There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>
Hi list,
We have a setup with 2 redundant stateful SER SIP-Servers with accounting.
For calls to the PSTN we have 2 ISDN-PRI gateways connected to different ISDN-PRI lines.
We need to implement a dynamic failover mechanism and load balancing.
a) Incoming calls: No problem, the PSTN switch takes care of that b) Outgoing calls: Both GWs are registered at both SERs; calls are usually relayed with "t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover including load-distribution between those gatways? I.e. if I receive "busy here" or no response after a short period after the "invite", the call should be redirected to the other GW.
Thanks in advance for your help!
Best regards, Gerhard
Howdy Adrian!
Do you know when you whing you'll be finnish with this project? or atleast ready for use?
-atle
* Adrian Georgescu ag@ag-projects.com [040902 00:42]:
There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>>
Hi list,
We have a setup with 2 redundant stateful SER SIP-Servers with accounting.
For calls to the PSTN we have 2 ISDN-PRI gateways connected to different ISDN-PRI lines.
We need to implement a dynamic failover mechanism and load balancing.
a) Incoming calls: No problem, the PSTN switch takes care of that b) Outgoing calls: Both GWs are registered at both SERs; calls are usually relayed with "t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover including load-distribution between those gatways? I.e. if I receive "busy here" or no response after a short period after the "invite", the call should be redirected to the other GW.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Adrian/Atle,
We did this using failure routes. Only problem was, wait time ( invite timer ) for both gateways is same as we cant have two timers with different timeout values in ser.
See inline comments.
-Ranga
--- Atle Samuelsen clona@camaro.no wrote:
Howdy Adrian!
Do you know when you whing you'll be finnish with this project? or atleast ready for use?
-atle
- Adrian Georgescu ag@ag-projects.com [040902
00:42]:
There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>>>
Hi list,
We have a setup with 2 redundant stateful SER
SIP-Servers with
accounting.
For calls to the PSTN we have 2 ISDN-PRI gateways
connected to
different ISDN-PRI lines.
We need to implement a dynamic failover mechanism
and load balancing.
a) Incoming calls: No problem, the PSTN switch
takes care of that
b) Outgoing calls: Both GWs are registered at both SERs; calls are
usually relayed with
"t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover
including load-distribution
between those gatways?
Load distribution may require a seperate module/exec script to take routing decisions. When acc is enabled you can query database for active calls. We can quickly check which gateway is handling how many calls and change the host portion of URI before t_relay.
I.e. if I receive "busy here" or no response
after a short period
after the "invite", the call should be redirected
to the other GW.
Busy here can be easily handled in failure route. Having short duration for first call and little longer timeout for next call is not possible at the moment, I guess.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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Hi Ranga (and Adrian)
I was just qurius on when they would be finished with the software. Failure routes is a good thing ;-)
-Atle
* Ranga rrao_v@yahoo.com [040902 07:02]:
Adrian/Atle,
We did this using failure routes. Only problem was, wait time ( invite timer ) for both gateways is same as we cant have two timers with different timeout values in ser.
See inline comments.
-Ranga
--- Atle Samuelsen clona@camaro.no wrote:
Howdy Adrian!
Do you know when you whing you'll be finnish with this project? or atleast ready for use?
-atle
- Adrian Georgescu ag@ag-projects.com [040902
00:42]:
There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>>>>
Hi list,
We have a setup with 2 redundant stateful SER
SIP-Servers with
accounting.
For calls to the PSTN we have 2 ISDN-PRI gateways
connected to
different ISDN-PRI lines.
We need to implement a dynamic failover mechanism
and load balancing.
a) Incoming calls: No problem, the PSTN switch
takes care of that
b) Outgoing calls: Both GWs are registered at both SERs; calls are
usually relayed with
"t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover
including load-distribution
between those gatways?
Load distribution may require a seperate module/exec script to take routing decisions. When acc is enabled you can query database for active calls. We can quickly check which gateway is handling how many calls and change the host portion of URI before t_relay.
I.e. if I receive "busy here" or no response
after a short period
after the "invite", the call should be redirected
to the other GW.
Busy here can be easily handled in failure route. Having short duration for first call and little longer timeout for next call is not possible at the moment, I guess.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
__________________________________ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail
Atle,
We are ready with this except for short period thing. We are not thinking about solving that issue as it is a least important thing for us. Yes, we are looking for the feature to be added in tm module so that we can have configurable ring times for individual calls.
