Re: [asterisk-users] Simple failover configuration
Take a look at this doc from Polycom...it answers your question I think. https://encrypted.google.com/url?sa=trct=jq=polycom%20redundant%20serversource=webcd=1cad=rjaved=0CEUQFjAAurl=http%3A%2F%2Fsupport.polycom.com%2Fglobal%2Fdocuments%2Fsupport%2Ftechnical%2Fproducts%2Fvoice%2FConfiguring_Optional.pdfei=TjGtUMuDD86E0QHPpYCQDAusg=AFQjCNGL4uuttNHorfaTnTGcqxCQAZrwCQsig2=-HbRXBZJR1nqEtT0VmYq1A On Thu, Nov 15, 2012 at 6:59 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: At present I have two hardware identically freepbx/asterisk boxes. The mysql db on one is slaved to the other and all config files are rsync'd once every 24 hours (we have few configuration changes). We use Polycom 321/331/550/650 phones, and I notice that these phones can be configured with two SIP servers. Would the simplest approach to failover be to just configure my primary asterisk server as the first SIP server and my backup as the second? Kind Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware 4.0.1 and paging
It appears you need the info= if the string you are using is enclosed in angle brackets. Alert-Info: fooworks Alert-Info:foo does not work Alert-Info:info=foo works On Wed, Feb 15, 2012 at 2:09 PM, Mike l...@net-wall.com wrote: With Polycom firmware 4.0.1b? I have 1.8, one of the latest can`t remember which is installed on that server. Maybe the fact that my alert info has two words, and isn`t parsed correctly by Polycom...? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Wednesday, February 15, 2012 10:20 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging Which version of asterisk are you using? I just have this in 1.4 and it works fine: SIPAddHeader(Alert-Info: intercom); -Dave On 02/14/2012 08:10 PM, Mike wrote: In case anybody was following this thread, or someone Googles it in the future, here is the solution: This worked fine with Polycom firmware 3.3x: exten = s,n,SIPAddHeader(Alert-Info:Ring Answer) For firmware 4.0+, apparently I needed to add info=, i.e.: exten = s,n,SIPAddHeader(Alert-Info: info=Ring Answer) Simple, yet quite obscure (for me at least). Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, February 13, 2012 10:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging Thanks Dave, it at least gives me hope that my efforts aren`t wasted. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Monday, February 13, 2012 9:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging On 02/10/2012 05:30 PM, Mike wrote: Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? I would second the others suggestions about rewriting the configs. Polycom made extensive changes between 3.2 and 3.3, and I think they made a fair number of changes between 3.3 and 4.0. I have two phones that I've upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I believe I have auto answer working as you describe. Here's the pertinent snippet from my config: polycomConfig voIpProt voIpProt.SIP voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.1.class=ringAutoAnswer voIpProt.SIP.alertInfo.1.value=intercom voIpProt.SIP.alertInfo.2.class=ringAnswerMute voIpProt.SIP.alertInfo.2.value=page voIpProt.SIP.alertInfo.3.class=autoAnswer voIpProt.SIP.alertInfo.3.value=silentanswer /voIpProt.SIP.alertInfo /voIpProt.SIP /voIpProt /polycomConfig I have also added anse.rt section to adjust the ringer and timeouts for these ring tones. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call pickup on Asterisk 1.8 and Polycom IP550s?
