Re: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA Support
You have to set the param to sylantro to get that mode. Or configure your phone to use broadsoft. On Feb 17, 2010 7:54 AM, "Yehavi Bourvine" <yehavi.bourvine@gmail.com> wrote: I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version 1.0.5-20100216-0400 (16659M)), and shared apearance stopped working... We have Polycom phones and I use the same configuration for both versions. With the old version I see that after the phone registers with FreeSwitch, the server subscribes to the phone for the extension; with the latest version this does not happen. Furthermore, the table sip_shared_appearance_dialogs is empty. I don't find anything I can change on the phone config (the line is already set to shared). Here is the relevant config from the sip profile: <param name="manage-presence" value="passive"/> (set to TRUE only on one profile). <param name="manage-shared-appearance" value="true"/> <param name="dbname" value="share_presence"/> <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il "/> Any idea where to look? Thanks, __Yehavi: _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Tony, do you mean in the internal.xml under /usr/local/freeswitch/local/sip_profiles we should set <param name="manage-shared- appearance" value="sylantro"/> to get the SCA mode working properly? Please confirm Thanks, Chris On Wed, Feb 17, 2010 at 8:57 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You have to set the param to sylantro to get that mode. Or configure your phone to use broadsoft.
On Feb 17, 2010 7:54 AM, "Yehavi Bourvine" <yehavi.bourvine@gmail.com> wrote:
I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version 1.0.5-20100216-0400 (16659M)), and shared apearance stopped working...
We have Polycom phones and I use the same configuration for both versions. With the old version I see that after the phone registers with FreeSwitch, the server subscribes to the phone for the extension; with the latest version this does not happen. Furthermore, the table sip_shared_appearance_dialogs is empty.
I don't find anything I can change on the phone config (the line is already set to shared).
Here is the relevant config from the sip profile:
<param name="manage-presence" value="passive"/> (set to TRUE only on one profile). <param name="manage-shared-appearance" value="true"/>
<param name="dbname" value="share_presence"/> <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il "/> Any idea where to look? Thanks, __Yehavi:
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Hello Anthony, Since Polycom has no place to define the server type I've set manage-shared-appearance="sylantro" and have some progress. Now I see that both the server and the phone subscribe to each other(the server subscribes twice), but the last NOTIFY from the phone to the server gets rejected with "Call does not exist". On the console I get the following error message: [ERR] sofia_presence.c:2135 Gateway information missing. Thanks, __Yehavi: 2010/2/17 Anthony Minessale <anthony.minessale@gmail.com>
You have to set the param to sylantro to get that mode. Or configure your phone to use broadsoft.
On Feb 17, 2010 7:54 AM, "Yehavi Bourvine" <yehavi.bourvine@gmail.com> wrote:
I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version 1.0.5-20100216-0400 (16659M)), and shared apearance stopped working...
We have Polycom phones and I use the same configuration for both versions. With the old version I see that after the phone registers with FreeSwitch, the server subscribes to the phone for the extension; with the latest version this does not happen. Furthermore, the table sip_shared_appearance_dialogs is empty.
I don't find anything I can change on the phone config (the line is already set to shared).
Here is the relevant config from the sip profile:
<param name="manage-presence" value="passive"/> (set to TRUE only on one profile). <param name="manage-shared-appearance" value="true"/>
<param name="dbname" value="share_presence"/> <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il "/> Any idea where to look? Thanks, __Yehavi:
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Step 1. Enable manage-shared-appearance=true Step 2. Now in the phone's config Configure the phone as usually, set the line shared and DO NOT set the third party name. Step 3. Reboot It should work. I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine. The gateway info missing is a gateway you have configured getting a notify. It has nothing to do with SCA. /b On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
.
Hello Brian, I'll try what you suggest on Friday, and if it works I will document it in the wiki under the Polycom page I already wrote. Thanks, __Yehavi: 2010/2/17 Brian West <brian@freeswitch.org>
Step 1. Enable manage-shared-appearance=true
Step 2. Now in the phone's config Configure the phone as usually, set the line shared and DO NOT set the third party name.
Step 3. Reboot
It should work.
I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine.
The gateway info missing is a gateway you have configured getting a notify. It has nothing to do with SCA.
/b
On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
.