-Ranga
--- Atle Samuelsen clona@camaro.no wrote:
Hi Ranga (and Adrian)
I was just qurius on when they would be finished with the software. Failure routes is a good thing ;-)
-Atle
- Ranga rrao_v@yahoo.com [040902 07:02]:
Adrian/Atle,
We did this using failure routes. Only problem
was,
wait time ( invite timer ) for both gateways is
same
as we cant have two timers with different timeout values in ser.
See inline comments.
-Ranga
--- Atle Samuelsen clona@camaro.no wrote:
Howdy Adrian!
Do you know when you whing you'll be finnish
with
this project? or atleast ready for use?
-atle
- Adrian Georgescu ag@ag-projects.com [040902
00:42]:
There is some work in progress:
Adrian
>>>>>>
Hi list,
We have a setup with 2 redundant stateful SER
SIP-Servers with
accounting.
For calls to the PSTN we have 2 ISDN-PRI
gateways
connected to
different ISDN-PRI lines.
We need to implement a dynamic failover
mechanism
and load balancing.
a) Incoming calls: No problem, the PSTN switch
takes care of that
b) Outgoing calls: Both GWs are registered at both SERs; calls
are
usually relayed with
"t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover
including load-distribution
between those gatways?
Load distribution may require a seperate
module/exec
script to take routing decisions. When acc is
enabled
you can query database for active calls. We can quickly check which gateway is handling how many
calls
and change the host portion of URI before t_relay.
I.e. if I receive "busy here" or no response
after a short period
after the "invite", the call should be
redirected
to the other GW.
Busy here can be easily handled in failure route. Having short duration for first call and little
longer
timeout for next call is not possible at the
moment, I
guess.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
__________________________________ Do you Yahoo!? Yahoo! Mail - 50x more storage than other
providers!
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Hey Atle/Adrian,
I guess I am lost in the discussion. I wasnt talking about the project that Adrian was mentioning. I seem to have overlooked.
-R --- Ranga rrao_v@yahoo.com wrote:
Atle,
We are ready with this except for short period thing. We are not thinking about solving that issue as it is a least important thing for us. Yes, we are looking for the feature to be added in tm module so that we can have configurable ring times for individual calls.
-Ranga
--- Atle Samuelsen clona@camaro.no wrote:
Hi Ranga (and Adrian)
I was just qurius on when they would be finished with the software. Failure routes is a good thing ;-)
-Atle
- Ranga rrao_v@yahoo.com [040902 07:02]:
Adrian/Atle,
We did this using failure routes. Only problem
was,
wait time ( invite timer ) for both gateways is
same
as we cant have two timers with different
timeout
values in ser.
See inline comments.
-Ranga
--- Atle Samuelsen clona@camaro.no wrote:
Howdy Adrian!
Do you know when you whing you'll be finnish
with
this project? or atleast ready for use?
-atle
- Adrian Georgescu ag@ag-projects.com
[040902
00:42]:
There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>>>>>>
Hi list,
We have a setup with 2 redundant stateful
SER
SIP-Servers with
accounting.
For calls to the PSTN we have 2 ISDN-PRI
gateways
connected to
different ISDN-PRI lines.
We need to implement a dynamic failover
mechanism
and load balancing.
a) Incoming calls: No problem, the PSTN
switch
takes care of that
b) Outgoing calls: Both GWs are registered at both SERs;
calls
are
usually relayed with
"t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover
including load-distribution
between those gatways?
Load distribution may require a seperate
module/exec
script to take routing decisions. When acc is
enabled
you can query database for active calls. We can quickly check which gateway is handling how many
calls
and change the host portion of URI before
t_relay.
I.e. if I receive "busy here" or no
response
after a short period
after the "invite", the call should be
redirected
to the other GW.
Busy here can be easily handled in failure
route.
Having short duration for first call and little
longer
timeout for next call is not possible at the
moment, I
guess.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
__________________________________ Do you Yahoo!? Yahoo! Mail - 50x more storage than other
providers!
__________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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December 2004
On Sep 2, 2004, at 6:31 AM, Atle Samuelsen wrote:
Howdy Adrian!
Do you know when you whing you'll be finnish with this project? or atleast ready for use?
-atle
- Adrian Georgescu ag@ag-projects.com [040902 00:42]:
There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>>>
Hi list,
We have a setup with 2 redundant stateful SER SIP-Servers with accounting.
For calls to the PSTN we have 2 ISDN-PRI gateways connected to different ISDN-PRI lines.