It sounds like the phone is not getting enough info to do a directed pickup, have you turned on NotifyCID in sip.conf? If that does'nt work try using the extended BLF stuff (described here http://www.excaliburtech.net/archives/147 and here http://www.voip-info.org/wiki/view/Asterisk+presence) gordu On Thu, Dec 15, 2011 at 12:10 PM, Justin Sherrill justin.sherr...@americanrocksalt.com wrote: This is one of those Is anyone else doing this?/Is anyone else seeing this? posts. We have an Asterisk 1.8.4 system, with Polycom IP550 phones running firmware 3.2.3. If someone on the 'buddy list' - the list of other extensions to watch - is called, the phone gets a NOTIFY event and displays a screen with the call information and a pickup softkey. However, if someone on that list is already on the phone and they get a second incoming call, the NOTIFY event comes in but the phone never displays the changed screen with the pickup button. It'll flash the light next to that extension, but that's it. Is anyone using a similar setup and seeing this? It's somewhat rare, but I have an office location where everyone there likes to pick up other people's calls, and they haven't been using a call queue like they oughta. Justin Sherrill - American Rock Salt P: 585-991-6825 F: 585-991-6925 C: 585-298-6826 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing problem with Polycom phones after SIP update
Does the phone show the line as registered? The little phone icon on the display should be solid for a registered line and just a outline for a unregistered line. Using wireshark to watch the SIP traffic is a easy way to ensure the REGISTER signally is complete. On Fri, Dec 16, 2011 at 1:02 PM, Marco Mooijekind marco.mooijek...@gmail.com wrote: Dear all, I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8. All worked well. After applying the new Polycom UC 4.0.1 software update to the phones I notice the following: When dialing an extension, either on- or off hook, the phone immediately displays SIP URL:... This does not allow me to enter a regular numeric extension. The Polycom admin manual states that the phone displays the SIP URL input message if the phone is not registered. This is strange since i do see the phones registering themselves in the Asterisk verbose logging. Anyone experiencing this problem , any tips! Thanks in advance! Marco Mooijekind. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycoms rebooting themselves
Take a look at the network traffic, things like arp storms etc. A lot of noise on the net can cause reboots. Even if you don't find anything try turning on the storm filter (if it is not on already), its in the Settings - Advanced- Administration - Network settings - Ethernet I think. g On Tue, Aug 30, 2011 at 2:20 PM, Mike Diehl mdi...@diehlnet.com wrote: On Tuesday 30 August 2011 3:14:50 pm Tim Nelson wrote: - Original Message - Well, we've taken the time to check out the wiring. It's only 3 years old and looks like the people who did it knew what they were doing. Nice work. Rebooting the cable modem, router, and switch didn't fix the problem. Also, we had an instance today where ALL of the phones went down within minutes of each other. The Internet connection was still active. Looks like more often than not, all of the phones die at the same time. Any other ideas? If they're all powered via PoE on the same switch, look to diagnosing the switch itself. Look for issues with heat (not enough cooling or circulation), or depending on the switch, you could be pulling too much power from the PoE module contained within. Does your switch's PoE module put out enough power for 'X' number of phones at 'Y' number of watts each? Either of these problems would lead to the switch shutting down or resetting the PoE module which causes your phone reboots. All of the phones are AC powered. Either via an injector or wall outlet; I don't remember which. Definitely NOT POE. -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Blf / Directed Pickup
Oliver Your problem is you have not turned on notifycid=yes in sip.conf. Back on June 28 in another thread you said With asterisk 1.6.1.18, I could make this work without setting notifycid=yes isn sip.conf. butyes that gets the monitored line to blink on an incoming call, but as you have discovered the phone will not do a directed pickup. This info is also available at http://www.voip-info.org/wiki/view/Asterisk+presence cheers gord On Wed, Jul 6, 2011 at 8:22 AM, Olivier oza_4...@yahoo.fr wrote: Using a Polycom 650 with 3.3.1, I could not have Directed Pickup working. More precisely, I configured the phone using call and attendant entries as described in this thread. Whenever a call comes in, BLF is blinking green. Pressing the associated key generate generates a general Call Pickup (*8), not a directed Call Pickup. Could you confirm this ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF
I missed one important parameter in my setup of BLF for polycom phones (at
least if you want to do one touch directed pickup)
In sip.conf add
notifycid=yes
the notifycid=yes causes asterisk to add a target uri = callID to the XML
of the SIP notify. Without this target uri the Polycom phone will not do a
directed pickup.
On Fri, Jun 17, 2011 at 2:17 PM, Gord Urquhart gord...@gmail.com wrote:
From http://www.voip-info.org/wiki/view/Asterisk+presence
Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With
SIP 3.2.X firmware (available on the Polycom download site) and Asterisk
1.6.1, Polycom phones now support a full featured BLF showing statuses of
Ringing, Inuse and Online and one touch directed call pickup.