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Thanks Brian. It now works better, but not fully (using 16659M). What happens is: - When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use". - When one of the Polycom phones initiates a call - all is ok: - The other side sees that the extension is in use. - When it is put to hold all phones who share this extension see it and can pick the call. - When a call arrives, both ring; the one that did not answer gets only a cancel mesage *without *any further notification that the extension is in use by the other phone. Thanks! __Yehavi: 2010/2/17 Brian West <brian@freeswitch.org>
Step 1. Enable manage-shared-appearance=true
Step 2. Now in the phone's config Configure the phone as usually, set the line shared and DO NOT set the third party name.
Step 3. Reboot
It should work.
I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine.
The gateway info missing is a gateway you have configured getting a notify. It has nothing to do with SCA.
/b
On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
That last bit is wrong... I need sip traces. /b On Feb 18, 2010, at 11:01 PM, Yehavi Bourvine wrote:
Thanks Brian. It now works better, but not fully (using 16659M).
What happens is: When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use". When one of the Polycom phones initiates a call - all is ok: The other side sees that the extension is in use. When it is put to hold all phones who share this extension see it and can pick the call. When a call arrives, both ring; the one that did not answer gets only a cancel mesage without any further notification that the extension is in use by the other phone. Thanks! __Yehavi:
When a call arrives, both ring; the one that did not answer gets only a cancel mesage without any further notification that the extension is in use by the other phone.
These are the same exact symptoms I posted about earlier this week, with the Cisco SPA-5xx series phones. I still have yet to figure out why this is happening, if you find out what's going on, please post back the solution, I'd like to know the resolution. Thanks, Gabe On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Thanks Brian. It now works better, but not fully (using 16659M).
What happens is:
When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use". When one of the Polycom phones initiates a call - all is ok:
The other side sees that the extension is in use. When it is put to hold all phones who share this extension see it and can pick the call.
Thanks! __Yehavi:
2010/2/17 Brian West <brian@freeswitch.org>
Step 1. Enable manage-shared-appearance=true
Step 2. Now in the phone's config Configure the phone as usually, set the line shared and DO NOT set the third party name.
Step 3. Reboot
It should work.
I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine.
The gateway info missing is a gateway you have configured getting a notify. It has nothing to do with SCA.
/b
On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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Hello Gabe, As you can see - Brian is actively investigating it, so you can expect for some fix soon... Regards, __Yehavi: 2010/2/19 Gabriel Kuri <gkuri@ieee.org>
When a call arrives, both ring; the one that did not answer gets only a cancel mesage without any further notification that the extension is in use by the other phone.
These are the same exact symptoms I posted about earlier this week, with the Cisco SPA-5xx series phones. I still have yet to figure out why this is happening, if you find out what's going on, please post back the solution, I'd like to know the resolution.
Thanks, Gabe
On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Thanks Brian. It now works better, but not fully (using 16659M).
What happens is:
When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use". When one of the Polycom phones initiates a call - all is ok:
The other side sees that the extension is in use. When it is put to hold all phones who share this extension see it and can pick the call.
Thanks! __Yehavi:
2010/2/17 Brian West <brian@freeswitch.org>
Step 1. Enable manage-shared-appearance=true
Step 2. Now in the phone's config Configure the phone as usually, set
the
line shared and DO NOT set the third party name.
Step 3. Reboot
It should work.
I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine.
The gateway info missing is a gateway you have configured getting a notify. It has nothing to do with SCA.
/b
On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
.
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If this issue is not already on jira could you please make sure it gets added? Mike On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello Gabe,
As you can see - Brian is actively investigating it, so you can expect for some fix soon...
Regards, __Yehavi:
2010/2/19 Gabriel Kuri <gkuri@ieee.org>
When a call arrives, both ring; the one that did not answer gets only a cancel mesage without any further notification that the extension is in use by the other phone.
These are the same exact symptoms I posted about earlier this week, with the Cisco SPA-5xx series phones. I still have yet to figure out why this is happening, if you find out what's going on, please post back the solution, I'd like to know the resolution.
Thanks, Gabe
On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Thanks Brian. It now works better, but not fully (using 16659M).
What happens is:
When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use". When one of the Polycom phones initiates a call - all is ok:
The other side sees that the extension is in use. When it is put to hold all phones who share this extension see it and can pick the call.
Thanks! __Yehavi:
2010/2/17 Brian West <brian@freeswitch.org>
Step 1. Enable manage-shared-appearance=true
Step 2. Now in the phone's config Configure the phone as usually,
set the
line shared and DO NOT set the third party name.
Step 3. Reboot
It should work.
I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine.
The gateway info missing is a gateway you have configured getting a notify. It has nothing to do with SCA.
