Hello to the FreeSWITCH users, I'm experiencing problems registering an Aastra 6731i phone to FreeSWITCH. The registration process itself works fine, but the phone never re-registers to FS after the expiration time specified in the expires paramater set in the Contact SIP Header Field. And therefore becomes unreachable from FS after some time. The phone seems to be buggy because it simply ignores the value given in the expires parameter. Testing with Asterisk, the phone remains accessible, because Asterisk adds an Expires HF in addition to the expires param in the Contact HF. Did anyone observed what's described here? I've tested several Aastra firmwares with the same result. I have also tried to set the registration expiration time on the phone itself without success. I believe I can cope with this situation by adding an Expires HF from the (SIP Express Router) proxy that stays between the phone and FS, but I'm sure there's a simpler way to get a proper registation process. I'm using FreeSWITCH 1.0.6 (that comes along with the Blue.box GUI). Kind regards, Philippe
Hi Philippe, I have been testing just about every model of phone that Aastra makes (or has made), on our Freeswitch system, and they all work fine, even behind NAT. It's probably just a setting that isn't right, as you can adjust a lot of this stuff in the phone config file. Can you post your aastra.cfg? I'll take a look and see if there is anything obvious. What is your topology? Are the phone behind NAT, or is this a LAN environment? -Tim Philippe Sultan wrote:
Hello to the FreeSWITCH users,
I'm experiencing problems registering an Aastra 6731i phone to FreeSWITCH. The registration process itself works fine, but the phone never re-registers to FS after the expiration time specified in the expires paramater set in the Contact SIP Header Field. And therefore becomes unreachable from FS after some time.
The phone seems to be buggy because it simply ignores the value given in the expires parameter. Testing with Asterisk, the phone remains accessible, because Asterisk adds an Expires HF in addition to the expires param in the Contact HF.
Did anyone observed what's described here? I've tested several Aastra firmwares with the same result. I have also tried to set the registration expiration time on the phone itself without success.
I believe I can cope with this situation by adding an Expires HF from the (SIP Express Router) proxy that stays between the phone and FS, but I'm sure there's a simpler way to get a proper registation process.
I'm using FreeSWITCH 1.0.6 (that comes along with the Blue.box GUI).
Kind regards,
Philippe
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Tim, Thanks a lot for your answer. No NAT involved here. However, the phone is configured with an outbound proxy and the registrar/proxy parameters are only used as realms (that is, they don't match with any DNS A nor SRV record). Here is my aastra.cfg file : [root@xcom ~]# cat /tftpboot/aastra.cfg upgrade file name: 6731i-2.6.0.st upgrade uri: tftp://128.93.136.254:69 firmware md5: d74b608df20a8e785a7361701d15551e boot count: 1 time server disabled: 0 time server1: 192.93.2.20 time server1: 128.93.1.9 time zone name: FR-Paris time zone code: CET date format: 5 dst config: 3 time format: 1 dhcp: 0 ip: 128.93.136.253 default gateway: 128.93.136.100 dns1: 128.93.1.9 dns2: 128.93.1.23 contact rcs: 0 sip transport protocol: 2 sip outbound support: 1 sip registration retry timer: 10 sip registration renewal timer: 15 sip registration timeout retry timer: 10 sip registration period: 60 sip rport: 1 sip auth name: 492 sip password: XXXXX sip user name: 492 sip screen name: Philippe sip screen name 2: 0178097204 sip proxy ip: vm.test.inria.fr sip proxy port: 5060 sip registrar ip: vm.test.inria.fr sip registrar port: 5060 sip outbound proxy: 128.93.136.252 sip outbound proxy port: 5060 sip backup proxy ip: sip backup registrar ip: vm.test.inria.fr sip backup registrar port: 5060 sip silence suppression: 0 sip transaction timer: 4000 sip rtp port: 10000 sip customized codec: payload=8;ptime=20;silsupp=off sip line1 registration period: 60 sip line2 registration period: 60 log module sip: 10 lldp: 0 sip gruu: 0 sip instance id: 0 [root@xcom ~]# Thanks again help, Philippe On Wed, Feb 9, 2011 at 4:57 PM, Tim St. Pierre <fs-list@communicatefreely.net> wrote:
Hi Philippe,
I have been testing just about every model of phone that Aastra makes (or has made), on our Freeswitch system, and they all work fine, even behind NAT. It's probably just a setting that isn't right, as you can adjust a lot of this stuff in the phone config file.
Can you post your aastra.cfg? I'll take a look and see if there is anything obvious.
What is your topology? Are the phone behind NAT, or is this a LAN environment?
-Tim
Philippe Sultan wrote:
Hello to the FreeSWITCH users,
I'm experiencing problems registering an Aastra 6731i phone to FreeSWITCH. The registration process itself works fine, but the phone never re-registers to FS after the expiration time specified in the expires paramater set in the Contact SIP Header Field. And therefore becomes unreachable from FS after some time.
The phone seems to be buggy because it simply ignores the value given in the expires parameter. Testing with Asterisk, the phone remains accessible, because Asterisk adds an Expires HF in addition to the expires param in the Contact HF.
Did anyone observed what's described here? I've tested several Aastra firmwares with the same result. I have also tried to set the registration expiration time on the phone itself without success.
I believe I can cope with this situation by adding an Expires HF from the (SIP Express Router) proxy that stays between the phone and FS, but I'm sure there's a simpler way to get a proper registation process.
I'm using FreeSWITCH 1.0.6 (that comes along with the Blue.box GUI).
Kind regards,
Philippe
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-- Philippe Sultan
I'm wondering about these values:
sip registration retry timer: 10 sip registration renewal timer: 15 sip registration timeout retry timer: 10 sip registration period: 60 sip rport: 1
They seem a bit short, especially on a LAN. Do you really need your phones to re-register every 45 seconds? Try setting sip registration period to 600 That's 10 minutes. sip registration renewal timer can be a bit longer - 30 seconds maybe, or 60 if you are at all worried about reliability. The sip registration renewal timer sets how long BEFORE expiry that the phone will re-register. I wonder if that value is too short, and something is rejecting it as invalid. Here's a section from my default config. It seems to work fairly well. If that doesn't work, we probably need to check the sofia profile. Let me know how you make out. -Tim sip dtmf method: 0 sip rtp port: 10000 sip mode: 0 sip silence suppression: 0 sip digit time out: 3 sip update callerid: 0 sip send mac: 1 sip rport: 0 sip blf subscription period: 1800 sip registration period: 3600 sip registration renewal timer: 1200 sip T2 timer: 2000 sip transaction timer: 12000 sip use basic codecs: 0 sip customized codec: payload=115;ptime=20;silsupp=off,payload=97;ptime=20;silsupp=off,payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=of$
Tim, That helped a lot, thanks! More comments inline. On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre <fs-list@communicatefreely.net> wrote:
I'm wondering about these values:
sip registration retry timer: 10 sip registration renewal timer: 15 sip registration timeout retry timer: 10 sip registration period: 60 sip rport: 1
They seem a bit short, especially on a LAN. Do you really need your phones to re-register every 45 seconds?
Indeed, I don't. But need a mechanism to keep NAT sessions alive in case phones are being deployed behind NAT boxes. It seems like transporting SIP over TCP from the phone, plus using TCP keepalives will solve that issue though. I also have Thomson ST2030 phones, and they send SIP OPTIONS packets every 50 seconds for that purpose (all handled by the SER proxy).
Try setting sip registration period to 600 That's 10 minutes. sip registration renewal timer can be a bit longer - 30 seconds maybe, or 60 if you are at all worried about reliability.
Here is what I put in the aastra.cfg file from your suggestions : sip registration period: 3600 sip registration renewal timer: 3565 And that does the trick, that is, the phone registers every 45 seconds. I'll try to update those values to have a larger registration period, rely on TCP keepalives in order to maintain NAT sessions, and have all that NAT handling in SER. Here is the topology : Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS Thanks again! Philippe
Philippe, can you confirm there is no NAT Keepalive feature in the Aastra ? That's quite weird. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 10/02/2011 à 11:39, Philippe Sultan a écrit :
Tim,
That helped a lot, thanks! More comments inline.