On the asterisk side all that needs to be done is to add a hint to the
extension and enable directed pickup. Directed pickup is enabled by adding
the following lines to extensios.conf
exten = _*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten = _*8.,n,Pickup(${EXTEN:2}@PICKUPMARK)
On the phone side for each line that is going to be monitored add lines
like the following to the phone's cfg file.
attendant.reg=1
attendant.resourceList.1.address=sip:205@192.168.1.102
attendant.resourceList.1.label=205
attendant.resourceList.2.address=sip:217@192.168.1.102
attendant.resourceList.2.label=217
call.directedCallPickupMethod=legacy
call.directedCallPickupString=*8
feature.12.name=directed-call-pickup
feature.12.enabled=1
Assuming my server is at 192.168.1.102, this will add two BLF lines to the
phone for extensions 205 and 217. Calls incoming to those extensions will
show a blinking green led on the monitoring phone, pressing the hard key
will pick the call up, if it is answered elsewhere the led will change to
solid red. AFAIK this cannot be configured via the phones web gui, you must
use the cfg files. You can also use versions of Asterisk older than 1.6.1 if
you remove the restriction on what asterisk thinks Polycom phones can
handle. Look in chan_sip.c for
if (strstr(p-useragent, Polycom)) {
p-subscribed = XPIDF_XML;
and change that line to
p-subscribed = DIALOG_INFO_XML;
On Tue, Jun 14, 2011 at 8:36 AM, Jeff LaCoursiere j...@sunfone.comwrote:
Struggling with an IP650 and 7 IP335s this morning. I have the following
hints defined (courtesy of FreePBX 2.9):
extensions_additional.conf:**exten = 300,hint,SIP/300
extensions_additional.conf:**exten = 301,hint,SIP/301
extensions_additional.conf:**exten = 302,hint,SIP/302
extensions_additional.conf:**exten = 303,hint,SIP/303
extensions_additional.conf:**exten = 304,hint,SIP/304
extensions_additional.conf:**exten = 305,hint,SIP/305
extensions_additional.conf:**exten = 307,hint,SIP/307
extensions_additional.conf:**exten = 308,hint,SIP/308
extensions_additional.conf:**exten = 322,hint,SIP/322
extensions_additional.conf:**exten = 350,hint,SIP/350
extensions_additional.conf:**exten = 400,hint,SIP/400
The Polycoms are all pulling an XML directory via FTP where each extension
has BW (Buddy Watch) set to 1:
item
lnMehra/ln
fnRay/fn
ct301/ct
sd101/sd
bw1/bw
/item
This all actually works fine, and from the reception phone (the 650) I can
see the status of all the extensions, and if I dig into some menus on the
335 I can see status as well. So I would expect that core show hints
would show '8' for all extensions, but it doesn't:
artha*CLI core show hints
-= Registered Asterisk Dial Plan Hints =-
300@ext-local : SIP/300 State:Idle
Watchers 7
301@ext-local : SIP/301 State:Idle
Watchers 8
302@ext-local : SIP/302 State:Idle
Watchers 8
303@ext-local : SIP/303 State:Idle
Watchers 8
304@ext-local : SIP/304 State:InUse
Watchers 8
305@ext-local : SIP/305 State:Idle
Watchers 7
307@ext-local : SIP/307 State:Idle
Watchers 1
308@ext-local : SIP/308 State:Idle
Watchers 7
350@ext-local : SIP/350 State:Idle
Watchers 1
400@ext-local : SIP/400 State:InUse
Watchers 7
- 11 hints registered
Something seems broken here. And the 650 seems to lose its hint for a
phone once in a while, and report it as unreachable, even though it can
easily make and receive calls from it.
Am I tilting at windmills? Is this really unstable or has someone made it
work solidly?
Thanks!
--
Jeff LaCoursiere
SunFone
340-715
Re: [asterisk-users] Polycom BLF
From http://www.voip-info.org/wiki/view/Asterisk+presence
Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With
SIP 3.2.X firmware (available on the Polycom download site) and Asterisk
1.6.1, Polycom phones now support a full featured BLF showing statuses of
Ringing, Inuse and Online and one touch directed call pickup.
On the asterisk side all that needs to be done is to add a hint to the
extension and enable directed pickup. Directed pickup is enabled by adding
the following lines to extensios.conf
exten = _*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten = _*8.,n,Pickup(${EXTEN:2}@PICKUPMARK)
On the phone side for each line that is going to be monitored add lines like
the following to the phone's cfg file.
