Posted: Mon May 18, 2009 3:42 pm Post subject: [asterisk-dev] Version Dial() comma separation was introduce
Does anyone know which stable version(s) of Asterisk introduced using
commas rather than "|" to separate the arguments for Dial()? I'd like to
run a diff between it and the previous version.
Posted: Mon May 18, 2009 3:50 pm Post subject: [asterisk-dev] Version Dial() comma separation was introduce
On Monday 18 May 2009 11:33:48 am Alistair Cunningham wrote:
Quote:
Does anyone know which stable version(s) of Asterisk introduced using
commas rather than "|" to separate the arguments for Dial()? I'd like to
run a diff between it and the previous version.
1.6.0
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Tilghman
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Posted: Mon May 18, 2009 3:56 pm Post subject: [asterisk-dev] Version Dial() comma separation was introduce
Alistair Cunningham wrote:
Quote:
Does anyone know which stable version(s) of Asterisk introduced using
commas rather than "|" to separate the arguments for Dial()? I'd like to
run a diff between it and the previous version.
All versions accepted both commas and pipes. Asterisk 1.6.X is what
requires commas.
--
Russell Bryant
Digium, Inc. | Senior Software Engineer, Open Source Team Lead
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
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Russell Bryant
Digium, Inc. | Senior Software Engineer, Open Source Team Lead
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
_______________________________________________
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Posted: Mon May 18, 2009 4:03 pm Post subject: [asterisk-dev] Version Dial() comma separation was introduce
Tilghman Lesher wrote:
Quote:
On Monday 18 May 2009 11:33:48 am Alistair Cunningham wrote:
> Does anyone know which stable version(s) of Asterisk introduced using
> commas rather than "|" to separate the arguments for Dial()? I'd like to
> run a diff between it and the previous version.
1.6.0
Thanks, that's what I feared. Looks like I'm going to need to get my
hands dirty rather than using diff.
Posted: Mon May 18, 2009 4:26 pm Post subject: [asterisk-dev] Version Dial() comma separation was introduce
Alistair Cunningham wrote:
Quote:
Tilghman Lesher wrote:
> On Monday 18 May 2009 11:33:48 am Alistair Cunningham wrote:
>> Does anyone know which stable version(s) of Asterisk introduced using
>> commas rather than "|" to separate the arguments for Dial()? I'd like to
>> run a diff between it and the previous version.
>
> 1.6.0
Thanks, that's what I feared. Looks like I'm going to need to get my
hands dirty rather than using diff.
The reason I ask is that the parser seems to behave differently between
1.4.21.1 and 1.6.1.0. Here's from 1.6.1.0:
<SIP/enswitch-local-0826d558>AGI Rx << EXEC Dial
SIP/0123456789@5-sip.example.com,600,oL(86398000:60000)
-- AGI Script Executing Application: (Dial) Options:
(SIP/0123456789@5-sip.example.com,600,oL(86398000:60000))
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
[May 18 18:03:05] WARNING[24557]: chan_sip.c:4526 create_addr: No such
host: 5-sip.example.com,600,ooL(86398000
[May 18 18:03:05] WARNING[24557]: app_dial.c:1518 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
With 1.4.21.1, "EXEC Dial
SIP/0123456789@5-sip.example.com|600|oL(86398000:60000)" worked. The
peer does exist:
mysql> select name from sippeers where name='5-sip.example.com';
+-------------------+
| name |
+-------------------+
| 5-sip.example.com |
+-------------------+
1 row in set (0.00 sec)
Perhaps the allowed characters in a peer name have changed? I'll
continue looking at the source for app_dial.c to see if I can spot it.
Posted: Mon May 18, 2009 4:40 pm Post subject: [asterisk-dev] Version Dial() comma separation was introduce
On Monday 18 May 2009 11:54:37 am Alistair Cunningham wrote:
Quote:
Tilghman Lesher wrote:
> On Monday 18 May 2009 11:33:48 am Alistair Cunningham wrote:
>> Does anyone know which stable version(s) of Asterisk introduced using
>> commas rather than "|" to separate the arguments for Dial()? I'd like to
>> run a diff between it and the previous version.