On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre <fs-list@communicatefreely.net> wrote:
I'm wondering about these values:
sip registration retry timer: 10 sip registration renewal timer: 15 sip registration timeout retry timer: 10 sip registration period: 60 sip rport: 1
They seem a bit short, especially on a LAN. Do you really need your phones to re-register every 45 seconds?
Indeed, I don't. But need a mechanism to keep NAT sessions alive in case phones are being deployed behind NAT boxes. It seems like transporting SIP over TCP from the phone, plus using TCP keepalives will solve that issue though. I also have Thomson ST2030 phones, and they send SIP OPTIONS packets every 50 seconds for that purpose (all handled by the SER proxy).
Try setting sip registration period to 600 That's 10 minutes. sip registration renewal timer can be a bit longer - 30 seconds maybe, or 60 if you are at all worried about reliability.
Here is what I put in the aastra.cfg file from your suggestions :
sip registration period: 3600 sip registration renewal timer: 3565
And that does the trick, that is, the phone registers every 45 seconds. I'll try to update those values to have a larger registration period, rely on TCP keepalives in order to maintain NAT sessions, and have all that NAT handling in SER. Here is the topology :
Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS
ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS
Thanks again!
Philippe
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Tim, Can you share your success stories FreeSWITCH and Aastra. Aastra Phones Behind the NAT? In my case Aastra phones registration not a problem. But calls drooped every 60 sec ... in the same environment Linksys and Polycom works perfectly. How stable the Aastra phones with FreeSWITCH system. TIA Lloyd On Thu, Feb 10, 2011 at 6:23 AM, David Ponzone <david.ponzone@ipeva.fr>wrote:
Philippe,
can you confirm there is no NAT Keepalive feature in the Aastra ? That's quite weird.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
*Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. **IPeva** décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.* * *
Le 10/02/2011 à 11:39, Philippe Sultan a écrit :
Tim,
That helped a lot, thanks! More comments inline.
On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre <fs-list@communicatefreely.net> wrote:
I'm wondering about these values:
sip registration retry timer: 10
sip registration renewal timer: 15
sip registration timeout retry timer: 10
sip registration period: 60
sip rport: 1
They seem a bit short, especially on a LAN. Do you really need your
phones to re-register every 45 seconds?
Indeed, I don't. But need a mechanism to keep NAT sessions alive in case phones are being deployed behind NAT boxes. It seems like transporting SIP over TCP from the phone, plus using TCP keepalives will solve that issue though. I also have Thomson ST2030 phones, and they send SIP OPTIONS packets every 50 seconds for that purpose (all handled by the SER proxy).
Try setting sip registration period to 600 That's 10 minutes.
sip registration renewal timer can be a bit longer - 30 seconds maybe,
or 60 if you are at all worried about reliability.
Here is what I put in the aastra.cfg file from your suggestions :
sip registration period: 3600 sip registration renewal timer: 3565
And that does the trick, that is, the phone registers every 45 seconds. I'll try to update those values to have a larger registration period, rely on TCP keepalives in order to maintain NAT sessions, and have all that NAT handling in SER. Here is the topology :
Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS
ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS
Thanks again!
Philippe
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Im using freeswitch with dozen aastra phones (6757i and 6739i) for about a year. They work perfectly, all the time. If you're behind NAT, it would also be possible that the NAT device is the source of your calls dropping. Erik Van: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Namens Aloysius Lloyd Verzonden: donderdag 10 februari 2011 15:25 Aan: FreeSWITCH Users Help Onderwerp: Re: [Freeswitch-users] Aastra phone registration lost Tim, Can you share your success stories FreeSWITCH and Aastra. Aastra Phones Behind the NAT? In my case Aastra phones registration not a problem. But calls drooped every 60 sec ... in the same environment Linksys and Polycom works perfectly. How stable the Aastra phones with FreeSWITCH system. TIA Lloyd On Thu, Feb 10, 2011 at 6:23 AM, David Ponzone <david.ponzone@ipeva.fr<mailto:david.ponzone@ipeva.fr>> wrote: Philippe, can you confirm there is no NAT Keepalive feature in the Aastra ? That's quite weird. David Ponzone Direction Technique email: david.ponzone@ipeva.fr<mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr<http://www.ipeva.fr> - www.ipeva-studio.com<http://www.ipeva-studio.com> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 10/02/2011 à 11:39, Philippe Sultan a écrit : Tim, That helped a lot, thanks! More comments inline. On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre <fs-list@communicatefreely.net<mailto:fs-list@communicatefreely.net>> wrote: I'm wondering about these values: sip registration retry timer: 10 sip registration renewal timer: 15 sip registration timeout retry timer: 10 sip registration period: 60 sip rport: 1 They seem a bit short, especially on a LAN. Do you really need your phones to re-register every 45 seconds? Indeed, I don't. But need a mechanism to keep NAT sessions alive in case phones are being deployed behind NAT boxes. It seems like transporting SIP over TCP from the phone, plus using TCP keepalives will solve that issue though. I also have Thomson ST2030 phones, and they send SIP OPTIONS packets every 50 seconds for that purpose (all handled by the SER proxy). Try setting sip registration period to 600 That's 10 minutes. sip registration renewal timer can be a bit longer - 30 seconds maybe, or 60 if you are at all worried about reliability. Here is what I put in the aastra.cfg file from your suggestions : sip registration period: 3600 sip registration renewal timer: 3565 And that does the trick, that is, the phone registers every 45 seconds. I'll try to update those values to have a larger registration period, rely on TCP keepalives in order to maintain NAT sessions, and have all that NAT handling in SER. Here is the topology : Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS Thanks again! Philippe _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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<mailto: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 all, i like to write a luascript for callforwaring. The number store in the internal database "db". I find a dbh example. i understand: dbh is mysql or postres-databases. how can i get or insert entrys in the database "db"? tahk you --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc
We do not use db if you are not using odbc for the core than it uses sqlite. Am 10.02.2011 um 16:38 schrieb Thomas Hoellriegel:
Hi all, i like to write a luascript for callforwaring. The number store in the internal database "db". I find a dbh example. i understand: dbh is mysql or postres-databases. how can i get or insert entrys in the database "db"? tahk you
--------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc _______________________________________________ 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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You can't use the internal db for your application. You must use an external DB (Posgres, MySQL, or anyting accessible through ODBC). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 10/02/2011 à 16:38, Thomas Hoellriegel a écrit :
Hi all, i like to write a luascript for callforwaring. The number store in the internal database "db". I find a dbh example. i understand: dbh is mysql or postres-databases. how can i get or insert entrys in the database "db"? tahk you
--------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc _______________________________________________ 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
On Thu, Feb 10, 2011 at 10:36 AM, David Ponzone <david.ponzone@ipeva.fr>wrote:
You can't use the internal db for your application. You must use an external DB (Posgres, MySQL, or anyting accessible through ODBC).