attendant.reg=1
attendant.resourceList.1.address=sip:205@192.168.1.102
attendant.resourceList.1.label=205
attendant.resourceList.2.address=sip:217@192.168.1.102
attendant.resourceList.2.label=217
call.directedCallPickupMethod=legacy
call.directedCallPickupString=*8
feature.12.name=directed-call-pickup
feature.12.enabled=1
Assuming my server is at 192.168.1.102, this will add two BLF lines to the
phone for extensions 205 and 217. Calls incoming to those extensions will
show a blinking green led on the monitoring phone, pressing the hard key
will pick the call up, if it is answered elsewhere the led will change to
solid red. AFAIK this cannot be configured via the phones web gui, you must
use the cfg files. You can also use versions of Asterisk older than 1.6.1 if
you remove the restriction on what asterisk thinks Polycom phones can
handle. Look in chan_sip.c for
if (strstr(p-useragent, Polycom)) {
p-subscribed = XPIDF_XML;
and change that line to
p-subscribed = DIALOG_INFO_XML;
On Tue, Jun 14, 2011 at 8:36 AM, Jeff LaCoursiere j...@sunfone.com wrote:
Struggling with an IP650 and 7 IP335s this morning. I have the following
hints defined (courtesy of FreePBX 2.9):
extensions_additional.conf:**exten = 300,hint,SIP/300
extensions_additional.conf:**exten = 301,hint,SIP/301
extensions_additional.conf:**exten = 302,hint,SIP/302
extensions_additional.conf:**exten = 303,hint,SIP/303
extensions_additional.conf:**exten = 304,hint,SIP/304
extensions_additional.conf:**exten = 305,hint,SIP/305
extensions_additional.conf:**exten = 307,hint,SIP/307
extensions_additional.conf:**exten = 308,hint,SIP/308
extensions_additional.conf:**exten = 322,hint,SIP/322
extensions_additional.conf:**exten = 350,hint,SIP/350
extensions_additional.conf:**exten = 400,hint,SIP/400
The Polycoms are all pulling an XML directory via FTP where each extension
has BW (Buddy Watch) set to 1:
item
lnMehra/ln
fnRay/fn
ct301/ct
sd101/sd
bw1/bw
/item
This all actually works fine, and from the reception phone (the 650) I can
see the status of all the extensions, and if I dig into some menus on the
335 I can see status as well. So I would expect that core show hints
would show '8' for all extensions, but it doesn't:
artha*CLI core show hints
-= Registered Asterisk Dial Plan Hints =-
300@ext-local : SIP/300 State:Idle
Watchers 7
301@ext-local : SIP/301 State:Idle
Watchers 8
302@ext-local : SIP/302 State:Idle
Watchers 8
303@ext-local : SIP/303 State:Idle
Watchers 8
304@ext-local : SIP/304 State:InUse
Watchers 8
305@ext-local : SIP/305 State:Idle
Watchers 7
307@ext-local : SIP/307 State:Idle
Watchers 1
308@ext-local : SIP/308 State:Idle
Watchers 7
350@ext-local : SIP/350 State:Idle
Watchers 1
400@ext-local : SIP/400 State:InUse
Watchers 7
- 11 hints registered
Something seems broken here. And the 650 seems to lose its hint for a
phone once in a while, and report it as unreachable, even though it can
easily make and receive calls from it.
Am I tilting at windmills? Is this really unstable or has someone made it
work solidly?
Thanks!
--
Jeff LaCoursiere
SunFone
340-715-7600 x222
j...@sunfone.com
--
__**__**_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Polycom Blf / Directed Pickup
After someone sent me an email saying his directed pickup did not work. I
realized I forgot to mention that directed pickup needs to be enabled in
extensions.conf i.e. add the following
exten=_*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten = _*8.,n,Pickup(${EXTEN:2}@PICKUPMARK)
On Mon, Jan 17, 2011 at 4:00 PM, Gord Urquhart gord...@gmail.com wrote:
With SIP 3.2.X firmware (available on the Polycom download site) and
Asterisk 1.6.1, Polycom phones now support a full featured BLF showing
statuses of Ringing, Inuse and Online and one touch directed call pickup.
On the asterisk side all that needs to be done is to add a hint to the
extension. On the phone side for each line that is going to be monitored add
lines like the following to the phone's cfg file.