/b
On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ freeswitch-users http://www.freeswitch.org
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A jira issue has been created: *MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61> . Thanks, __Yehavi: 2010/2/19 Michael Jerris <mike@jerris.com>
If this issue is not already on jira could you please make sure it gets added?
Mike
On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello Gabe,
As you can see - Brian is actively investigating it, so you can expect for some fix soon...
Regards, __Yehavi:
2010/2/19 Gabriel Kuri <gkuri@ieee.org>
When a call arrives, both ring; the one that did not answer gets only a cancel mesage without any further notification that the extension is in use by the other phone.
These are the same exact symptoms I posted about earlier this week, with the Cisco SPA-5xx series phones. I still have yet to figure out why this is happening, if you find out what's going on, please post back the solution, I'd like to know the resolution.
Thanks, Gabe
On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Thanks Brian. It now works better, but not fully (using 16659M).
What happens is:
When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use". When one of the Polycom phones initiates a call - all is ok:
The other side sees that the extension is in use. When it is put to hold all phones who share this extension see it and can pick the call.
Thanks! __Yehavi:
2010/2/17 Brian West <brian@freeswitch.org>
Step 1. Enable manage-shared-appearance=true
Step 2. Now in the phone's config Configure the phone as usually, set
the
line shared and DO NOT set the third party name.
Step 3. Reboot
It should work.
I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine.
The gateway info missing is a gateway you have configured getting a notify. It has nothing to do with SCA.
/b
On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
.
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go see my comments on that bug note. be prepared to give us ssh access and call or irc so we can can see you reproducing it. If you are not on the latest firmware on all the phones, we will not continue with this process. On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com>wrote:
A jira issue has been created: *MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61> .
Thanks, __Yehavi:
2010/2/19 Michael Jerris <mike@jerris.com>
If this issue is not already on jira could you please make sure it gets added?
Mike
On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello Gabe,
As you can see - Brian is actively investigating it, so you can expect for some fix soon...
Regards, __Yehavi:
2010/2/19 Gabriel Kuri < <gkuri@ieee.org>gkuri@ieee.org>
When a call arrives, both ring; the one that did not answer gets only a cancel mesage without any further notification that the extension is in use by the other phone.
These are the same exact symptoms I posted about earlier this week, with the Cisco SPA-5xx series phones. I still have yet to figure out why this is happening, if you find out what's going on, please post back the solution, I'd like to know the resolution.
Thanks, Gabe
On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine < <yehavi.bourvine@gmail.com>yehavi.bourvine@gmail.com> wrote:
Thanks Brian. It now works better, but not fully (using 16659M).
What happens is:
When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use". When one of the Polycom phones initiates a call - all is ok:
The other side sees that the extension is in use. When it is put to hold all phones who share this extension see it and can pick the call.
Thanks! __Yehavi:
2010/2/17 Brian West < <brian@freeswitch.org>brian@freeswitch.org>
Step 1. Enable manage-shared-appearance=true
Step 2. Now in the phone's config Configure the phone as usually, set
the
line shared and DO NOT set the third party name.
Step 3. Reboot
It should work.
I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine.
The gateway info missing is a gateway you have configured getting a notify. It has nothing to do with SCA.
/b
On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
I see now that Polycom released newer versions of firmware for the phones recently. On Sunday's mornning I'll download them and retest with the latest FreeSwitch snapshot. __Yehavi: 2010/2/19 Anthony Minessale <anthony.minessale@gmail.com>
go see my comments on that bug note. be prepared to give us ssh access and call or irc so we can can see you reproducing it.
If you are not on the latest firmware on all the phones, we will not continue with this process.
On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine < yehavi.bourvine@gmail.com> wrote:
A jira issue has been created: *MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61> .
Thanks, __Yehavi:
2010/2/19 Michael Jerris <mike@jerris.com>
If this issue is not already on jira could you please make sure it gets added?
Mike
On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello Gabe,
As you can see - Brian is actively investigating it, so you can expect for some fix soon...
Regards, __Yehavi:
2010/2/19 Gabriel Kuri < <gkuri@ieee.org>gkuri@ieee.org>
When a call arrives, both ring; the one that did not answer gets only a cancel mesage without any further notification that the extension is in use by the other phone.
These are the same exact symptoms I posted about earlier this week, with the Cisco SPA-5xx series phones. I still have yet to figure out why this is happening, if you find out what's going on, please post back the solution, I'd like to know the resolution.