Technically you *could* use the db (or hash) API to store the call-forward information in much the same way that the Local_Extension and the call_return & call-pickup extensions work. I'll leave it as an exercise for the OP to review these extensions in conf/dialplan/default.xml for hints on how to proceed: Local_Extension global-intercept group-intercept intercept-ext redial call_return -MC
Hmm, actually you can use the internal sqlite engine of freeswitch. Ledr and I recently added that to freeswitch.dbh. You can create new tables and use them immediately. If you use 'core:tablename' as the dsn, freeswitch will automatically create a new tablename.db sqlite file if it did not exist yet. For simple queries and when using lua , there is no need to install mysql, postgres or whatever odbc database, and there is also no need to use LuaSQL, and yet you still have persistent storage (unless you use in memory filesystem). Garmt From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, 10 February, 2011 18:01 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] lua question db On Thu, Feb 10, 2011 at 10:36 AM, David Ponzone <david.ponzone@ipeva.fr> wrote: You can't use the internal db for your application. You must use an external DB (Posgres, MySQL, or anyting accessible through ODBC). Technically you *could* use the db (or hash) API to store the call-forward information in much the same way that the Local_Extension and the call_return & call-pickup extensions work. I'll leave it as an exercise for the OP to review these extensions in conf/dialplan/default.xml for hints on how to proceed: Local_Extension global-intercept group-intercept intercept-ext redial call_return -MC
Sure, I normally administer about 300 Aastra phones, with every model they make represented. I have 22 connected to our Freeswitch "beta" system, which will eventually become production. All the endpoints are behind NAT without exception. There are a number of legacy 9133i and 480i phones on the network that don't have the newer NAT traversal features available, but this doesn't seem to be a problem. I have some of the nat traversal options turned on in the sofia profile though, so fs will send media back to the originating address and port. They have been quite reliable, and the sound quality has been excellent, with the newer phones using g722 at 16KHz. There are a few advanced features that I haven't had a chance to play with yet, but here's what I have working: Regular calls, in and out. Intercom calls (auto-answer to speaker phone) Automatic update of destination name and number (updates when checking voice mail, and when calling an extension). Only on newer phones Blind and attended transfer Music on hold SIP using udp or tcp (haven't tried TLS yet) Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no issues as of yet). BLF lamps work correctly, flashing when the phone rings, lit steady when they are on the phone. Distinctive ringing works. I haven't tried SLA yet, but Aastra recently released a firmware update that fixes a missing header, reported to have broken correct SLA operation. I'm hoping to test that in the next week or two. The phones provision very nicely - we auto generate config using PHP scripts that generate a config file on the fly from the user database. These are very easy phones to deploy in large installations, or to the outside world (not readily accessible). They have just added some new features that allow for remote diagnostics of the phones as well. There is a great deal of XML programmability in the phones too, which I'm starting to use for call control and other useful things (updating forwarding rules in the database, or conference and recording control using ESL). Hope that helps! -Tim Aloysius Lloyd wrote:
Tim,
Can you share your success stories FreeSWITCH and Aastra.
Aastra Phones Behind the NAT?
In my case Aastra phones registration not a problem.
But calls drooped every 60 sec ... in the same environment Linksys and Polycom works perfectly.
How stable the Aastra phones with FreeSWITCH system.
TIA
Lloyd
Tim, Based on your input, I kept those settings : sip registration period: 3600 sip registration renewal timer: 3000 The phone registers to FS every 10 minutes. However, any other registration period does not work like it should. It's obvious to me that the phone ignores the expires parameter given in the Contact HF and actually uses what I believe to be its default registation period (3600 sec). I tried to insert an Expires HF to the 200 REGISTER response sent back from FS, and then the phone re-registers based on the time given in this Header Field. It therefore seems to me that the phone is buggy, even though adjusting the two parameters like you suggested it perfectly solves the registration issue. Thanks again, Philippe On Fri, Feb 11, 2011 at 6:54 PM, Tim St. Pierre <fs-list@communicatefreely.net> wrote:
Sure,
I normally administer about 300 Aastra phones, with every model they make represented.
I have 22 connected to our Freeswitch "beta" system, which will eventually become production.
All the endpoints are behind NAT without exception. There are a number of legacy 9133i and 480i phones on the network that don't have the newer NAT traversal features available, but this doesn't seem to be a problem. I have some of the nat traversal options turned on in the sofia profile though, so fs will send media back to the originating address and port.
They have been quite reliable, and the sound quality has been excellent, with the newer phones using g722 at 16KHz.
There are a few advanced features that I haven't had a chance to play with yet, but here's what I have working:
Regular calls, in and out. Intercom calls (auto-answer to speaker phone) Automatic update of destination name and number (updates when checking voice mail, and when calling an extension). Only on newer phones Blind and attended transfer Music on hold SIP using udp or tcp (haven't tried TLS yet) Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no issues as of yet). BLF lamps work correctly, flashing when the phone rings, lit steady when they are on the phone. Distinctive ringing works. I haven't tried SLA yet, but Aastra recently released a firmware update that fixes a missing header, reported to have broken correct SLA operation. I'm hoping to test that in the next week or two.
The phones provision very nicely - we auto generate config using PHP scripts that generate a config file on the fly from the user database. These are very easy phones to deploy in large installations, or to the outside world (not readily accessible). They have just added some new features that allow for remote diagnostics of the phones as well.
There is a great deal of XML programmability in the phones too, which I'm starting to use for call control and other useful things (updating forwarding rules in the database, or conference and recording control using ESL).
Hope that helps!
-Tim
Aloysius Lloyd wrote:
Tim,
Can you share your success stories FreeSWITCH and Aastra.
Aastra Phones Behind the NAT?
In my case Aastra phones registration not a problem.
But calls drooped every 60 sec ... in the same environment Linksys and Polycom works perfectly.
How stable the Aastra phones with FreeSWITCH system.
TIA
Lloyd
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-- Philippe Sultan
Tim, Thank you for the information. I have around 275 Aastra 9133i models phones in production .These phones installed 31/2 years ago. I am trying to migrate to FreeSWITCH could not make it work reliably. What are the profile settings turned on for these phones works reliably? Thanks Lloyd On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre < fs-list@communicatefreely.net> wrote:
Sure,
I normally administer about 300 Aastra phones, with every model they make represented.
I have 22 connected to our Freeswitch "beta" system, which will eventually become production.
All the endpoints are behind NAT without exception. There are a number of legacy 9133i and 480i phones on the network that don't have the newer NAT traversal features available, but this doesn't seem to be a problem. I have some of the nat traversal options turned on in the sofia profile though, so fs will send media back to the originating address and port.
They have been quite reliable, and the sound quality has been excellent, with the newer phones using g722 at 16KHz.
There are a few advanced features that I haven't had a chance to play with yet, but here's what I have working:
Regular calls, in and out. Intercom calls (auto-answer to speaker phone) Automatic update of destination name and number (updates when checking voice mail, and when calling an extension). Only on newer phones Blind and attended transfer Music on hold SIP using udp or tcp (haven't tried TLS yet) Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no issues as of yet). BLF lamps work correctly, flashing when the phone rings, lit steady when they are on the phone. Distinctive ringing works. I haven't tried SLA yet, but Aastra recently released a firmware update that fixes a missing header, reported to have broken correct SLA operation. I'm hoping to test that in the next week or two.
The phones provision very nicely - we auto generate config using PHP scripts that generate a config file on the fly from the user database. These are very easy phones to deploy in large installations, or to the outside world (not readily accessible). They have just added some new features that allow for remote diagnostics of the phones as well.
There is a great deal of XML programmability in the phones too, which I'm starting to use for call control and other useful things (updating forwarding rules in the database, or conference and recording control using ESL).
Hope that helps!
-Tim
Aloysius Lloyd wrote:
Tim,
Can you share your success stories FreeSWITCH and Aastra.
Aastra Phones Behind the NAT?
In my case Aastra phones registration not a problem.
But calls drooped every 60 sec ... in the same environment Linksys and Polycom works perfectly.
How stable the Aastra phones with FreeSWITCH system.
TIA
Lloyd
_______________________________________________ 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
<domains> <domain name="all" alias="true" parse="true"/> <domain name="pbx.MYDOMAIN.NET" alias="true" parse="false"/> </domains> <settings> <param name="user-agent-string" value="Communicate Freely 2.0"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${internal_sip_port}"/> <param name="dialplan" value="xml"/> <param name="context" value="internal"/> <param name="dtmf-duration" value="100"/> <param name="inbound-codec-prefs" value="$${internal_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${internal_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="sip-ip" value="$${public_ip}"/> <param name="rtp-ip" value="$${public_ip}"/> <param name="hold-music" value="$${moh_prefix}alt"/> <param name="dtmf-type" value="rfc2833"/> <param name="force-register-domain" value="$${domain}"/> <param name="force-subscription-domain" value="pbx.$${domain}"/> <param name="force-register-db-domain" value="$${domain}"/> <param name="force-subscription-expires" value="600"/> These are the most important ones I think. <param name="NDLB-received-in-nat-reg-contact" value="true"/> <param name="sip-force-contact" value=" NDLB-connectile-dysfunction"/> I'm also using sip-force-expires to 600 at the moment, and ping = 10. I will probably increase those eventually to reduce bandwidth. I'm still in Beta right now, but I'm not having too many issues. Some of those params are added per-device using the directory, so I can tweak them depending on which device registers, and what the NAT status is of that device. I'm really pushing to get IPv6 on the phones, as well as on some of the more prominent (but competitive) DSL providers here so that we can forego NAT altogether some day. Hope that's helpful. I haven't really gone through and figured out which variables do what at the moment, but it seems to work as it is. -Tim Aloysius Lloyd wrote:
Tim,
Thank you for the information.