attendant.reg=1
attendant.resourceList.1.address=sip:205@192.168.1.102sip%3A205@192.168.1.102
attendant.resourceList.1.label=205
attendant.resourceList.2.address=sip:217@192.168.1.102sip%3A217@192.168.1.102
attendant.resourceList.2.label=217
Following 4 lines added Sept/10
call.directedCallPickupMethod=legacy
call.directedCallPickupString=*8
feature.12.name=directed-call-pickup
feature.12.enabled=1
Assuming my server is at 192.168.1.102, this will add two BLF lines to the
phone for extensions 205 and 217. Calls incoming to those extensions will
show a blinking green led on the monitoring phone, pressing the hard key
will pick the call up, if it is answered elsewhere the led will change to
solid red. AFAIK this cannot be configured via the phones web gui, you must
use the cfg files. You can also use versions of Asterisk older than 1.6.1 if
you remove the restriction on what asterisk thinks Polycom phones can
handle. Look in chan_sip.c for
if (strstr(p-useragent, Polycom)) {
p-subscribed = XPIDF_XML;
and change that line to
p-subscribed = DIALOG_INFO_XML;
cheers
gord
On Thu, Jan 13, 2011 at 4:26 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Thanks! Blf is working now. I forgot I had to set set subscribecontext.
When a phone is ringing, the blf light is solid red and the icon is a (/)
type icon indicating unavailable. I'm also interested in directed pickup.
I set up the following:
call.directedCallPickupString=*6 call.directedCallPickupMethod=legacy
Hitting the button next to the contact will speed dial the contact instead
of pick up the ringing call.
On 01/13/2011 10:54 AM, Sebastien Thomas wrote:
Ok, that looks good.
We use FreePBX, and I know I had to modify a couple Asterisk files to
get the BLF working ... here are some of my mods but may also be used
for FOP2 (I dont recall which go for BLF and which go FOP2).
vi /etc/asterisk/sip_registrations_custom.conf
allowsubscribe=yes
vi /etc/asterisk/sip_custom.conf
callevents=yes
notifyringing=yes
limitonpeers=yes
I also override some of the sip.cfg settings in the polycom dir with:
feature
feature.1.enabled=1
feature.9.enabled=0
feature.18.enabled=1
/
pres
pres.reg=1
pres.idleSoftkeys=0
/
---
Sebastien Thomas
Amplisys Inc. - Digital Telephony Integration Specialists
T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS
*** Need help? Contact supp...@amplisys.ca mailto:supp...@amplisys.ca
***
On 2011-01-13, at 10:29 AM, Mark Murawski wrote:
Yeah... My directory looks like this:
directory
item_list
item
ln6288/lnfn/fnct6288/ctsd1/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6208/lnfn/fnct6208/ctsd2/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6234/lnfn/fnct6234/ctsd3/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6205/lnfn/fnct6205/ctsd4/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6231/lnfn/fnct6231/ctsd5/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
/item_list
/directory
On 01/13/2011 10:20 AM, Sebastien Thomas wrote:
Is the buddy watch tag activated in yourmac-directory.xml file
?bw1/bw
item
lbSebastien/lb
fnSebastien/fn
lnThomas/ln
ct222/ct
sd1/sd
bw1/bw
/item
---
Sebastien Thomas
Amplisys Inc. - Digital Telephony Integration Specialists
T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS
On 2011-01-13, at 1:32 AM, Mark Murawski wrote:
Would anyone happen to have some examples of polycom configs,
specifically the 650 with sidecar for blf.
I have the asterisk side all configured since I've set up blf with
other types of phones, but I'm missing the polycom side.
I've put together amac-directory.xml, and the sidecar now lists
numbers as speed dials but does not subscribe to blf.
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Re: [asterisk-users] Polycom Blf / Directed Pickup
With SIP 3.2.X firmware (available on the Polycom download site) and
Asterisk 1.6.1, Polycom phones now support a full featured BLF showing
statuses of Ringing, Inuse and Online and one touch directed call pickup.
On the asterisk side all that needs to be done is to add a hint to the
extension. On the phone side for each line that is going to be monitored add
lines like the following to the phone's cfg file.
attendant.reg=1
attendant.resourceList.1.address=sip:205@192.168.1.102sip%3A205@192.168.1.102
attendant.resourceList.1.label=205
attendant.resourceList.2.address=sip:217@192.168.1.102sip%3A217@192.168.1.102
attendant.resourceList.2.label=217
Following 4 lines added Sept/10
call.directedCallPickupMethod=legacy
call.directedCallPickupString=*8
feature.12.name=directed-call-pickup
feature.12.enabled=1
Assuming my server is at 192.168.1.102, this will add two BLF lines to the
phone for extensions 205 and 217. Calls incoming to those extensions will
show a blinking green led on the monitoring phone, pressing the hard key
will pick the call up, if it is answered elsewhere the led will change to
solid red. AFAIK this cannot be configured via the phones web gui, you must
use the cfg files. You can also use versions of Asterisk older than 1.6.1 if
you remove the restriction on what asterisk thinks Polycom phones can
handle. Look in chan_sip.c for
if (strstr(p-useragent, Polycom)) {
p-subscribed = XPIDF_XML;
and change that line to
p-subscribed = DIALOG_INFO_XML;
cheers
gord
On Thu, Jan 13, 2011 at 4:26 PM, Mark Murawski
markm-li...@intellasoft.netwrote:
Thanks! Blf is working now. I forgot I had to set set subscribecontext.