>
> 1.6.0
Thanks, that's what I feared. Looks like I'm going to need to get my
hands dirty rather than using diff.
Posted: Mon May 18, 2009 5:36 pm Post subject: [asterisk-dev] Version Dial() comma separation was introduce
Tilghman Lesher wrote:
Quote:
On Monday 18 May 2009 11:54:37 am Alistair Cunningham wrote:
> Tilghman Lesher wrote:
>> On Monday 18 May 2009 11:33:48 am Alistair Cunningham wrote:
>>> Does anyone know which stable version(s) of Asterisk introduced using
>>> commas rather than "|" to separate the arguments for Dial()? I'd like to
>>> run a diff between it and the previous version.
>> 1.6.0
> Thanks, that's what I feared. Looks like I'm going to need to get my
> hands dirty rather than using diff.
Posted: Mon May 18, 2009 5:46 pm Post subject: [asterisk-dev] Version Dial() comma separation was introduce
On Monday 18 May 2009 12:09:14 pm Alistair Cunningham wrote:
Quote:
Alistair Cunningham wrote:
> Tilghman Lesher wrote:
>> On Monday 18 May 2009 11:33:48 am Alistair Cunningham wrote:
>>> Does anyone know which stable version(s) of Asterisk introduced using
>>> commas rather than "|" to separate the arguments for Dial()? I'd like
>>> to run a diff between it and the previous version.
>>
>> 1.6.0
>
> Thanks, that's what I feared. Looks like I'm going to need to get my
> hands dirty rather than using diff.
The reason I ask is that the parser seems to behave differently between
1.4.21.1 and 1.6.1.0. Here's from 1.6.1.0:
<SIP/enswitch-local-0826d558>AGI Rx << EXEC Dial
SIP/0123456789@5-sip.example.com,600,oL(86398000:60000)
-- AGI Script Executing Application: (Dial) Options:
(SIP/0123456789@5-sip.example.com,600,oL(86398000:60000))
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
[May 18 18:03:05] WARNING[24557]: chan_sip.c:4526 create_addr: No such
host: 5-sip.example.com,600,ooL(86398000
[May 18 18:03:05] WARNING[24557]: app_dial.c:1518 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
With 1.4.21.1, "EXEC Dial
SIP/0123456789@5-sip.example.com|600|oL(86398000:60000)" worked. The
peer does exist:
My guess is that you don't have a [compat] section in your asterisk.conf.
Starting in 1.6.0, Asterisk looks for a value called res_agi=1.6 in this
section. It defaults to the value "1.4" in 1.6.0, though if you clean
install, it sets the value as "1.6". This value governs which delimiter
AGI uses.
This section and value were created at the direct request of users who thought
it was confusing that their 1.4 scripts no longer worked correctly in 1.6.0
and wanted a compatibility setting.
--
Tilghman
_______________________________________________
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Posted: Mon May 18, 2009 5:58 pm Post subject: [asterisk-dev] Version Dial() comma separation was introduce
Alistair Cunningham wrote:
Quote:
Alistair Cunningham wrote:
> Tilghman Lesher wrote:
>> On Monday 18 May 2009 11:33:48 am Alistair Cunningham wrote:
>>> Does anyone know which stable version(s) of Asterisk introduced using
>>> commas rather than "|" to separate the arguments for Dial()? I'd like to
>>> run a diff between it and the previous version.
>> 1.6.0
> Thanks, that's what I feared. Looks like I'm going to need to get my
> hands dirty rather than using diff.
The reason I ask is that the parser seems to behave differently between
1.4.21.1 and 1.6.1.0. Here's from 1.6.1.0:
<SIP/enswitch-local-0826d558>AGI Rx << EXEC Dial
SIP/0123456789@5-sip.example.com,600,oL(86398000:60000)
-- AGI Script Executing Application: (Dial) Options:
(SIP/0123456789@5-sip.example.com,600,oL(86398000:60000))
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
[May 18 18:03:05] WARNING[24557]: chan_sip.c:4526 create_addr: No such
host: 5-sip.example.com,600,ooL(86398000
[May 18 18:03:05] WARNING[24557]: app_dial.c:1518 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
With 1.4.21.1, "EXEC Dial
SIP/0123456789@5-sip.example.com|600|oL(86398000:60000)" worked. The
peer does exist:
mysql> select name from sippeers where name='5-sip.example.com';
+-------------------+
| name |
+-------------------+
| 5-sip.example.com |
+-------------------+
1 row in set (0.00 sec)
Perhaps the allowed characters in a peer name have changed? I'll
continue looking at the source for app_dial.c to see if I can spot it.