Thanks, Gabe
On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine < <yehavi.bourvine@gmail.com>yehavi.bourvine@gmail.com> wrote:
Thanks Brian. It now works better, but not fully (using 16659M).
What happens is:
When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use". When one of the Polycom phones initiates a call - all is ok:
The other side sees that the extension is in use. When it is put to hold all phones who share this extension see it and can pick the call.
Thanks! __Yehavi:
2010/2/17 Brian West < <brian@freeswitch.org>brian@freeswitch.org>
Step 1. Enable manage-shared-appearance=true
Step 2. Now in the phone's config Configure the phone as usually, set
the
line shared and DO NOT set the third party name.
Step 3. Reboot
It should work.
I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine.
The gateway info missing is a gateway you have configured getting a notify. It has nothing to do with SCA.
/b
On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
> .
_______________________________________________ FreeSWITCH-users mailing list <FreeSWITCH-users@lists.freeswitch.org> FreeSWITCH-users@lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://lists.freeswitch.org/mailman/options/freeswitch-users <http://www.freeswitch.org/>http://www.freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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Hello all, The problem is solved, at least for my Polycoms. The solution is to have "domain" variable (in vars.xml) set to the IP address of the profile that is used for phones registration (the default is ok if you have only one interface). Define the phone's registration and proxy servers to that address. You must use IP addresses and not DNS names. Thanks to Anthony for his helpfull tips! __Yehavi: 2010/2/19 Anthony Minessale <anthony.minessale@gmail.com>
go see my comments on that bug note. be prepared to give us ssh access and call or irc so we can can see you reproducing it.
If you are not on the latest firmware on all the phones, we will not continue with this process.
On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine < yehavi.bourvine@gmail.com> wrote:
A jira issue has been created: *MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61> .
Thanks, __Yehavi:
2010/2/19 Michael Jerris <mike@jerris.com>
If this issue is not already on jira could you please make sure it gets added?
Mike
On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello Gabe,
As you can see - Brian is actively investigating it, so you can expect for some fix soon...
Regards, __Yehavi:
2010/2/19 Gabriel Kuri < <gkuri@ieee.org>gkuri@ieee.org>
When a call arrives, both ring; the one that did not answer gets only a cancel mesage without any further notification that the extension is in use by the other phone.
These are the same exact symptoms I posted about earlier this week, with the Cisco SPA-5xx series phones. I still have yet to figure out why this is happening, if you find out what's going on, please post back the solution, I'd like to know the resolution.
Thanks, Gabe
On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine < <yehavi.bourvine@gmail.com>yehavi.bourvine@gmail.com> wrote:
Thanks Brian. It now works better, but not fully (using 16659M).
What happens is:
When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use". When one of the Polycom phones initiates a call - all is ok:
The other side sees that the extension is in use. When it is put to hold all phones who share this extension see it and can pick the call.
Thanks! __Yehavi:
2010/2/17 Brian West < <brian@freeswitch.org>brian@freeswitch.org>
Step 1. Enable manage-shared-appearance=true
Step 2. Now in the phone's config Configure the phone as usually, set
the
line shared and DO NOT set the third party name.
Step 3. Reboot
It should work.
I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine.
The gateway info missing is a gateway you have configured getting a notify. It has nothing to do with SCA.
/b
On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
> .
_______________________________________________ FreeSWITCH-users mailing list <FreeSWITCH-users@lists.freeswitch.org> FreeSWITCH-users@lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://lists.freeswitch.org/mailman/options/freeswitch-users <http://www.freeswitch.org/>http://www.freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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That's great it's working, but doesn't that seems more like a workaround than an actual solution? What if you want to avail of DNS and use SRV records, do you really want to be hardcoding all your phones with IPs? Cheers, Gabe On Sun, Feb 28, 2010 at 1:49 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello all,
The problem is solved, at least for my Polycoms.
The solution is to have "domain" variable (in vars.xml) set to the IP address of the profile that is used for phones registration (the default is ok if you have only one interface). Define the phone's registration and proxy servers to that address. You must use IP addresses and not DNS names.
Thanks to Anthony for his helpfull tips! __Yehavi:
2010/2/19 Anthony Minessale <anthony.minessale@gmail.com>
go see my comments on that bug note. be prepared to give us ssh access and call or irc so we can can see you reproducing it.
If you are not on the latest firmware on all the phones, we will not continue with this process.
On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
A jira issue has been created: MODSOFIA-61.
