment: Using GnuPG with Fedora - http://enigmail.mozdev.org/ > > iEYEARECAAYFAk3AUc0ACgkQJHjEUGX5D+Ff8wCfbFIRi0EmJcO1tA+19fWrDH09 > 9ygAn2OTeyYSxzi3f1quD2jmL4a7tjap > =WT1I > -----END PGP SIGNATURE----- > > > > > *From: *Tony Graziano <[email protected]> > *Date: *May 3, 2011 12:14:43 PM PDT > *To: *jmiccich <[email protected]> > *Cc: *Discussion list for users of sipXecs software < > [email protected]> > *Subject: **Re: [sipx-users] Multiple Public IP's* > *Reply-To: *Discussion list for users of sipXecs software < > [email protected]> > > > ah, but you are using an SBC, so this can all be "undone". If your > gateways/siptrunks are accessed via the SBC, and your remote users > come through the sbc, nonw of those boxes should be checked. > > On Tue, May 3, 2011 at 3:04 PM, Joe Micciche <[email protected]> wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > > Hash: SHA1 > > > On 05/03/2011 09:14 AM, Tony Graziano wrote: > > I think Joegen is trying to point out that you don't populate this field > > with an external controller. You specify that you use an external > controller > > and route your sip calls via that. > > > Sorry for being dense: I take your comment to mean this field need not > > be populated at all, in spite of the direction to do so? > > > "When the server is deployed behind a NAT, the "Public IP address" field > > must be set to the Internet-facing IP address of the NAT / firewall > > device fronting the server." > > > Our servers are behind NAT. > > > - -- > > ================================================================== > > Joe Micciche [email protected] > > Red Hat, Inc. http://www.redhat.com > > Senior Communications Engineer X (81) 44554 > > +1.919.754.4554 Key: 65F90FE1 > > ================================================================== > > > -----BEGIN PGP SIGNATURE----- > > Version: GnuPG v1.4.11 (GNU/Linux) > > Comment: Using GnuPG with Fedora - http://enigmail.mozdev.org/ > > > iEYEARECAAYFAk3AUc0ACgkQJHjEUGX5D+Ff8wCfbFIRi0EmJcO1tA+19fWrDH09 > > 9ygAn2OTeyYSxzi3f1quD2jmL4a7tjap > > =WT1I > > -----END PGP SIGNATURE----- > > > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.326.5325 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Contract Customers: > http://support.myitdepartment.net > Blog: > http://blog.myitdepartment.net > > Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > > > > > *From: *Joe Micciche <[email protected]> > *Date: *May 3, 2011 12:16:28 PM PDT > *To: *Tony Graziano <[email protected]> > *Cc: *Discussion list for users of sipXecs software < > [email protected]> > *Subject: **Re: [sipx-users] Multiple Public IP's* > *Reply-To: *[email protected], Discussion list for users of sipXecs > software <[email protected]> > > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Thanks for the clarification Tony! I'll fuss with the config off hours > to test. > > On 05/03/2011 03:14 PM, Tony Graziano wrote: > > ah, but you are using an SBC, so this can all be "undone". If your > > gateways/siptrunks are accessed via the SBC, and your remote users > > come through the sbc, nonw of those boxes should be checked. > > > On Tue, May 3, 2011 at 3:04 PM, Joe Micciche <[email protected]> wrote: > > On 05/03/2011 09:14 AM, Tony Graziano wrote: > > I think Joegen is trying to point out that you don't populate this field > > with an external controller. You specify that you use an external > controller > > and route your sip calls via that. > > > Sorry for being dense: I take your comment to mean this field need not > > be populated at all, in spite of the direction to do so? > > > "When the server is deployed behind a NAT, the "Public IP address" field > > must be set to the Internet-facing IP address of the NAT / firewall > > device fronting the server." > > > Our servers are behind NAT. > > > > > - -- > ================================================================== > Joe Micciche [email protected] > Red Hat, Inc. http://www.redhat.com > Senior Communications Engineer X (81) 44554 > +1.919.754.4554 Key: 65F90FE1 > ================================================================== > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.11 (GNU/Linux) > Comment: Using GnuPG with Fedora - http://enigmail.mozdev.org/ > > iEYEARECAAYFAk3AVIwACgkQJHjEUGX5D+GovACgsGnOm64k3et15Cdisp9Jo9YE > mMQAn1gmDZdxhVHLd29WrQMpXvOkQf1m > =LTjE > -----END PGP SIGNATURE----- > > > > > *From: *"Wayne A. Green" <[email protected]> > *Date: *May 3, 2011 3:18:23 PM PDT > *To: *[email protected] > *Subject: **[sipx-users] Audiocodes mp114fxo gateway* > *Reply-To: *"Wayne A. Green" <[email protected]>, Discussion list > for users of sipXecs software <[email protected]> > > > I have the following scenario: > 1. I am running 4.4 code. Centos 5.6 > 2. Audiocodes MP114-FXO version 6.0 code > 3. Snom 821 hand-sets with snom821-SIP 8.4.18 42604 code > > Internal calls work perfectly but I have an issue with external calling. > External call initiation works perfect but call are terminate at