I have around 275 Aastra 9133i models phones in production .These phones installed 31/2 years ago. I am trying to migrate to FreeSWITCH could not make it work reliably.
What are the profile settings turned on for these phones works reliably?
Thanks Lloyd
On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre <fs-list@communicatefreely.net <mailto:fs-list@communicatefreely.net>> wrote:
Sure,
I normally administer about 300 Aastra phones, with every model they make represented.
I have 22 connected to our Freeswitch "beta" system, which will eventually become production.
All the endpoints are behind NAT without exception. There are a number of legacy 9133i and 480i phones on the network that don't have the newer NAT traversal features available, but this doesn't seem to be a problem. I have some of the nat traversal options turned on in the sofia profile though, so fs will send media back to the originating address and port.
They have been quite reliable, and the sound quality has been excellent, with the newer phones using g722 at 16KHz.
There are a few advanced features that I haven't had a chance to play with yet, but here's what I have working:
Regular calls, in and out. Intercom calls (auto-answer to speaker phone) Automatic update of destination name and number (updates when checking voice mail, and when calling an extension). Only on newer phones Blind and attended transfer Music on hold SIP using udp or tcp (haven't tried TLS yet) Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no issues as of yet). BLF lamps work correctly, flashing when the phone rings, lit steady when they are on the phone. Distinctive ringing works. I haven't tried SLA yet, but Aastra recently released a firmware update that fixes a missing header, reported to have broken correct SLA operation. I'm hoping to test that in the next week or two.
The phones provision very nicely - we auto generate config using PHP scripts that generate a config file on the fly from the user database. These are very easy phones to deploy in large installations, or to the outside world (not readily accessible). They have just added some new features that allow for remote diagnostics of the phones as well.
There is a great deal of XML programmability in the phones too, which I'm starting to use for call control and other useful things (updating forwarding rules in the database, or conference and recording control using ESL).
Hope that helps!
-Tim
Aloysius Lloyd wrote: > Tim, > > Can you share your success stories FreeSWITCH and Aastra. > > Aastra Phones Behind the NAT? > > In my case Aastra phones registration not a problem. > > But calls drooped every 60 sec ... in the same environment Linksys and > Polycom works perfectly. > > How stable the Aastra phones with FreeSWITCH system. > > TIA > > Lloyd
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Tim, Thank you for the settings will give a try. Thanks Lloyd On Mon, Feb 14, 2011 at 6:53 PM, Tim St. Pierre < fs-list@communicatefreely.net> wrote:
<domains> <domain name="all" alias="true" parse="true"/> <domain name="pbx.MYDOMAIN.NET" alias="true" parse="false"/> </domains>
<settings> <param name="user-agent-string" value="Communicate Freely 2.0"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${internal_sip_port}"/> <param name="dialplan" value="xml"/> <param name="context" value="internal"/> <param name="dtmf-duration" value="100"/> <param name="inbound-codec-prefs" value="$${internal_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${internal_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="sip-ip" value="$${public_ip}"/> <param name="rtp-ip" value="$${public_ip}"/> <param name="hold-music" value="$${moh_prefix}alt"/> <param name="dtmf-type" value="rfc2833"/>
<param name="force-register-domain" value="$${domain}"/> <param name="force-subscription-domain" value="pbx.$${domain}"/> <param name="force-register-db-domain" value="$${domain}"/> <param name="force-subscription-expires" value="600"/>
These are the most important ones I think.
<param name="NDLB-received-in-nat-reg-contact" value="true"/> <param name="sip-force-contact" value=" NDLB-connectile-dysfunction"/>
I'm also using sip-force-expires to 600 at the moment, and ping = 10. I will probably increase those eventually to reduce bandwidth. I'm still in Beta right now, but I'm not having too many issues.
Some of those params are added per-device using the directory, so I can tweak them depending on which device registers, and what the NAT status is of that device. I'm really pushing to get IPv6 on the phones, as well as on some of the more prominent (but competitive) DSL providers here so that we can forego NAT altogether some day.
Hope that's helpful. I haven't really gone through and figured out which variables do what at the moment, but it seems to work as it is.
-Tim Aloysius Lloyd wrote:
Tim,
Thank you for the information.
I have around 275 Aastra 9133i models phones in production .These phones installed 31/2 years ago. I am trying to migrate to FreeSWITCH could not make it work reliably.
What are the profile settings turned on for these phones works reliably?
Thanks Lloyd
On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre <fs-list@communicatefreely.net <mailto:fs-list@communicatefreely.net>> wrote:
Sure,
I normally administer about 300 Aastra phones, with every model they make represented.
I have 22 connected to our Freeswitch "beta" system, which will eventually become production.
All the endpoints are behind NAT without exception. There are a number of legacy 9133i and 480i phones on the network that don't have the newer NAT traversal features available, but this doesn't seem to be a problem. I have some of the nat traversal options turned on in the sofia profile though, so fs will send media back to the originating address and port.
They have been quite reliable, and the sound quality has been excellent, with the newer phones using g722 at 16KHz.
There are a few advanced features that I haven't had a chance to play with yet, but here's what I have working:
Regular calls, in and out. Intercom calls (auto-answer to speaker phone) Automatic update of destination name and number (updates when checking voice mail, and when calling an extension). Only on newer phones Blind and attended transfer Music on hold SIP using udp or tcp (haven't tried TLS yet) Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no issues as of yet). BLF lamps work correctly, flashing when the phone rings, lit steady when they are on the phone. Distinctive ringing works. I haven't tried SLA yet, but Aastra recently released a firmware update that fixes a missing header, reported to have broken correct SLA operation. I'm hoping to test that in the next week or two.
The phones provision very nicely - we auto generate config using PHP scripts that generate a config file on the fly from the user database. These are very easy phones to deploy in large installations, or to the outside world (not readily accessible). They have just added some new features that allow for remote diagnostics of the phones as well.
There is a great deal of XML programmability in the phones too, which I'm starting to use for call control and other useful things (updating forwarding rules in the database, or conference and recording control using ESL).
Hope that helps!
-Tim
Aloysius Lloyd wrote: > Tim, > > Can you share your success stories FreeSWITCH and Aastra. > > Aastra Phones Behind the NAT? > > In my case Aastra phones registration not a problem. > > But calls drooped every 60 sec ... in the same environment Linksys and > Polycom works perfectly. > > How stable the Aastra phones with FreeSWITCH system. > > TIA > > Lloyd
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Good luck! There are some aastra.cfg sections earlier in this thread. One thing I found about the 9133i and other legacy phones is that they work better if you don't subscribe to MWI but instead send unsolicited updates. They also have odd default codec settings, so you want to specify the codec string with the ptime that you want (I think they default to 30 instead of 20). I like the newer phones much better, but we have about 150 of these in the field, so we have to make them work. -Tim Aloysius Lloyd wrote:
Tim,
Thank you for the settings will give a try.
Thanks Lloyd
On Mon, Feb 14, 2011 at 6:53 PM, Tim St. Pierre <fs-list@communicatefreely.net <mailto:fs-list@communicatefreely.net>> wrote:
<domains> <domain name="all" alias="true" parse="true"/> <domain name="pbx.MYDOMAIN.NET <http://pbx.MYDOMAIN.NET>" alias="true" parse="false"/> </domains>
<settings> <param name="user-agent-string" value="Communicate Freely 2.0"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${internal_sip_port}"/> <param name="dialplan" value="xml"/> <param name="context" value="internal"/> <param name="dtmf-duration" value="100"/> <param name="inbound-codec-prefs" value="$${internal_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${internal_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="sip-ip" value="$${public_ip}"/> <param name="rtp-ip" value="$${public_ip}"/> <param name="hold-music" value="$${moh_prefix}alt"/> <param name="dtmf-type" value="rfc2833"/>
<param name="force-register-domain" value="$${domain}"/> <param name="force-subscription-domain" value="pbx.$${domain}"/> <param name="force-register-db-domain" value="$${domain}"/> <param name="force-subscription-expires" value="600"/>
These are the most important ones I think.