We need to implement a dynamic failover mechanism and load balancing.
a) Incoming calls: No problem, the PSTN switch takes care of that b) Outgoing calls: Both GWs are registered at both SERs; calls are usually relayed with "t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover including load-distribution between those gatways? I.e. if I receive "busy here" or no response after a short period after the "invite", the call should be redirected to the other GW.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Ok.
Cool
-Atle
* Adrian Georgescu ag@ag-projects.com [040902 11:18]:
December 2004
On Sep 2, 2004, at 6:31 AM, Atle Samuelsen wrote:
Howdy Adrian!
Do you know when you whing you'll be finnish with this project? or atleast ready for use?
-atle
- Adrian Georgescu ag@ag-projects.com [040902 00:42]:
There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>>>>
Hi list,
We have a setup with 2 redundant stateful SER SIP-Servers with accounting.
For calls to the PSTN we have 2 ISDN-PRI gateways connected to different ISDN-PRI lines.
We need to implement a dynamic failover mechanism and load balancing.
a) Incoming calls: No problem, the PSTN switch takes care of that b) Outgoing calls: Both GWs are registered at both SERs; calls are usually relayed with "t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover including load-distribution between those gatways? I.e. if I receive "busy here" or no response after a short period after the "invite", the call should be redirected to the other GW.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Just out of curiosity, in the document you mention that subsequent requests (ACK, BYE) go to the same gateway, but that is something that would happen anyway because the gateway puts its Contact into reply and record-routing will route subsequent messages to that contact. Do you plan to write any additional support for that or did I just misunderstand the text ?
Jan.
On 02-09 11:18, Adrian Georgescu wrote:
December 2004
On Sep 2, 2004, at 6:31 AM, Atle Samuelsen wrote:
Howdy Adrian!
Do you know when you whing you'll be finnish with this project? or atleast ready for use?
-atle
- Adrian Georgescu ag@ag-projects.com [040902 00:42]:
There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>>>>
Hi list,
We have a setup with 2 redundant stateful SER SIP-Servers with accounting.
For calls to the PSTN we have 2 ISDN-PRI gateways connected to different ISDN-PRI lines.
We need to implement a dynamic failover mechanism and load balancing.
a) Incoming calls: No problem, the PSTN switch takes care of that b) Outgoing calls: Both GWs are registered at both SERs; calls are usually relayed with "t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover including load-distribution between those gatways? I.e. if I receive "busy here" or no response after a short period after the "invite", the call should be redirected to the other GW.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
On Sep 2, 2004, at 12:22 PM, Jan Janak wrote:
Just out of curiosity, in the document you mention that subsequent requests (ACK, BYE) go to the same gateway, but that is something that would happen anyway because the gateway puts its Contact into reply and record-routing will route subsequent messages to that contact. Do you plan to write any additional support for that or did I just misunderstand the text ?
Yes there will be some additional support to make sure same route si followed regardless of how the phone (mis)behaves.
Jan.
On 02-09 11:18, Adrian Georgescu wrote:
December 2004
On Sep 2, 2004, at 6:31 AM, Atle Samuelsen wrote:
Howdy Adrian!
Do you know when you whing you'll be finnish with this project? or atleast ready for use?
-atle
- Adrian Georgescu ag@ag-projects.com [040902 00:42]:
There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>>>>>
Hi list,
We have a setup with 2 redundant stateful SER SIP-Servers with accounting.
For calls to the PSTN we have 2 ISDN-PRI gateways connected to different ISDN-PRI lines.
We need to implement a dynamic failover mechanism and load balancing.
a) Incoming calls: No problem, the PSTN switch takes care of that b) Outgoing calls: Both GWs are registered at both SERs; calls are usually relayed with "t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover including load-distribution between those gatways? I.e. if I receive "busy here" or no response after a short period after the "invite", the call should be redirected to the other GW.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
On 02-09 13:06, Adrian Georgescu wrote:
On Sep 2, 2004, at 12:22 PM, Jan Janak wrote:
Just out of curiosity, in the document you mention that subsequent requests (ACK, BYE) go to the same gateway, but that is something that would happen anyway because the gateway puts its Contact into reply and record-routing will route subsequent messages to that contact. Do you plan to write any additional support for that or did I just misunderstand the text ?
Yes there will be some additional support to make sure same route si followed regardless of how the phone (mis)behaves.
What does that mean ? Record-routing will do the job for you and if it is broken in the phone then there is a really high chance that it will have severe interoperability problems, so why bother with storing additional session state ?
Jan.
Yes Jan, if less work the better.