This looks like the offending code (from handle_exec in res_agi.c):
The first if statement will escape any commas it comes across in the options.
Interestingly, it appears that if you were to use vertical bars as you showed
with 1.4.21.1, you would likely see a successful operation since they are
replaced with commas.
Was it intended that AGI commands would still use vertical bars as option
separators?
Mark Michelson
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Posted: Mon May 18, 2009 5:58 pm Post subject: [asterisk-dev] Version Dial() comma separation was introduce
Tilghman Lesher wrote:
Quote:
My guess is that you don't have a [compat] section in your asterisk.conf.
Starting in 1.6.0, Asterisk looks for a value called res_agi=1.6 in this
section. It defaults to the value "1.4" in 1.6.0, though if you clean
install, it sets the value as "1.6". This value governs which delimiter
AGI uses.
This section and value were created at the direct request of users who thought
it was confusing that their 1.4 scripts no longer worked correctly in 1.6.0
and wanted a compatibility setting.
Yup, that was it. Thank you, and sorry to bother -dev with a question
that turned out to be -users.
Posted: Mon May 18, 2009 6:07 pm Post subject: [asterisk-dev] Version Dial() comma separation was introduce
Alistair Cunningham schrieb:
Quote:
Does anyone know which stable version(s) of Asterisk introduced using
commas rather than "|" to separate the arguments for Dial()? I'd like to
run a diff between it and the previous version.
---cut---
--- in Asterisk 1.4
+++ in Asterisk 1.6
@@ -4,7 +4,7 @@
Place a call and connect to the current channel
[Description]
- Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]):
+ Dial(Technology/resource[&Tech2/resource2...][,timeout][,options][,URL]):
This application will place calls to one or more specified channels. As soon
as one of the requested channels answers, the originating channel will be
answered, if it has not already been answered. These two channels will then
@@ -39,6 +39,8 @@
Options:
A(x) - Play an announcement to the called party, using 'x' as the file.
C - Reset the CDR for this call.
+ c - If DIAL cancels this call, always set the flag to tell the channel
+ driver that the call is answered elsewhere.
d - Allow the calling user to dial a 1 digit extension while waiting for
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the EXITCONTEXT variable,
@@ -48,10 +50,14 @@
DTMF string is sent to the called party, and the 'calling' DTMF
string is sent to the calling party. Both parameters can be used
alone.
+ e - execute the 'h' extension for peer after the call ends. This
+ operation will not be performed if the peer was parked
f - Force the callerid of the *calling* channel to be set as the
extension associated with the channel using a dialplan 'hint'.
For example, some PSTNs do not allow CallerID to be set to anything
other than the number assigned to the caller.
+ F(context^exten^pri) - When the caller hangs up, transfer the called party
+ to the specified context and extension and continue execution.
g - Proceed with dialplan execution at the current extension if the
destination channel hangs up.
G(context^exten^pri) - If the call is answered, transfer the calling party to
@@ -59,15 +65,18 @@
Optionally, an extension, or extension and context may be specified.
Otherwise, the current extension is used. You cannot use any additional
action post answer options in conjunction with this option.
- h - Allow the called party to hang up by sending the '*' DTMF digit.
- H - Allow the calling party to hang up by hitting the '*' DTMF digit.
+ h - Allow the called party to hang up by sending the '*' DTMF digit, or
+ whatever sequence was defined in the featuremap section for
+ 'disconnect' in features.conf
+ H - Allow the calling party to hang up by hitting the '*' DTMF digit, or
+ whatever sequence was defined in the featuremap section for
+ 'disconnect' in features.conf
i - Asterisk will ignore any forwarding requests it may receive on this
dial attempt.