<param name="NDLB-received-in-nat-reg-contact" value="true"/> <param name="sip-force-contact" value=" NDLB-connectile-dysfunction"/>
I'm also using sip-force-expires to 600 at the moment, and ping = 10. I will probably increase those eventually to reduce bandwidth. I'm still in Beta right now, but I'm not having too many issues.
Some of those params are added per-device using the directory, so I can tweak them depending on which device registers, and what the NAT status is of that device. I'm really pushing to get IPv6 on the phones, as well as on some of the more prominent (but competitive) DSL providers here so that we can forego NAT altogether some day.
Hope that's helpful. I haven't really gone through and figured out which variables do what at the moment, but it seems to work as it is.
-Tim Aloysius Lloyd wrote: > Tim, > > Thank you for the information. > > I have around 275 Aastra 9133i models phones in production .These > phones installed 31/2 years ago. I am trying to migrate to FreeSWITCH > could not make it work reliably. > > What are the profile settings turned on for these phones works reliably? > > > Thanks > Lloyd > > > > On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre > <fs-list@communicatefreely.net <mailto:fs-list@communicatefreely.net> <mailto:fs-list@communicatefreely.net <mailto:fs-list@communicatefreely.net>>> > wrote: > > Sure, > > I normally administer about 300 Aastra phones, with every model they > make represented. > > I have 22 connected to our Freeswitch "beta" system, which will > eventually become production. > > All the endpoints are behind NAT without exception. There are a > number > of legacy 9133i and 480i phones on the network that don't have the > newer > NAT traversal features available, but this doesn't seem to be a > problem. I have some of the nat traversal options turned on in the > sofia profile though, so fs will send media back to the originating > address and port. > > They have been quite reliable, and the sound quality has been > excellent, > with the newer phones using g722 at 16KHz. > > There are a few advanced features that I haven't had a chance to play > with yet, but here's what I have working: > > Regular calls, in and out. > Intercom calls (auto-answer to speaker phone) > Automatic update of destination name and number (updates when checking > voice mail, and when calling an extension). Only on newer phones > Blind and attended transfer > Music on hold > SIP using udp or tcp (haven't tried TLS yet) > Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no > issues > as of yet). > BLF lamps work correctly, flashing when the phone rings, lit > steady when > they are on the phone. > Distinctive ringing works. > I haven't tried SLA yet, but Aastra recently released a firmware > update > that fixes a missing header, reported to have broken correct SLA > operation. I'm hoping to test that in the next week or two. > > The phones provision very nicely - we auto generate config using PHP > scripts that generate a config file on the fly from the user database. > These are very easy phones to deploy in large installations, or to the > outside world (not readily accessible). They have just added some new > features that allow for remote diagnostics of the phones as well. > > There is a great deal of XML programmability in the phones too, which > I'm starting to use for call control and other useful things (updating > forwarding rules in the database, or conference and recording control > using ESL). > > Hope that helps! > > -Tim > > Aloysius Lloyd wrote: > > Tim, > > > > Can you share your success stories FreeSWITCH and Aastra. > > > > Aastra Phones Behind the NAT? > > > > In my case Aastra phones registration not a problem. > > > > But calls drooped every 60 sec ... in the same environment > Linksys and > > Polycom works perfectly. > > > > How stable the Aastra phones with FreeSWITCH system. > > > > TIA > > > > Lloyd > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > <mailto:FreeSWITCH-users@lists.freeswitch.org <mailto: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 <mailto: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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I have the same issue, around 275 phones in the field. I want the 275 phones work with FreeSWITCH. Thanks Lloyd On Tue, Feb 15, 2011 at 10:29 AM, Tim St. Pierre < fs-list@communicatefreely.net> wrote:
Good luck!
There are some aastra.cfg sections earlier in this thread. One thing I found about the 9133i and other legacy phones is that they work better if you don't subscribe to MWI but instead send unsolicited updates. They also have odd default codec settings, so you want to specify the codec string with the ptime that you want (I think they default to 30 instead of 20).
I like the newer phones much better, but we have about 150 of these in the field, so we have to make them work.
-Tim
Aloysius Lloyd wrote:
Tim,
Thank you for the settings will give a try.
Thanks Lloyd
On Mon, Feb 14, 2011 at 6:53 PM, Tim St. Pierre <fs-list@communicatefreely.net <mailto:fs-list@communicatefreely.net>> wrote:
<domains> <domain name="all" alias="true" parse="true"/> <domain name="pbx.MYDOMAIN.NET <http://pbx.MYDOMAIN.NET>" alias="true" parse="false"/> </domains>
<settings> <param name="user-agent-string" value="Communicate Freely 2.0"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${internal_sip_port}"/> <param name="dialplan" value="xml"/> <param name="context" value="internal"/> <param name="dtmf-duration" value="100"/> <param name="inbound-codec-prefs" value="$${internal_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${internal_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="sip-ip" value="$${public_ip}"/> <param name="rtp-ip" value="$${public_ip}"/> <param name="hold-music" value="$${moh_prefix}alt"/> <param name="dtmf-type" value="rfc2833"/>
<param name="force-register-domain" value="$${domain}"/> <param name="force-subscription-domain" value="pbx.$${domain}"/> <param name="force-register-db-domain" value="$${domain}"/> <param name="force-subscription-expires" value="600"/>
These are the most important ones I think.
<param name="NDLB-received-in-nat-reg-contact" value="true"/> <param name="sip-force-contact" value=" NDLB-connectile-dysfunction"/>
I'm also using sip-force-expires to 600 at the moment, and ping = 10. I will probably increase those eventually to reduce bandwidth. I'm still in Beta right now, but I'm not having too many issues.
Some of those params are added per-device using the directory, so I can tweak them depending on which device registers, and what the NAT status is of that device. I'm really pushing to get IPv6 on the phones, as well as on some of the more prominent (but competitive) DSL providers here so that we can forego NAT altogether some day.
Hope that's helpful. I haven't really gone through and figured out which variables do what at the moment, but it seems to work as it is.
-Tim Aloysius Lloyd wrote: > Tim, > > Thank you for the information. > > I have around 275 Aastra 9133i models phones in production .These > phones installed 31/2 years ago. I am trying to migrate to FreeSWITCH > could not make it work reliably. > > What are the profile settings turned on for these phones works reliably? > > > Thanks > Lloyd > > > > On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre > <fs-list@communicatefreely.net <mailto:fs-list@communicatefreely.net> <mailto:fs-list@communicatefreely.net <mailto:fs-list@communicatefreely.net>>> > wrote: > > Sure, > > I normally administer about 300 Aastra phones, with every model they > make represented. > > I have 22 connected to our Freeswitch "beta" system, which will > eventually become production. > > All the endpoints are behind NAT without exception. There are a > number > of legacy 9133i and 480i phones on the network that don't have the > newer > NAT traversal features available, but this doesn't seem to be a > problem. I have some of the nat traversal options turned on in the > sofia profile though, so fs will send media back to the originating > address and port. > > They have been quite reliable, and the sound quality has been > excellent, > with the newer phones using g722 at 16KHz. > > There are a few advanced features that I haven't had a chance to play > with yet, but here's what I have working: > > Regular calls, in and out. > Intercom calls (auto-answer to speaker phone) > Automatic update of destination name and number (updates when checking > voice mail, and when calling an extension). Only on newer phones > Blind and attended transfer > Music on hold > SIP using udp or tcp (haven't tried TLS yet) > Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no > issues > as of yet). > BLF lamps work correctly, flashing when the phone rings, lit > steady when > they are on the phone. > Distinctive ringing works. > I haven't tried SLA yet, but Aastra recently released a firmware > update > that fixes a missing header, reported to have broken correct SLA > operation. I'm hoping to test that in the next week or two. > > The phones provision very nicely - we auto generate config using PHP > scripts that generate a config file on the fly from the user database. > These are very easy phones to deploy in large installations, or to the > outside world (not readily accessible). They have just added some new > features that allow for remote diagnostics of the phones as well. > > There is a great deal of XML programmability in the phones too, which > I'm starting to use for call control and other useful things (updating > forwarding rules in the database, or conference and recording control > using ESL). > > Hope that helps! > > -Tim > > Aloysius Lloyd wrote: > > Tim, > > > > Can you share your success stories FreeSWITCH and Aastra. > > > > Aastra Phones Behind the NAT? > > > > In my case Aastra phones registration not a problem. > > > > But calls drooped every 60 sec ... in the same environment > Linksys and > > Polycom works perfectly. > > > > How stable the Aastra phones with FreeSWITCH system. > > > > TIA > > > > Lloyd > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > <mailto:FreeSWITCH-users@lists.freeswitch.org <mailto: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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I have been setting my expires to 600. Brian West wrote:
Are you setting your expires to > 300 seconds?