Thanks, __Yehavi:
2010/2/19 Michael Jerris <mike@jerris.com>
If this issue is not already on jira could you please make sure it gets added? Mike
On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello Gabe,
As you can see - Brian is actively investigating it, so you can expect for some fix soon...
Regards, __Yehavi:
2010/2/19 Gabriel Kuri <gkuri@ieee.org>
When a call arrives, both ring; the one that did not answer gets only a cancel mesage without any further notification that the extension is in use by the other phone.
These are the same exact symptoms I posted about earlier this week, with the Cisco SPA-5xx series phones. I still have yet to figure out why this is happening, if you find out what's going on, please post back the solution, I'd like to know the resolution.
Thanks, Gabe
On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Thanks Brian. It now works better, but not fully (using 16659M).
What happens is:
When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use". When one of the Polycom phones initiates a call - all is ok:
The other side sees that the extension is in use. When it is put to hold all phones who share this extension see it and can pick the call.
Thanks! __Yehavi:
2010/2/17 Brian West <brian@freeswitch.org> > > Step 1. Enable manage-shared-appearance=true > > Step 2. Now in the phone's config Configure the phone as usually, > set the > line shared and DO NOT set the third party name. > > Step 3. Reboot > > It should work. > > I wish someone that has this working would write some wiki docs > these > threads about it not working are getting rather old when I know for > a fact > they work fine. > > The gateway info missing is a gateway you have configured getting a > notify. It has nothing to do with SCA. > > /b > > On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote: > > > . > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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Hello Gabe, I agree that this is somewhat limiting, but with Polycom's central provisioning (via XML files) I don't see this as a major drawback. __Yehavi: 2010/2/28 Gabriel Kuri <gkuri@ieee.org>
That's great it's working, but doesn't that seems more like a workaround than an actual solution? What if you want to avail of DNS and use SRV records, do you really want to be hardcoding all your phones with IPs?
Cheers, Gabe
On Sun, Feb 28, 2010 at 1:49 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello all,
The problem is solved, at least for my Polycoms.
The solution is to have "domain" variable (in vars.xml) set to the IP address of the profile that is used for phones registration (the default is ok if you have only one interface). Define the phone's registration and proxy servers to that address. You must use IP addresses and not DNS names.
Thanks to Anthony for his helpfull tips! __Yehavi:
2010/2/19 Anthony Minessale <anthony.minessale@gmail.com>
go see my comments on that bug note. be prepared to give us ssh access and call or irc so we can can see you reproducing it.
If you are not on the latest firmware on all the phones, we will not continue with this process.
On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
A jira issue has been created: MODSOFIA-61.
Thanks, __Yehavi:
2010/2/19 Michael Jerris <mike@jerris.com>
If this issue is not already on jira could you please make sure it
gets
added? Mike
On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello Gabe,
As you can see - Brian is actively investigating it, so you can expect for some fix soon...
Regards, __Yehavi:
2010/2/19 Gabriel Kuri <gkuri@ieee.org>
> When a call arrives, both ring; the one that did not answer gets > only a > cancel mesage without any further notification that the extension
is
> in use > by the other phone.
These are the same exact symptoms I posted about earlier this week, with the Cisco SPA-5xx series phones. I still have yet to figure out why this is happening, if you find out what's going on, please post back the solution, I'd like to know the resolution.
Thanks, Gabe
On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote: > Thanks Brian. It now works better, but not fully (using 16659M). > > What happens is: > > When one of the Polycoms seize the line it is ok - the other phone > gets > notification and the extension status is "in use". > When one of the Polycom phones initiates a call - all is ok: > > The other side sees that the extension is in use. > When it is put to hold all phones who share this extension see it and > can > pick the call. >
> > Thanks! __Yehavi: > > 2010/2/17 Brian West <brian@freeswitch.org> >> >> Step 1. Enable manage-shared-appearance=true >> >> Step 2. Now in the phone's config Configure the phone as usually, >> set the >> line shared and DO NOT set the third party name. >> >> Step 3. Reboot >> >> It should work. >> >> I wish someone that has this working would write some wiki docs >> these >> threads about it not working are getting rather old when I know for >> a fact >> they work fine. >> >> The gateway info missing is a gateway you have configured getting a >> notify. It has nothing to do with SCA. >> >> /b >> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote: >> >> > . >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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Really? Come on guys... the feature is something you can't find elsewhere without paying and you're all still not totally pleased with it? /me shakes his head. We have already started talking about how to make the feature more robust. /b On Feb 28, 2010, at 11:56 AM, Yehavi Bourvine wrote:
Hello Gabe,
I agree that this is somewhat limiting, but with Polycom's central provisioning (via XML files) I don't see this as a major drawback.
