Routing tips from various users - Revision history

Monica: /* Not letting things go on forever */

2013-10-31T15:07:45Z

‎Not letting things go on forever

2013-08-22T11:23:47Z

‎Redirecting Calls to another User in the Regex Module

2013-08-22T11:23:33Z

‎Regex-Routing

2013-08-19T07:29:11Z

‎Not letting things go on forever

2013-08-16T08:51:07Z

‎Slow devices

2013-08-16T08:50:24Z

‎Setting up outgoing number for each user

2013-08-16T08:47:00Z

‎Setting up outgoing number for each user:

2013-08-16T08:42:11Z

2013-08-16T08:41:14Z

‎Modifying call information

2013-08-16T08:40:42Z

← Older revision		Revision as of 15:07, 31 October 2013
Line 113:		Line 113:
	* [[Routing]]		* [[Routing]]
	* [[Regular expressions]]		* [[Regular expressions]]
		+
		+	[[Category:Routing]] [[Category:Tips]] [[Category:Regexroute]]

← Older revision		Revision as of 11:23, 22 August 2013
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	====Redirecting Calls to another User in the Regex Module====		====Redirecting Calls to another User in the Regex Module====
		+
		+	* I use the full international format to assign unique numbers to my customers (Their fixed line numbers). If regexroute.conf has the highest actual priority, these customers always will be routed using the external provider. I could add rules to route them, by redirecting:

	In regexroute.conf:		In regexroute.conf:

@@ Line 28: / Line 28: @@
   ^0007$=wave/record//root/recordings/7.slin
   ^00077$=wave/play//root/recordings/7.slin
 ====Modifying call information====

← Older revision		Revision as of 07:29, 19 August 2013
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	^voicemail$=sip/sip:voicemail@example.com;caller=originator;callername=;maxcall=10000,timeout=180000		^voicemail$=sip/sip:voicemail@example.com;caller=originator;callername=;maxcall=10000,timeout=180000
		+
		+	'''See also'''
		+	* [[Routing]]
		+	* [[Regular expressions]]

@@ Line 102: / Line 102: @@
 Timeout runs for the overall length of the call. Whereas is maxcall is how long YATE will wait until the other side responds. Both parameters are in units of milliseconds. So, to wait up to 10 seconds before cancelling the call to the voicemail system, we can change our line in regexroute.conf to be:
   ^voicemail$=sip/sip:voicemail@example.com;caller=originator;callername=;maxcall=10000
 ====Not letting things go on forever====

← Older revision		Revision as of 08:50, 16 August 2013
Line 95:		Line 95:
	${username}^1001$=;caller=234561		${username}^1001$=;caller=234561
	^9\(.*\)$=h323/\1@xxx.xxx.xxx.xxx		^9\(.*\)$=h323/\1@xxx.xxx.xxx.xxx
		+
		+	====Slow devices====
		+
		+	I discovered that a few times I'd call my voicemail machine and YATE would go ahead and cancel the session just prior to establishing it. It turned out that that the voicemail machine was somewhat loaded and things were not responding in time. For each call that YATE initiates, it starts two timers: maxcall and timeout.
		+
		+	Timeout runs for the overall length of the call. Whereas is maxcall is how long YATE will wait until the other side responds. Both parameters are in units of milliseconds. So, to wait up to 10 seconds before cancelling the call to the voicemail system, we can change our line in regexroute.conf to be:
		+
		+
		+	^voicemail$=sip/sip:voicemail@example.com;caller=originator;callername=;maxcall=10000
		+
		+
		+	====Not letting things go on forever====
		+
		+	Sometime the devices will be slow and sometimes the people using those devices will be slow (at talking). If you wish you can constrain the duration of a call as well. This is known as the timeout. So, to not let people leave voicemails longer than 180 seconds (3 minutes) you would have:
		+
		+	^voicemail$=sip/sip:voicemail@example.com;caller=originator;callername=;maxcall=10000,timeout=180000

@@ Line 86: / Line 86: @@
 The purpose of this setup is to distribute the balance of the calls across the two providers; Also it gives callers the possibility to try again with a different provider (well both are Betamax actually, but ...)
-====Setting up outgoing number for each user:====
+====Setting up outgoing number for each user====
   [default]

@@ Line 30: / Line 30: @@
 * I use the full international format to assign unique numbers to my customers (Their fixed line numbers). If regexroute.conf has the highest actual priority, these customers always will be routed using the external provider. I could add rules to route them, by redirecting:
 ====Redirecting Calls to another User in the Regex Module====
@@ Line 75: / Line 95: @@
   ${username}^1001$=;caller=234561
   ^9\(.*\)$=h323/\1@xxx.xxx.xxx.xxx

@@ Line 78: / Line 78: @@
 ====Modifying call information====
-In to properly do this, I had to modify the sending information. Initially the SIP message looked like (anonymised):
+To do this I had to modify the sending information. Initially the SIP message looked like (anonymised):
   INVITE sip:voicemail@example.com SIP/2.0

← Older revision		Revision as of 08:40, 16 August 2013
Line 75:		Line 75:
	${username}^1001$=;caller=234561		${username}^1001$=;caller=234561
	^9\(.*\)$=h323/\1@xxx.xxx.xxx.xxx		^9\(.*\)$=h323/\1@xxx.xxx.xxx.xxx
		+
		+	====Modifying call information====
		+
		+	In to properly do this, I had to modify the sending information. Initially the SIP message looked like (anonymised):
		+
		+	INVITE sip:voicemail@example.com SIP/2.0
		+	From: "stuff" <sip:random@10.0.0.1>;tag=108375482
		+	To: <sip:voicemail@example.com>
		+
		+	What I wanted to do, was change it into:
		+
		+	INVITE sip:voicemail@example.com SIP/2.0
		+	From: <sip:originator@10.0.0.1>;tag=1402492972
		+	To: <sip:voicemail@example.com>
		+
		+	To change "stuff" into something else, or remove it entirely, you can use the callername property. To change 'random' into 'originator' you set the caller property. Combining these together gives you:
		+
		+	^voicemail$=sip/sip:voicemail@example.com;caller=originator;callername=
		+
		+	which you can place in regexroute.conf