/b
On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:
I have the same issue, around 275 phones in the field. I want the 275 phones work with FreeSWITCH.
Thanks Lloyd
------------------------------------------------------------------------
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Hi All, I stay away with Aastra phone for a long time and today I did some tests. All of my test ... the phones not reliable with FreeSWITCH I try both 6731i and 57i with the most recent firmware Here is the configuration files *aastra.cfg* dhcp: 1 sip digit timeout: 3 sip dial plan: "x+#|xx+*|[2-9]XX[2-9]XXXXXX|1[2-9]XX[2-9]XXXXXX|1XXXXXXXXXX|[2-3]XX|67[2-9]XX[2-9]XXXXXX" sip rport: 1 sip customized codec:payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=off,payload=18;ptime=20;silsupp=off #sip registration period: 120 #sip registration renewal timer: 15 headset tx gain: -3 headset sidetone gain: -3 handset tx gain: -3 handset sidetone gain: -3 handsfree tx gain: 0 handset volume: 5 #RX volumes - user adjustable, so easily changed speaker volume: 5 ringer volume: 3 web interface enabled: 1 live dialpad: 1 missed calls indicator disabled: 1 suppress dtmf playback: 0 #audio mode: 2 #0 = speaker (default)1 = headset 2 = speaker/headset 3 = headset/speaker time server disabled: 0 time server1: pool.ntp.org #directory directory 1: internal_list.csv directory 2: external_list.csv *mac.cfg* directed call pickup: 1 directed call pickup prefix: ** # sip line1 screen name: Ext 203 sip line1 display name: Ext 203 sip line1 auth name: 203 sip line1 user name: 203 sip line1 password: ********* sip line1 vmail: *97 sip line1 mode: 0 # sip line1 proxy ip: aastra.mydomain.com #sip line1 proxy port: 5060 sip line1 registrar ip: aastra.mydomain.com sip line1 registration period: 300 #sip line1 registrar port: 5060 ------------ 1. FreeSWITCH Registration shows two entires in the internal profile and when I try to call to the extension two lines on the phones rings .... no idea why this is happen? Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@192.168.240.106:5060 ;transport=udp;fs_nat=yes;fs_path=sip%3A203%40173.33.178.49%3A1627%3Btransport%3Dudp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP-NAT)(unknown) EXP(2011-02-16 22:24:36) EXPSECS(310) Host: li176-12 IP: 173.33.178.49 Port: 1627 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@173.33.178.49:1627;transport=udp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP)(unknown) EXP(2011-02-16 22:24:36) EXPSECS(310) Host: li176-12 IP: 173.33.178.49 Port: 1627 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com 2. After first registration expires .... FreeSWITCH internal registration status shows the following entry. When I dial the extension now there is a long delay ... FreeSWITCH dialing and waiting then goes to voicemail. Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@173.33.178.49:1627;transport=udp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP)(unknown) EXP(2011-02-16 22:31:25) EXPSECS(266) Host: li176-12 IP: 173.33.178.49 Port: 1631 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com 3. Phones Goes randomly "No Service" .... I see this problem several times in the past. ------------- At the same time in my lab Linksys and Polycom Phones working without any issues with the default settings. ------------- my question what is Aastra Doing differently? Thanks Lloyd On Wed, Feb 16, 2011 at 10:53 AM, Tim St. Pierre < fs-list@communicatefreely.net> wrote:
I have been setting my expires to 600.
Brian West wrote:
Are you setting your expires to > 300 seconds?
/b
On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:
I have the same issue, around 275 phones in the field. I want the 275 phones work with FreeSWITCH.
Thanks Lloyd
------------------------------------------------------------------------
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_______________________________________________ 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 did some traces and figure it out the problem losing the Registrations. 1. FreeSWITCH Send the response to a wrong port . Is this a Bug? 2. Why FreeSWITCH + Aastra Registration contact not have the port number ? 3. See the below ... 203 & 202 Aastra Phones on Domain - *aastra.mydomain.com* But user Agent Shows Polycom. BTW in the same LAN I have two polycom phones on different domain *foo.mydomain.com* ============== SIP/2.0 200 OK Via: SIP/2.0/UDP 173.230.136.12;rport;branch=z9hG4bKZ6DDSDNZ11gvS *From: <sip:203@aastra.mydomain.com>;tag=3jFNS2KBvNX0S* *To: "Ext 202" <sip:203@aastra.mydomain.com>;tag=B550915E-F491D63B* CSeq: 8639912 NOTIFY Call-ID: cdad5068-b542-122e-ef86-fefdade6880c *Contact: <sip:202@192.168.240.128:5060>* Event: message-summary *User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.3.1.0769* Accept-Language: en Content-Length: 0 ============= Any help on this why sending wrong ports? Thanks Lloyd On Wed, Feb 16, 2011 at 10:34 PM, Aloysius Lloyd <lloyd.aloysius@gmail.com>wrote:
Hi All,
I stay away with Aastra phone for a long time and today I did some tests. All of my test ... the phones not reliable with FreeSWITCH
I try both 6731i and 57i with the most recent firmware
Here is the configuration files
*aastra.cfg*
dhcp: 1 sip digit timeout: 3 sip dial plan: "x+#|xx+*|[2-9]XX[2-9]XXXXXX|1[2-9]XX[2-9]XXXXXX|1XXXXXXXXXX|[2-3]XX|67[2-9]XX[2-9]XXXXXX" sip rport: 1 sip customized codec:payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=off,payload=18;ptime=20;silsupp=off #sip registration period: 120 #sip registration renewal timer: 15 headset tx gain: -3 headset sidetone gain: -3 handset tx gain: -3 handset sidetone gain: -3 handsfree tx gain: 0 handset volume: 5 #RX volumes - user adjustable, so easily changed speaker volume: 5 ringer volume: 3 web interface enabled: 1 live dialpad: 1 missed calls indicator disabled: 1 suppress dtmf playback: 0 #audio mode: 2 #0 = speaker (default)1 = headset 2 = speaker/headset 3 = headset/speaker time server disabled: 0 time server1: pool.ntp.org #directory directory 1: internal_list.csv directory 2: external_list.csv
*mac.cfg*
directed call pickup: 1 directed call pickup prefix: ** # sip line1 screen name: Ext 203 sip line1 display name: Ext 203 sip line1 auth name: 203 sip line1 user name: 203 sip line1 password: ********* sip line1 vmail: *97 sip line1 mode: 0 # sip line1 proxy ip: aastra.mydomain.com #sip line1 proxy port: 5060 sip line1 registrar ip: aastra.mydomain.com sip line1 registration period: 300 #sip line1 registrar port: 5060
------------
1. FreeSWITCH Registration shows two entires in the internal profile and when I try to call to the extension two lines on the phones rings .... no idea why this is happen?
Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@192.168.240.106:5060 ;transport=udp;fs_nat=yes;fs_path=sip%3A203%40173.33.178.49%3A1627%3Btransport%3Dudp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP-NAT)(unknown) EXP(2011-02-16 22:24:36) EXPSECS(310) Host: li176-12 IP: 173.33.178.49 Port: 1627 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com
Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@173.33.178.49:1627;transport=udp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP)(unknown) EXP(2011-02-16 22:24:36) EXPSECS(310) Host: li176-12 IP: 173.33.178.49 Port: 1627 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com
2. After first registration expires .... FreeSWITCH internal registration status shows the following entry. When I dial the extension now there is a long delay ... FreeSWITCH dialing and waiting then goes to voicemail.
Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@173.33.178.49:1627;transport=udp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP)(unknown) EXP(2011-02-16 22:31:25) EXPSECS(266) Host: li176-12 IP: 173.33.178.49 Port: 1631 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com
3. Phones Goes randomly "No Service" .... I see this problem several times in the past.
-------------
At the same time in my lab Linksys and Polycom Phones working without any issues with the default settings.