__Yehavi:
This is one of the most important feature my users want. They don't care how I do it, they are just happy it works. Thanks! __Yehavi: 2010/2/28 Brian West <brian@freeswitch.org>
Really? Come on guys... the feature is something you can't find elsewhere without paying and you're all still not totally pleased with it?
/me shakes his head.
We have already started talking about how to make the feature more robust.
/b
On Feb 28, 2010, at 11:56 AM, Yehavi Bourvine wrote:
Hello Gabe,
I agree that this is somewhat limiting, but with Polycom's central provisioning (via XML files) I don't see this as a major drawback.
__Yehavi:
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I am trying my best not to be completely annoyed at the "somewhat limiting" remark after going over the countless hours I spent making sure this feature work and then being patient with 2 people insisting there was a bug when they both shared an identical misconfiguration. If we started doing DNS lookups everywhere so hostname could be interchanged with IP we would have large blocking all over the place when the DNS was not available and horribly messy code doing all the lookups. There was already an experiment called project asterisk that proved the flaws in this design approach. FreeSWITCH requires that you keep the hostname uniform for all configuration be it a domain name or an IP so long as its identical everywhere. an IP *is* a domain in sip it's not another form of some DNS lookup. On Sun, Feb 28, 2010 at 12:23 PM, Yehavi Bourvine <yehavi.bourvine@gmail.com
wrote:
This is one of the most important feature my users want. They don't care how I do it, they are just happy it works.
Thanks! __Yehavi:
2010/2/28 Brian West <brian@freeswitch.org>
Really? Come on guys... the feature is something you can't find elsewhere
without paying and you're all still not totally pleased with it?
/me shakes his head.
We have already started talking about how to make the feature more robust.
/b
On Feb 28, 2010, at 11:56 AM, Yehavi Bourvine wrote:
Hello Gabe,
I agree that this is somewhat limiting, but with Polycom's central provisioning (via XML files) I don't see this as a major drawback.
__Yehavi:
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Hello Anthony, 2010/3/1 Anthony Minessale <anthony.minessale@gmail.com>
I am trying my best not to be completely annoyed at the "somewhat limiting" remark after going over the countless hours I spent making sure this feature work and then being patient with 2 people insisting there was a bug when they both shared an identical misconfiguration.
"somewhat limiting" was because my experiments up to now with Polycom and SIP/TLS were successfull with domain names only. However, I am going to inverstigate this further to see how this can work with IP only. In the meantime I've documented the required configuration under the Polycom wiki page.
If we started doing DNS lookups everywhere so hostname could be interchanged with IP we would have large blocking all over the place when the DNS was not available and horribly messy code doing all the lookups. There was already an experiment called project asterisk that proved the flaws in this design approach.
I think that's why some of us move from Asterisk to FreeSwitch. I also preffer less to no DNS lookups on a critical system which must work even when there are some kinds of communication problems.
Thanks, __Yehavi:
you can use names if you use *all names* you just can't mix, but all IP is much faster and safer because there is no need to do constant dns lookups on something that never changes anyway, you can easily alias your profile to a friendly name, it also makes it more portable. On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine <yehavi.bourvine@gmail.com>wrote:
Hello Anthony,
2010/3/1 Anthony Minessale <anthony.minessale@gmail.com>
I am trying my best not to be completely annoyed at the "somewhat limiting" remark after going over the countless hours I spent making sure this feature work and then being patient with 2 people insisting there was a bug when they both shared an identical misconfiguration.
"somewhat limiting" was because my experiments up to now with Polycom and SIP/TLS were successfull with domain names only. However, I am going to inverstigate this further to see how this can work with IP only.
In the meantime I've documented the required configuration under the Polycom wiki page.
If we started doing DNS lookups everywhere so hostname could be interchanged with IP we would have large blocking all over the place when the DNS was not available and horribly messy code doing all the lookups. There was already an experiment called project asterisk that proved the flaws in this design approach.
I think that's why some of us move from Asterisk to FreeSwitch. I also preffer less to no DNS lookups on a critical system which must work even when there are some kinds of communication problems.