When a phone is ringing, the blf light is solid red and the icon is a (/)
type icon indicating unavailable. I'm also interested in directed pickup.
I set up the following:
call.directedCallPickupString=*6 call.directedCallPickupMethod=legacy
Hitting the button next to the contact will speed dial the contact instead
of pick up the ringing call.
On 01/13/2011 10:54 AM, Sebastien Thomas wrote:
Ok, that looks good.
We use FreePBX, and I know I had to modify a couple Asterisk files to
get the BLF working ... here are some of my mods but may also be used
for FOP2 (I dont recall which go for BLF and which go FOP2).
vi /etc/asterisk/sip_registrations_custom.conf
allowsubscribe=yes
vi /etc/asterisk/sip_custom.conf
callevents=yes
notifyringing=yes
limitonpeers=yes
I also override some of the sip.cfg settings in the polycom dir with:
feature
feature.1.enabled=1
feature.9.enabled=0
feature.18.enabled=1
/
pres
pres.reg=1
pres.idleSoftkeys=0
/
---
Sebastien Thomas
Amplisys Inc. - Digital Telephony Integration Specialists
T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS
*** Need help? Contact supp...@amplisys.ca mailto:supp...@amplisys.ca
***
On 2011-01-13, at 10:29 AM, Mark Murawski wrote:
Yeah... My directory looks like this:
directory
item_list
item
ln6288/lnfn/fnct6288/ctsd1/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6208/lnfn/fnct6208/ctsd2/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6234/lnfn/fnct6234/ctsd3/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6205/lnfn/fnct6205/ctsd4/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6231/lnfn/fnct6231/ctsd5/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
/item_list
/directory
On 01/13/2011 10:20 AM, Sebastien Thomas wrote:
Is the buddy watch tag activated in yourmac-directory.xml file
?bw1/bw
item
lbSebastien/lb
fnSebastien/fn
lnThomas/ln
ct222/ct
sd1/sd
bw1/bw
/item
---
Sebastien Thomas
Amplisys Inc. - Digital Telephony Integration Specialists
T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS
On 2011-01-13, at 1:32 AM, Mark Murawski wrote:
Would anyone happen to have some examples of polycom configs,
specifically the 650 with sidecar for blf.
I have the asterisk side all configured since I've set up blf with
other types of phones, but I'm missing the polycom side.
I've put together amac-directory.xml, and the sidecar now lists
numbers as speed dials but does not subscribe to blf.
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Re: [asterisk-users] Polycom Park by EFK
According to the Admin guide EFK is not supported on 501s This capability applies to the SoundPoint IP 32x/33x, 450, 550, 560, 650, and 670 desktop phones, the SoundStation IP 5000, 6000, and 7000 conference phones, and Polycom VVX 1500 business media phones On Fri, Dec 3, 2010 at 5:02 PM, Andrew Joakimsen joakim...@gmail.comwrote: Has anyone gotten one-touch call parking to work on Polycom phones via the Enhanced Feature Keys feature working? I've looked at various examples, they appear correct, but the phones (501, 3.1.x firmware) show the Park button while in a call but this does not actually cause the call to be parked. Doing a SIP debug, I don't see that anything is transmitted as a result of pressing the call park key. My understanding of the below configuration is it should cause the DTMF sequence #70 to be sent across the SIP channel -- but it isn't. efk version efk.version=2 / efklist efk.efklist.1.mname=callpark efk.efklist.1.status=1 efk.efklist.1.label=Call Park efk.efklist.1.use.active=1 efk.efklist.1.action.string=#70 efk.efklist.2.mname=blindxfer efk.efklist.2.status=1 efk.efklist.2.label=Blind XFer efk.efklist.2.use.active=1 efk.efklist.2.action.string=$P1N10$$Trefer$ efk.efklist.3.mname=daynight efk.efklist.3.status=1 efk.efklist.3.label=Day Night 1 efk.efklist.3.use.active=1 efk.efklist.3.action.string=*281 