-------------
my question what is Aastra Doing differently?
Thanks Lloyd
On Wed, Feb 16, 2011 at 10:53 AM, Tim St. Pierre < fs-list@communicatefreely.net> wrote:
I have been setting my expires to 600.
Brian West wrote:
Are you setting your expires to > 300 seconds?
/b
On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:
I have the same issue, around 275 phones in the field. I want the 275 phones work with FreeSWITCH.
Thanks Lloyd
------------------------------------------------------------------------
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_______________________________________________ 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
Get the SIP trace of the phone doing its registration. Also, is the phone on the same LAN as FreeSWITCH? Is there any NAT involved? -MC On Fri, Feb 18, 2011 at 12:43 PM, Aloysius Lloyd <lloyd.aloysius@gmail.com>wrote:
I did some traces and figure it out the problem losing the Registrations.
1. FreeSWITCH Send the response to a wrong port . Is this a Bug?
2. Why FreeSWITCH + Aastra Registration contact not have the port number ?
3. See the below ...
203 & 202 Aastra Phones on Domain - *aastra.mydomain.com*
But user Agent Shows Polycom. BTW in the same LAN I have two polycom phones on different domain *foo.mydomain.com*
==============
SIP/2.0 200 OK Via: SIP/2.0/UDP 173.230.136.12;rport;branch=z9hG4bKZ6DDSDNZ11gvS *From: <sip:203@aastra.mydomain.com>;tag=3jFNS2KBvNX0S* *To: "Ext 202" <sip:203@aastra.mydomain.com>;tag=B550915E-F491D63B* CSeq: 8639912 NOTIFY Call-ID: cdad5068-b542-122e-ef86-fefdade6880c *Contact: <sip:202@192.168.240.128:5060>* Event: message-summary *User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.3.1.0769* Accept-Language: en Content-Length: 0
=============
Any help on this why sending wrong ports?
Thanks Lloyd
On Wed, Feb 16, 2011 at 10:34 PM, Aloysius Lloyd <lloyd.aloysius@gmail.com
wrote:
Hi All,
I stay away with Aastra phone for a long time and today I did some tests. All of my test ... the phones not reliable with FreeSWITCH
I try both 6731i and 57i with the most recent firmware
Here is the configuration files
*aastra.cfg*
dhcp: 1 sip digit timeout: 3 sip dial plan: "x+#|xx+*|[2-9]XX[2-9]XXXXXX|1[2-9]XX[2-9]XXXXXX|1XXXXXXXXXX|[2-3]XX|67[2-9]XX[2-9]XXXXXX" sip rport: 1 sip customized codec:payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=off,payload=18;ptime=20;silsupp=off #sip registration period: 120 #sip registration renewal timer: 15 headset tx gain: -3 headset sidetone gain: -3 handset tx gain: -3 handset sidetone gain: -3 handsfree tx gain: 0 handset volume: 5 #RX volumes - user adjustable, so easily changed speaker volume: 5 ringer volume: 3 web interface enabled: 1 live dialpad: 1 missed calls indicator disabled: 1 suppress dtmf playback: 0 #audio mode: 2 #0 = speaker (default)1 = headset 2 = speaker/headset 3 = headset/speaker time server disabled: 0 time server1: pool.ntp.org #directory directory 1: internal_list.csv directory 2: external_list.csv
*mac.cfg*
directed call pickup: 1 directed call pickup prefix: ** # sip line1 screen name: Ext 203 sip line1 display name: Ext 203 sip line1 auth name: 203 sip line1 user name: 203 sip line1 password: ********* sip line1 vmail: *97 sip line1 mode: 0 # sip line1 proxy ip: aastra.mydomain.com #sip line1 proxy port: 5060 sip line1 registrar ip: aastra.mydomain.com sip line1 registration period: 300 #sip line1 registrar port: 5060
------------
1. FreeSWITCH Registration shows two entires in the internal profile and when I try to call to the extension two lines on the phones rings .... no idea why this is happen?
Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@192.168.240.106:5060 ;transport=udp;fs_nat=yes;fs_path=sip%3A203%40173.33.178.49%3A1627%3Btransport%3Dudp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP-NAT)(unknown) EXP(2011-02-16 22:24:36) EXPSECS(310) Host: li176-12 IP: 173.33.178.49 Port: 1627 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com
Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@173.33.178.49:1627;transport=udp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP)(unknown) EXP(2011-02-16 22:24:36) EXPSECS(310) Host: li176-12 IP: 173.33.178.49 Port: 1627 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com
2. After first registration expires .... FreeSWITCH internal registration status shows the following entry. When I dial the extension now there is a long delay ... FreeSWITCH dialing and waiting then goes to voicemail.
Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@173.33.178.49:1627;transport=udp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP)(unknown) EXP(2011-02-16 22:31:25) EXPSECS(266) Host: li176-12 IP: 173.33.178.49 Port: 1631 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com
3. Phones Goes randomly "No Service" .... I see this problem several times in the past.
-------------
At the same time in my lab Linksys and Polycom Phones working without any issues with the default settings.
-------------
my question what is Aastra Doing differently?
Thanks Lloyd
On Wed, Feb 16, 2011 at 10:53 AM, Tim St. Pierre < fs-list@communicatefreely.net> wrote:
I have been setting my expires to 600.
Brian West wrote:
Are you setting your expires to > 300 seconds?
/b
On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:
I have the same issue, around 275 phones in the field. I want the 275 phones work with FreeSWITCH.
Thanks Lloyd
------------------------------------------------------------------------
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http://lists.freeswitch.org/mailman/options/freeswitch-users
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FreeSWITCH is on public IP and Phones Behind the NAT. I see sometimes FreeSWITCH send the reply to a wrong port or not responses for the registration request. Please see the following contact Call-ID: 1bb9e78e1cba12f8 User: 202@aastra.mydomain.com *Contact: "Ext 202" <sip:202@173.33.178.49:1287;transport=udp>* Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP)(unknown) EXP(2011-02-18 20:42:53) EXPSECS(358) Host: li176-12 IP: 173.33.178.49 Port: 1287 Auth-User: 202 Auth-Realm: aastra.mydomain.com MWI-Account: 202@aastra.mydomain.com --------- I solve the problem using the <param name="sip-force-contact" value=" NDLB-connectile-dysfunction"/> in the user directory. But why this is happening to Aastra Phones only? Please let me know. Thanks Lloyd On Fri, Feb 18, 2011 at 8:29 PM, Michael Collins <msc@freeswitch.org> wrote:
Get the SIP trace of the phone doing its registration. Also, is the phone on the same LAN as FreeSWITCH? Is there any NAT involved? -MC
On Fri, Feb 18, 2011 at 12:43 PM, Aloysius Lloyd <lloyd.aloysius@gmail.com
wrote:
I did some traces and figure it out the problem losing the Registrations.
1. FreeSWITCH Send the response to a wrong port . Is this a Bug?
2. Why FreeSWITCH + Aastra Registration contact not have the port number ?
3. See the below ...
203 & 202 Aastra Phones on Domain - *aastra.mydomain.com*
But user Agent Shows Polycom. BTW in the same LAN I have two polycom phones on different domain *foo.mydomain.com*
==============
SIP/2.0 200 OK Via: SIP/2.0/UDP 173.230.136.12;rport;branch=z9hG4bKZ6DDSDNZ11gvS *From: <sip:203@aastra.mydomain.com>;tag=3jFNS2KBvNX0S* *To: "Ext 202" <sip:203@aastra.mydomain.com>;tag=B550915E-F491D63B* CSeq: 8639912 NOTIFY Call-ID: cdad5068-b542-122e-ef86-fefdade6880c *Contact: <sip:202@192.168.240.128:5060>* Event: message-summary *User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.3.1.0769* Accept-Language: en Content-Length: 0
=============
Any help on this why sending wrong ports?