Thanks, __Yehavi:
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Hi Anthony, I appreciate all the hard work yourself, Brian, Mike and all the other developers have put in FS! By no means am I discounting any of it, it's truly invaluable. Our "philosophy" has always been to use DNS where available, so that we're not worrying about re-configuring a bunch of devices if IP addresses change and this obviously goes to setting the phones up as well to use DNS for the proxy and registrar fields. And it allows us to do fancy things, like split-view DNS, where if a phone is taken off-site and we want it to remotely register to FS, it will get the public IP address of the FS server, rather than the private, without re-configuring the phone. If I'm understanding your email correctly, you're saying we need to use all IP or all DNS for SLA, is that correct? AFAIK, we're using all DNS, but perhaps I'm missing something somewhere. How would we be able to use the DNS name of our SIP domain in phones (rather than IP) and ensure SLA works for both incoming/outgoing calls? Is that possible at this point in time or do we need to use the IP addresses in the proxy field on the phone if we want to use SLA? Thanks for all your help. Gabriel Kuri On Mon, Mar 1, 2010 at 11:42 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you can use names if you use *all names* you just can't mix, but all IP is much faster and safer because there is no need to do constant dns lookups on something that never changes anyway, you can easily alias your profile to a friendly name, it also makes it more portable.
On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello Anthony,
2010/3/1 Anthony Minessale <anthony.minessale@gmail.com>
I am trying my best not to be completely annoyed at the "somewhat limiting" remark after going over the countless hours I spent making sure this feature work and then being patient with 2 people insisting there was a bug when they both shared an identical misconfiguration.
"somewhat limiting" was because my experiments up to now with Polycom and SIP/TLS were successfull with domain names only. However, I am going to inverstigate this further to see how this can work with IP only.
In the meantime I've documented the required configuration under the Polycom wiki page.
If we started doing DNS lookups everywhere so hostname could be interchanged with IP we would have large blocking all over the place when the DNS was not available and horribly messy code doing all the lookups. There was already an experiment called project asterisk that proved the flaws in this design approach.
I think that's why some of us move from Asterisk to FreeSwitch. I also preffer less to no DNS lookups on a critical system which must work even when there are some kinds of communication problems.
Thanks, __Yehavi:
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I really want to be done with this thread...... You were using the IP in some places and the hostnames in others, this makes it so the SQL statements not match because sql statements cannot do dns lookups. I will not be doing any work to resolve hostnames back and forth to IP inside FS. The subscribes were using the hostname and the db and invites were using the IP. you have to have the same exact domain in your user directory and have the force-register-domain and force-register-domain-db also set to that value or omit them completely and use the domain names in 100% of the config fields on the phone. On Mon, Mar 1, 2010 at 3:09 PM, Gabriel Kuri <gkuri@ieee.org> wrote:
Hi Anthony,
I appreciate all the hard work yourself, Brian, Mike and all the other developers have put in FS! By no means am I discounting any of it, it's truly invaluable.
Our "philosophy" has always been to use DNS where available, so that we're not worrying about re-configuring a bunch of devices if IP addresses change and this obviously goes to setting the phones up as well to use DNS for the proxy and registrar fields. And it allows us to do fancy things, like split-view DNS, where if a phone is taken off-site and we want it to remotely register to FS, it will get the public IP address of the FS server, rather than the private, without re-configuring the phone.
If I'm understanding your email correctly, you're saying we need to use all IP or all DNS for SLA, is that correct? AFAIK, we're using all DNS, but perhaps I'm missing something somewhere. How would we be able to use the DNS name of our SIP domain in phones (rather than IP) and ensure SLA works for both incoming/outgoing calls? Is that possible at this point in time or do we need to use the IP addresses in the proxy field on the phone if we want to use SLA?
Thanks for all your help.
Gabriel Kuri
On Mon, Mar 1, 2010 at 11:42 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you can use names if you use *all names* you just can't mix, but all IP is much faster and safer because there is no need to do constant dns lookups on something that never changes anyway, you can easily alias your profile to a friendly name, it also makes it more portable.
On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine < yehavi.bourvine@gmail.com> wrote:
Hello Anthony,
2010/3/1 Anthony Minessale <anthony.minessale@gmail.com>
I am trying my best not to be completely annoyed at the "somewhat limiting" remark after going over the countless hours I spent making
sure
this feature work and then being patient with 2 people insisting there was a bug when they both shared an identical misconfiguration.
"somewhat limiting" was because my experiments up to now with Polycom and SIP/TLS were successfull with domain names only. However, I am going to inverstigate this further to see how this can work with IP only.
In the meantime I've documented the required configuration under the Polycom wiki page.