On Sep 2, 2004, at 1:41 PM, Jan Janak wrote:
On 02-09 13:06, Adrian Georgescu wrote:
On Sep 2, 2004, at 12:22 PM, Jan Janak wrote:
Just out of curiosity, in the document you mention that subsequent requests (ACK, BYE) go to the same gateway, but that is something that would happen anyway because the gateway puts its Contact into reply and record-routing will route subsequent messages to that contact. Do you plan to write any additional support for that or did I just misunderstand the text ?
Yes there will be some additional support to make sure same route si followed regardless of how the phone (mis)behaves.
What does that mean ? Record-routing will do the job for you and if it is broken in the phone then there is a really high chance that it will have severe interoperability problems, so why bother with storing additional session state ?
Jan.
One more thing, I like the E.164 normalization, I would, however, recommend to make it part of the enum module, instead of the LCR module, if you plan to contribute it into the main source tree, of course.
Jan.
On 02-09 11:18, Adrian Georgescu wrote:
December 2004
On Sep 2, 2004, at 6:31 AM, Atle Samuelsen wrote:
Howdy Adrian!
Do you know when you whing you'll be finnish with this project? or atleast ready for use?
-atle
- Adrian Georgescu ag@ag-projects.com [040902 00:42]:
There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>>>>
Hi list,
We have a setup with 2 redundant stateful SER SIP-Servers with accounting.
For calls to the PSTN we have 2 ISDN-PRI gateways connected to different ISDN-PRI lines.
We need to implement a dynamic failover mechanism and load balancing.
a) Incoming calls: No problem, the PSTN switch takes care of that b) Outgoing calls: Both GWs are registered at both SERs; calls are usually relayed with "t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover including load-distribution between those gatways? I.e. if I receive "busy here" or no response after a short period after the "invite", the call should be redirected to the other GW.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
E164 normalization I was referring into the document to has (almost) nothing to do with ENUM. It has to do with a consistent table with destination prefixes in PSTN (country codes e.g. 49171 geman mobile/ network numbers to make it more clear). The destinations have names and prices you might link to/from external systems. The normalization means all PSTN calls will be brought to a normalized form according to E164 standard. If you dial extra 9 before pst numbers, local numbers without country codes or not or you send to the gateway the country prefix or add two zero is totally up to each configuration at each provider. LCR makes sure it looks up the destination from one consistent table which matches what Telco today are used to in building a dest/cost database.
Adrian
On Sep 2, 2004, at 12:32 PM, Jan Janak wrote:
One more thing, I like the E.164 normalization, I would, however, recommend to make it part of the enum module, instead of the LCR module, if you plan to contribute it into the main source tree, of course.
Jan.
On 02-09 11:18, Adrian Georgescu wrote:
December 2004
On Sep 2, 2004, at 6:31 AM, Atle Samuelsen wrote:
Howdy Adrian!
Do you know when you whing you'll be finnish with this project? or atleast ready for use?
-atle
- Adrian Georgescu ag@ag-projects.com [040902 00:42]:
There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>>>>>
Hi list,
We have a setup with 2 redundant stateful SER SIP-Servers with accounting.
For calls to the PSTN we have 2 ISDN-PRI gateways connected to different ISDN-PRI lines.
We need to implement a dynamic failover mechanism and load balancing.
a) Incoming calls: No problem, the PSTN switch takes care of that b) Outgoing calls: Both GWs are registered at both SERs; calls are usually relayed with "t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover including load-distribution between those gatways? I.e. if I receive "busy here" or no response after a short period after the "invite", the call should be redirected to the other GW.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Yes, I understand that. I find the the idea of using timezones to create the prefix interesting, other stuff can be more or less achieved using generic rewrite rules. In my opinion it would be nice to have that separated in some way, so that even users who do not need least cost routing (like me) can use the code without the need to load all additional stuff.
Jan.
On 02-09 13:13, Adrian Georgescu wrote:
E164 normalization I was referring into the document to has (almost) nothing to do with ENUM. It has to do with a consistent table with destination prefixes in PSTN (country codes e.g. 49171 geman mobile/ network numbers to make it more clear). The destinations have names and prices you might link to/from external systems. The normalization means all PSTN calls will be brought to a normalized form according to E164 standard. If you dial extra 9 before pst numbers, local numbers without country codes or not or you send to the gateway the country prefix or add two zero is totally up to each configuration at each provider. LCR makes sure it looks up the destination from one consistent table which matches what Telco today are used to in building a dest/cost database.
Adrian
On Sep 2, 2004, at 12:32 PM, Jan Janak wrote:
One more thing, I like the E.164 normalization, I would, however, recommend to make it part of the enum module, instead of the LCR module, if you plan to contribute it into the main source tree, of course.