Thanks Lloyd
On Wed, Feb 16, 2011 at 10:34 PM, Aloysius Lloyd < lloyd.aloysius@gmail.com> wrote:
Hi All,
I stay away with Aastra phone for a long time and today I did some tests. All of my test ... the phones not reliable with FreeSWITCH
I try both 6731i and 57i with the most recent firmware
Here is the configuration files
*aastra.cfg*
dhcp: 1 sip digit timeout: 3 sip dial plan: "x+#|xx+*|[2-9]XX[2-9]XXXXXX|1[2-9]XX[2-9]XXXXXX|1XXXXXXXXXX|[2-3]XX|67[2-9]XX[2-9]XXXXXX" sip rport: 1 sip customized codec:payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=off,payload=18;ptime=20;silsupp=off #sip registration period: 120 #sip registration renewal timer: 15 headset tx gain: -3 headset sidetone gain: -3 handset tx gain: -3 handset sidetone gain: -3 handsfree tx gain: 0 handset volume: 5 #RX volumes - user adjustable, so easily changed speaker volume: 5 ringer volume: 3 web interface enabled: 1 live dialpad: 1 missed calls indicator disabled: 1 suppress dtmf playback: 0 #audio mode: 2 #0 = speaker (default)1 = headset 2 = speaker/headset 3 = headset/speaker time server disabled: 0 time server1: pool.ntp.org #directory directory 1: internal_list.csv directory 2: external_list.csv
*mac.cfg*
directed call pickup: 1 directed call pickup prefix: ** # sip line1 screen name: Ext 203 sip line1 display name: Ext 203 sip line1 auth name: 203 sip line1 user name: 203 sip line1 password: ********* sip line1 vmail: *97 sip line1 mode: 0 # sip line1 proxy ip: aastra.mydomain.com #sip line1 proxy port: 5060 sip line1 registrar ip: aastra.mydomain.com sip line1 registration period: 300 #sip line1 registrar port: 5060
------------
1. FreeSWITCH Registration shows two entires in the internal profile and when I try to call to the extension two lines on the phones rings .... no idea why this is happen?
Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@192.168.240.106:5060 ;transport=udp;fs_nat=yes;fs_path=sip%3A203%40173.33.178.49%3A1627%3Btransport%3Dudp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP-NAT)(unknown) EXP(2011-02-16 22:24:36) EXPSECS(310) Host: li176-12 IP: 173.33.178.49 Port: 1627 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com
Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@173.33.178.49:1627;transport=udp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP)(unknown) EXP(2011-02-16 22:24:36) EXPSECS(310) Host: li176-12 IP: 173.33.178.49 Port: 1627 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com
2. After first registration expires .... FreeSWITCH internal registration status shows the following entry. When I dial the extension now there is a long delay ... FreeSWITCH dialing and waiting then goes to voicemail.
Call-ID: d8fd4c8795801cd2 User: 203@aastra.mydomain.com Contact: "Ext 203" <sip:203@173.33.178.49:1627;transport=udp> Agent: Aastra 6731i/2.6.0.2010 Status: Registered(UDP)(unknown) EXP(2011-02-16 22:31:25) EXPSECS(266) Host: li176-12 IP: 173.33.178.49 Port: 1631 Auth-User: 203 Auth-Realm: aastra.mydomain.com MWI-Account: 203@aastra.mydomain.com
3. Phones Goes randomly "No Service" .... I see this problem several times in the past.
-------------
At the same time in my lab Linksys and Polycom Phones working without any issues with the default settings.
-------------
my question what is Aastra Doing differently?
Thanks Lloyd
On Wed, Feb 16, 2011 at 10:53 AM, Tim St. Pierre < fs-list@communicatefreely.net> wrote:
I have been setting my expires to 600.
Brian West wrote:
Are you setting your expires to > 300 seconds?
/b
On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:
I have the same issue, around 275 phones in the field. I want the 275 phones work with FreeSWITCH.
Thanks Lloyd
------------------------------------------------------------------------
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http://lists.freeswitch.org/mailman/options/freeswitch-users
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Well, I was looking for something like what's implemented in the Thomson ST2030 series, that is, have the phone initiate SIP OPTIONS, and process them with SER. I was not able to find such a mechanism, but digging in the docs, it appears that I can just use SIP over TCP and keep the connection maintained using TCP keepalive. That wil be just fine I guess. On the other hand, you can of course have FS send SIP OPTIONS packets to the phones, but I'd like to keep the NAT handling on the front end SIP proxy as much as possible. Philippe On Thu, Feb 10, 2011 at 12:23 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
Philippe, can you confirm there is no NAT Keepalive feature in the Aastra ? That's quite weird. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 10/02/2011 à 11:39, Philippe Sultan a écrit :
Tim,
That helped a lot, thanks! More comments inline.
On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre <fs-list@communicatefreely.net> wrote:
I'm wondering about these values:
sip registration retry timer: 10
sip registration renewal timer: 15
sip registration timeout retry timer: 10
sip registration period: 60
sip rport: 1
They seem a bit short, especially on a LAN. Do you really need your
phones to re-register every 45 seconds?
Indeed, I don't. But need a mechanism to keep NAT sessions alive in case phones are being deployed behind NAT boxes. It seems like transporting SIP over TCP from the phone, plus using TCP keepalives will solve that issue though. I also have Thomson ST2030 phones, and they send SIP OPTIONS packets every 50 seconds for that purpose (all handled by the SER proxy).
Try setting sip registration period to 600 That's 10 minutes.
sip registration renewal timer can be a bit longer - 30 seconds maybe,
or 60 if you are at all worried about reliability.
Here is what I put in the aastra.cfg file from your suggestions :
sip registration period: 3600 sip registration renewal timer: 3565
And that does the trick, that is, the phone registers every 45 seconds. I'll try to update those values to have a larger registration period, rely on TCP keepalives in order to maintain NAT sessions, and have all that NAT handling in SER. Here is the topology :
Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS
ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS
Thanks again!
Philippe
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-- Philippe Sultan
I get you. The good thing with TCP is that most routers have a TCP NAT KeepAlive timer around 1 hour. The onlly issue I see with a 3600s registration timer is if the registrar looses the registration for any reason. It seems some phones do not always try to re-register before the timeout if they loose their registration. Have you tested the Aastra with this ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 10/02/2011 à 17:33, Philippe Sultan a écrit :
Well, I was looking for something like what's implemented in the Thomson ST2030 series, that is, have the phone initiate SIP OPTIONS, and process them with SER.
I was not able to find such a mechanism, but digging in the docs, it appears that I can just use SIP over TCP and keep the connection maintained using TCP keepalive. That wil be just fine I guess.
On the other hand, you can of course have FS send SIP OPTIONS packets to the phones, but I'd like to keep the NAT handling on the front end SIP proxy as much as possible.
Philippe
On Thu, Feb 10, 2011 at 12:23 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
Philippe, can you confirm there is no NAT Keepalive feature in the Aastra ? That's quite weird. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 10/02/2011 à 11:39, Philippe Sultan a écrit :
Tim,
That helped a lot, thanks! More comments inline.
On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre <fs-list@communicatefreely.net> wrote:
I'm wondering about these values:
sip registration retry timer: 10
sip registration renewal timer: 15
sip registration timeout retry timer: 10
sip registration period: 60
sip rport: 1
They seem a bit short, especially on a LAN. Do you really need your
phones to re-register every 45 seconds?
Indeed, I don't. But need a mechanism to keep NAT sessions alive in case phones are being deployed behind NAT boxes. It seems like transporting SIP over TCP from the phone, plus using TCP keepalives will solve that issue though. I also have Thomson ST2030 phones, and they send SIP OPTIONS packets every 50 seconds for that purpose (all handled by the SER proxy).
Try setting sip registration period to 600 That's 10 minutes.
sip registration renewal timer can be a bit longer - 30 seconds maybe,
or 60 if you are at all worried about reliability.
Here is what I put in the aastra.cfg file from your suggestions :
sip registration period: 3600 sip registration renewal timer: 3565
And that does the trick, that is, the phone registers every 45 seconds. I'll try to update those values to have a larger registration period, rely on TCP keepalives in order to maintain NAT sessions, and have all that NAT handling in SER. Here is the topology :
Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS
ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS
Thanks again!
Philippe
_______________________________________________ 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
-- Philippe Sultan
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participants (11)
-
Aloysius Lloyd
-
Aloysius Lloyd
-
Brian West
-
David Ponzone
-
Erik Dekkers
-
Grmt
-
Michael Collins
-
Michal Bielicki
-
Philippe Sultan
-
Thomas Hoellriegel
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Tim St. Pierre