If we started doing DNS lookups everywhere so hostname could be interchanged with IP we would have large blocking all over the place
when
the DNS was not available and horribly messy code doing all the lookups. There was already an experiment called project asterisk that proved the flaws in this design approach.
I think that's why some of us move from Asterisk to FreeSwitch. I also preffer less to no DNS lookups on a critical system which must work even when there are some kinds of communication problems.
Thanks, __Yehavi:
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
I've torn my hair out for hours and I still can't get it to work with the domain name set everywhere <sigh> ... I can get it to work if I hardcode the IP address on the phones and in all the config files, but I can't get the domain name to work on incoming calls, I'm at a total loss. If I had a mismatch somewhere, I'd think it wouldn't work on outgoing calls either, but that seems to work fine. Any other ideas? Might it have anything to do with the incoming call coming in from the PSTN and going through the external profile? Cheers, Gabe On Mon, Mar 1, 2010 at 1:22 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I really want to be done with this thread......
You were using the IP in some places and the hostnames in others, this makes it so the SQL statements not match because sql statements cannot do dns lookups. I will not be doing any work to resolve hostnames back and forth to IP inside FS.
The subscribes were using the hostname and the db and invites were using the IP.
you have to have the same exact domain in your user directory and have the force-register-domain and force-register-domain-db also set to that value or omit them completely and use the domain names in 100% of the config fields on the phone.
On Mon, Mar 1, 2010 at 3:09 PM, Gabriel Kuri <gkuri@ieee.org> wrote:
Hi Anthony,
I appreciate all the hard work yourself, Brian, Mike and all the other developers have put in FS! By no means am I discounting any of it, it's truly invaluable.
Our "philosophy" has always been to use DNS where available, so that we're not worrying about re-configuring a bunch of devices if IP addresses change and this obviously goes to setting the phones up as well to use DNS for the proxy and registrar fields. And it allows us to do fancy things, like split-view DNS, where if a phone is taken off-site and we want it to remotely register to FS, it will get the public IP address of the FS server, rather than the private, without re-configuring the phone.
If I'm understanding your email correctly, you're saying we need to use all IP or all DNS for SLA, is that correct? AFAIK, we're using all DNS, but perhaps I'm missing something somewhere. How would we be able to use the DNS name of our SIP domain in phones (rather than IP) and ensure SLA works for both incoming/outgoing calls? Is that possible at this point in time or do we need to use the IP addresses in the proxy field on the phone if we want to use SLA?
Thanks for all your help.
Gabriel Kuri
On Mon, Mar 1, 2010 at 11:42 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you can use names if you use *all names* you just can't mix, but all IP is much faster and safer because there is no need to do constant dns lookups on something that never changes anyway, you can easily alias your profile to a friendly name, it also makes it more portable.
On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello Anthony,
2010/3/1 Anthony Minessale <anthony.minessale@gmail.com>
I am trying my best not to be completely annoyed at the "somewhat limiting" remark after going over the countless hours I spent making sure this feature work and then being patient with 2 people insisting there was a bug when they both shared an identical misconfiguration.
"somewhat limiting" was because my experiments up to now with Polycom and SIP/TLS were successfull with domain names only. However, I am going to inverstigate this further to see how this can work with IP only.
In the meantime I've documented the required configuration under the Polycom wiki page.
If we started doing DNS lookups everywhere so hostname could be interchanged with IP we would have large blocking all over the place when the DNS was not available and horribly messy code doing all the lookups. There was already an experiment called project asterisk that proved the flaws in this design approach.
I think that's why some of us move from Asterisk to FreeSwitch. I also preffer less to no DNS lookups on a critical system which must work even when there are some kinds of communication problems.
Thanks, __Yehavi:
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You have to watch phones that might do the switch on you... like resolve the IP then do the subscribe incorrectly... that could happen. Verify the subscribe packet... /b On Mar 1, 2010, at 11:49 PM, Gabriel Kuri wrote:
I've torn my hair out for hours and I still can't get it to work with the domain name set everywhere <sigh> ...
I can get it to work if I hardcode the IP address on the phones and in all the config files, but I can't get the domain name to work on incoming calls, I'm at a total loss. If I had a mismatch somewhere, I'd think it wouldn't work on outgoing calls either, but that seems to work fine.
Any other ideas? Might it have anything to do with the incoming call coming in from the PSTN and going through the external profile?
Cheers, Gabe
participants (6)
-
Anthony Minessale
-
Brian West
-
Chris Chen
-
Gabriel Kuri
-
Michael Jerris
-
Yehavi Bourvine