- j - Jump to priority n+101 if all of the requested channels were busy.
k - Allow the called party to enable parking of the call by sending
- the DTMF sequence defined for call parking in features.conf.
+ the DTMF sequence defined for call parking in the featuremap section of features.conf.
K - Allow the calling party to enable parking of the call by sending
- the DTMF sequence defined for call parking in features.conf.
+ the DTMF sequence defined for call parking in the featuremap section of features.conf.
L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are
left. Repeat the warning every 'z' ms. The following special
variables can be used with this option:
@@ -84,14 +93,12 @@
specified.
M(x[^arg]) - Execute the Macro for the *called* channel before connecting
to the calling channel. Arguments can be specified to the Macro
- using '^' as a delimeter. The Macro can set the variable
+ using '^' as a delimiter. The Macro can set the variable
MACRO_RESULT to specify the following actions after the Macro is
finished executing.
* ABORT Hangup both legs of the call.
* CONGESTION Behave as if line congestion was encountered.
- * BUSY Behave as if a busy signal was encountered. This will also
- have the application jump to priority n+101 if the
- 'j' option is set.
+ * BUSY Behave as if a busy signal was encountered.
* CONTINUE Hangup the called party and allow the calling party
to continue dialplan execution at the next priority.
* GOTO:<context>^<exten>^<priority> - Transfer the call to the
@@ -108,8 +115,8 @@
o - Specify that the CallerID that was present on the *calling* channel
be set as the CallerID on the *called* channel. This was the
behavior of Asterisk 1.0 and earlier.
- O([x]) - "Operator Services" mode (Zaptel channel to Zaptel channel
- only, if specified on non-Zaptel interface, it will be ignored).
+ O([x]) - "Operator Services" mode (DAHDI channel to DAHDI channel
+ only, if specified on non-DAHDI interface, it will be ignored).
When the destination answers (presumably an operator services
station), the originator no longer has control of their line.
They may hang up, but the switch will not release their line
@@ -128,10 +135,35 @@
S(x) - Hang up the call after 'x' seconds *after* the called party has
answered the call.
t - Allow the called party to transfer the calling party by sending the
- DTMF sequence defined in features.conf.
+ DTMF sequence defined in the blindxfer setting in the featuremap section
+ of features.conf.
T - Allow the calling party to transfer the called party by sending the
- DTMF sequence defined in features.conf.
+ DTMF sequence defined in the blindxfer setting in the featuremap section
+ of features.conf.
+ U(x[^arg]) - Execute via Gosub the routine 'x' for the *called* channel before connecting
+ to the calling channel. Arguments can be specified to the Gosub
+ using '^' as a delimiter. The Gosub routine can set the variable
+ GOSUB_RESULT to specify the following actions after the Gosub returns.
+ * ABORT Hangup both legs of the call.
+ * CONGESTION Behave as if line congestion was encountered.
+ * BUSY Behave as if a busy signal was encountered.
+ * CONTINUE Hangup the called party and allow the calling party
+ to continue dialplan execution at the next priority.
+ * GOTO:<context>^<exten>^<priority> - Transfer the call to the
+ specified priority. Optionally, an extension, or
+ extension and priority can be specified.
+ You cannot use any additional action post answer options in conjunction
+ with this option. Also, pbx services are not run on the peer (called) channel,
+ so you will not be able to set timeouts via the TIMEOUT() function in this routine.
w - Allow the called party to enable recording of the call by sending
- the DTMF sequence defined for one-touch recording in features.conf.
+ the DTMF sequence defined in the automon setting in the featuremap section
+ of features.conf.
W - Allow the calling party to enable recording of the call by sending
- the DTMF sequence defined for one-touch recording in features.conf.
+ the DTMF sequence defined in the automon setting in the featuremap section
+ of features.conf.
+ x - Allow the called party to enable recording of the call by sending
+ the DTMF sequence defined in the automixmon setting in the featuremap section
+ of features.conf.
+ X - Allow the calling party to enable recording of the call by sending
+ the DTMF sequence defined in the automixmon setting in the featuremap section
+ of features.conf.
---cut---
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