Jan.
On 02-09 11:18, Adrian Georgescu wrote:
December 2004
On Sep 2, 2004, at 6:31 AM, Atle Samuelsen wrote:
Howdy Adrian!
Do you know when you whing you'll be finnish with this project? or atleast ready for use?
-atle
- Adrian Georgescu ag@ag-projects.com [040902 00:42]:
There is some work in progress:
http://download.dns-hosting.info/SERLCR/README
Adrian
>>>>>>>
Hi list,
We have a setup with 2 redundant stateful SER SIP-Servers with accounting.
For calls to the PSTN we have 2 ISDN-PRI gateways connected to different ISDN-PRI lines.
We need to implement a dynamic failover mechanism and load balancing.
a) Incoming calls: No problem, the PSTN switch takes care of that b) Outgoing calls: Both GWs are registered at both SERs; calls are usually relayed with "t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover including load-distribution between those gatways? I.e. if I receive "busy here" or no response after a short period after the "invite", the call should be redirected to the other GW.
Thanks in advance for your help!
Best regards, Gerhard
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
hello,Jan I meet some trouble when I use SER .I hope that when the call timeouts or is declined by the callee the call would be forward to SEMS. In my ser.cfg ,I wrote : if (method=="INVITE") { t_on_failure("1"); } But when the caller cancels the call,487 reply the callee sends aslo make INVITE enter the failure_route. I don't export it. So how can I distinguish 487 and 603 ?
Thanks for your help
Wei
You can use t_check_status("487") to check the code in failure_route[1].
Richard
-----Original Message----- From: serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Zhang Wei Sent: Friday, September 03, 2004 8:30 PM To: Jan Janak Cc: serusers@lists.iptel.org Subject: [Serusers] How can I distinguish 487 and 603 reply?
hello,Jan I meet some trouble when I use SER .I hope that when the call timeouts or is declined by the callee the call would be forward to SEMS. In my ser.cfg ,I wrote : if (method=="INVITE") { t_on_failure("1"); } But when the caller cancels the call,487 reply the callee sends aslo make INVITE enter the failure_route. I don't export it. So how can I distinguish 487 and 603 ?
Thanks for your help
Wei
Hi I have the same problem, please take a look into this report: http://lists.iptel.org/pipermail/serusers/2004-August/010930.html
...if the called-user belongs to the voicemail group and the caller-user cancels the call I see irregular behavior in both ser instances and no Sip-Response-Code=487 is recorded in radius acc (just an start record without stop).
I´ve just tried with "t_check_status" but I have the same problem... please send us some advice...
thank you Rafael
ser.cfg:: # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc if (is_user_in("Request-URI", "voicemail")) { setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found route(4); break; }; # if user is on-line and is in voicemail group, enable redirection if (method == "INVITE" && isflagset(4)) { t_on_failure("1"); }; t_relay(); } # ------------- handling of unavailable user ------------------ route[4] { # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL" || method=="BYE")) { sl_send_reply("404", "Not Found"); acc_rad_request("404 Not Found"); break; }; # not voicemail subscriber if (!isflagset(4)) { sl_send_reply("404", "Not Found and no voicemail turned on"); acc_rad_request("404 Not Found"); break; }; # forward to voicemail now rewritehostport("call.millicom.com.pe:5090"); t_relay_to_udp("call.millicom.com.pe", "5090"); } # if forwarding downstream did not succeed, try voicemail running # at bat.iptel.org:5090 failure_route[1]{ if (t_check_status("408|486|487")){ revert_uri (); rewritehostport ("call.millicom.com.pe:5090"); append_branch(); t_relay(); break; } }
Richard richard@o-matrix.org wrote: You can use t_check_status("487") to check the code in failure_route[1].
Richard
-----Original Message----- From: serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Zhang Wei Sent: Friday, September 03, 2004 8:30 PM To: Jan Janak Cc: serusers@lists.iptel.org Subject: [Serusers] How can I distinguish 487 and 603 reply?
hello,Jan I meet some trouble when I use SER .I hope that when the call timeouts or is declined by the callee the call would be forward to SEMS. In my ser.cfg ,I wrote : if (method=="INVITE") { t_on_failure("1"); } But when the caller cancels the call,487 reply the callee sends aslo make INVITE enter the failure_route. I don't export it. So how can I distinguish 487 and 603 ?
Thanks for your help
Wei
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Adrian Georgescu -
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Jan Janak -
Rafael J. Risco G.V -
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Zhang Wei