efk.efklist.4.mname=pageall efk.efklist.4.status=1 efk.efklist.4.label=PageAll efk.efklist.4.use.active=1 efk.efklist.4.action.string=800 / efkprompt efk.efkprompt.1.status=1 efk.efkprompt.1.label=Extension: efk.efkprompt.1.userfeedback=visible efk.efkprompt.1.type=numeric efk.efkprompt.2.status=1 efk.efkprompt.2.label=PIN Code: efk.efkprompt.2.userfeedback=masked efk.efkprompt.2.type=numeric efk.efkprompt.3.status=1 efk.efkprompt.3.label=Password: efk.efkprompt.3.userfeedback=masked efk.efkprompt.3.type=numeric efk.efkprompt.4.status=1 efk.efkprompt.4.label=Conf ID: efk.efkprompt.4.userfeedback=visible efk.efkprompt.4.type=numeric efk.efkprompt.5.status=1 efk.efkprompt.5.label=Extension: efk.efkprompt.5.userfeedback=visible efk.efkprompt.5.type=numeric / /efk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd Issue With Polycom Phones]
The phone is only making one call, notice the call-id did not change. The second INVITE is sent in responce to a 401 Authentication Required. The 401 will contain the necessary authentication information for the phone to use to build the Authorization header that it inserts in the second invite. THe mechanism uses a shared secret (the reg.X.auth.userId and reg.X.auth.password in the polycom cfg file, and the secret=X and the userID(I think thats what its called) in the asterisk config files). If you have other phones that are not doing this second invite I would bet its because on the asterisk side you have not configured them to use a secret. -- Thanks for the tip, I did just that, and now I am more confused. It does appear as though there is just one call ID (if my assumption that the tag= determines the call. The first time it sends like this: --- SIP read from UDP:x.x.x.x:5060 --- INVITE sip:3...@y.y.y.y;user=phone SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKe3e15c76913F8BDD From: 3271 sip:3271@ y.y.y.y sip:3...@y.y.y.y;tag=990EE6B0-8E3DCEA7 To: sip:3261@ y.y.y.y;user=phone sip:3...@y.y.y.y;user=phone CSeq: 1 INVITE Call-ID: 96a1fe9c-88f06c73-7e209...@x.x.x.x Contact: sip:3271@ x.x.x.x:5060 sip:3...@x.x.x.x:5060 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 461 v=0 o=- 1271881915 1271881915 IN IP4 x.x.x.x s=Polycom IP Phone c=IN IP4 x.x.x.x t=0 0 a=sendrecv m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 Asterisk responds with a SIP/2.0 401 Unauthorized, the phone then comes back with this: --- SIP read from UDP:x.x.x.x:5060 --- INVITE sip:3261@ y.y.y.y;user=phone SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK6f7a6692AF94008 From: 3271 sip:3271@ y.y.y.y sip:3...@y.y.y.y;tag=990EE6B0-8E3DCEA7 To: sip:3261@ y.y.y.y;user=phone sip:3...@y.y.y.y;user=phone CSeq: 2 INVITE Call-ID: 96a1fe9c-88f06c73-7e209322@ x.x.x.x Contact: sip:3271@ x.x.x.x:5060 sip:3...@x.x.x.x:5060 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username=3271, realm=asterisk, nonce=393a1b1f, uri=sip:3261@ y.y.y.y;user=phone sip:3...@y.y.y.y;user=phone, response=c8223e261c252c12172982ee661ad307, algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 461 v=0 o=- 1271881915 1271881915 IN IP4 x.x.x.x s=Polycom IP Phone c=IN IP4 x.x.x.x t=0 0 a=sendrecv m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 The difference is that the CSeq is now 2 and the following line is added: Authorization: Digest username=3271, realm=asterisk, nonce=393a1b1f, uri=sip:3...@y.y.y.y;user=phone sip:3...@y.y.y.y;user=phone, response=c8223e261c252c12172982ee661ad307, algorithm=MD5 So maybe I do have an issue in Asterisk, okay probably. Any clues as to how to debug? Let me know if need to post more information. Thanks. -Jay -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Brady Sent: Tuesday, April 20, 2010 4:57 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Odd Issue With Polycom Phones -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