exactly 150 > to 200 seconds into the call. The call can be reinitiated but it is > terminated again. > I have determine the issue lies within the gateway but not certain of what > the cause may be. Any assistance will be greatly appreciated... > > Thanks, > > > > > > > *From: *"Yuri (( SipXecs ))" <[email protected]> > *Date: *May 3, 2011 3:51:47 PM PDT > *To: *[email protected] > *Subject: **Re: [sipx-users] Bria 3.2 or 3.1 Doesn´t Work With (((( MWI > )))) Message waiting indicator* > *Reply-To: *Discussion list for users of sipXecs software < > [email protected]> > > > The problem was solved! > > Don´t you belive? > > I change de status to Avaliable for busy and busy to avaliable and MWI > Beginning to work!!! > > I do not know what to say! > I think there may be some problem with the software! > > > Thanks anyway. > > 2011/5/2 Yuri (( SipXecs )) <[email protected]> > >> Good night >> >> >> I am usinga the Bria softphone and Polycom phones here! >> Message waiting indicator that works only on phones Poycom! >> >> Bria does not appear in the statement that has voicemail! >> >> All the devices are being managed in sipXecs, Bria and this usually by >> logging into the system! >> >> Thank you. > > > > > > *From: *Tony Graziano <[email protected]> > *Date: *May 3, 2011 4:14:07 PM PDT > *To: *"Wayne A. Green" <[email protected]>, Discussion list for > users of sipXecs software <[email protected]> > *Subject: **Re: [sipx-users] Audiocodes mp114fxo gateway* > *Reply-To: *Discussion list for users of sipXecs software < > [email protected]> > > > look at the syslog on the gateway and determine what the reason is for the > BYE. > > On Tue, May 3, 2011 at 6:18 PM, Wayne A. Green <[email protected]> > wrote: > > I have the following scenario: > > 1. I am running 4.4 code. Centos 5.6 > > 2. Audiocodes MP114-FXO version 6.0 code > > 3. Snom 821 hand-sets with snom821-SIP 8.4.18 42604 code > > > Internal calls work perfectly but I have an issue with external calling. > > External call initiation works perfect but call are terminate at exactly > 150 > > to 200 seconds into the call. The call can be reinitiated but it is > > terminated again. > > I have determine the issue lies within the gateway but not certain of what > > the cause may be. Any assistance will be greatly appreciated... > > > Thanks, > > > > > > _______________________________________________ > > sipx-users mailing list > > [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.326.5325 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Contract Customers: > http://support.myitdepartment.net > Blog: > http://blog.myitdepartment.net > > Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > > > > > *From: *Michael Picher <[email protected]> > *Date: *May 3, 2011 4:55:22 PM PDT > *To: *Discussion list for users of sipXecs software < > [email protected]> > *Subject: **Re: [sipx-users] Polycom SIP application* > *Reply-To: *Discussion list for users of sipXecs software < > [email protected]> > > > On mail prob, check /var/sipxdata/media... drill down in mailstore and > find the offending mailbox... if there is a new message for a user a small > empty file is in the user directory indicating a new message. Just wipe > the file and restart phone. > > Try re:downloading polycom firmware 3.2.4 or 3.2.5 with bootrom 4.3.0... > put the zips into a new device file entry. > On May 2, 2011 9:48 PM, "Charles Chalekson" <[email protected]> wrote: > > Two questions unrelated: > > > > 1] I had to restore my sipxecs to an old configuration which worked fine > except, I now have a message icon/flash occurring where there is no message > to listen to. How do I turn off the indicator? > > > > 2] I was upgrading my SIP application some polycoms 601s to SIP 3.1.3 > through the device files menu which has worked fine in the past and has > activated fine that way. I tried to do it today, and the phones were not > picking up the new SIP app that is being delivered there. I am running 4.1.4 > Boot ROM. The phone config file lists the app as > APP_FILE_PATH_SPIP600="sip_3.1.X.ld. I however do not see any file by that > name in the tftproot folder. If I rename the split config sip app file that > is supposed to work for the 601s [2345-11605-001.sip_317.ld] to sip_3.1X.ld, > then the phone will pick it up, but I know I shouldn't have to do that and I > have never had to do that previously. > > > > Any clues? > > Charles > > > > > > _______________________________________________ > > sipx-users mailing list > > [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > _______________________________________________ > sipx-users mailing list > [email protected] > http://list.sipfoundry.org/mailman/listinfo/sipx-users > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- There are 10 kinds of people in this world, those who understand binary and those who don't. [email protected] blog: http://www.sipxecs.info call: sip:[email protected]
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