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mthomas at themarketbu...
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 Posted: Tue Mar 17, 2009 6:02 pm Post subject: [Freeswitch-users] Newbie question: Why can't I dial? |
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Hello, everyone.
I am new to Freeswitch, and telephony in general. I am trying to set up a Freeswitch system at work for a project, and I have hit a wall.
I have a dedicated LD T1 from Qwest and a Sangoma A104 card. I believe I have openzap correctly installed in wanpipe mode. I am trying to bridge an incoming SIP call from an IP phone to an openzap channel without success. The Freeswitch log shows that dialing takes place, but the call never completes.
The call log is here: http://pastebin.freeswitch.org/7805
The dialplan xml, openzap.conf, and openzap.conf.xml are here: http://pastebin.freeswitch.org/7806
Any help greatly appreciated.
--Mark
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brian at freeswitch.org
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| Posted: Tue Mar 17, 2009 6:02 pm Post subject: [Freeswitch-users] Newbie question: Why can't I dial? |
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Mark, You should join both #openzap and #freeswitch on irc.freenode.net there are way too many things to go over and the list would just be too slow.
/b
On Mar 17, 2009, at 5:35 PM, Mark Thomas wrote:
| Quote: | Hello, everyone.
I am new to Freeswitch, and telephony in general. I am trying to set up a Freeswitch system at work for a project, and I have hit a wall.
I have a dedicated LD T1 from Qwest and a Sangoma A104 card. I believe I have openzap correctly installed in wanpipe mode. I am trying to bridge an incoming SIP call from an IP phone to an openzap channel without success. The Freeswitch log shows that dialing takes place, but the call never completes.
The call log is here: http://pastebin.freeswitch.org/7805
The dialplan xml, openzap.conf, and openzap.conf.xml are here: http://pastebin.freeswitch.org/7806
Any help greatly appreciated.
--Mark
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msc at freeswitch.org
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| Posted: Tue Mar 17, 2009 6:25 pm Post subject: [Freeswitch-users] Newbie question: Why can't I dial? |
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On Tue, Mar 17, 2009 at 3:35 PM, Mark Thomas
<mthomas@themarketbuilder.com> wrote:
| Quote: | Hello, everyone.
I am new to Freeswitch, and telephony in general. I am trying to set up a Freeswitch system at work for a project, and I have hit a wall.
I have a dedicated LD T1 from Qwest and a Sangoma A104 card. I believe I have openzap correctly installed in wanpipe mode. I am trying to bridge an incoming SIP call from an IP phone to an openzap channel without success. The Freeswitch log shows that dialing takes place, but the call never completes.
The call log is here: http://pastebin.freeswitch.org/7805
The dialplan xml, openzap.conf, and openzap.conf.xml are here: http://pastebin.freeswitch.org/7806
Any help greatly appreciated.
|
Actually I found two things you need to change in the dialplan. What's
happening is that you are telling openzap to dial out span 1, lowest
channel number, but you don't actually give it a phone number to dial.
Here's the current dialplan:
<extension name="outgoing-pri">
<condition field="destination_number" expression="^.+$">
<action application="bridge" data="openzap/1/a"/>
</condition>
first, your expression is a bit dangerous. second, it doesn't actually
"capture" the dialed number. I recommend that you do something like
this:
<condition field="destination_number" expression="^9(\d+)$">
Note the leading nine, the \d+ and the parentheses. Essentially this regex says:
Match any string of digits that begins with a 9 and has at least one
additional digit.
The parens will put the value of (\d+) into the variable $1.
Your bridge then would be this:
<action application="bridge" data="openzap/1/a/$1"/>
Now, reload your dialplan (press F6 or type "reloadxml" at the CLI)
and dial out with a leading 9:
95551212 will send 5551212 to the telco.
Try it and report back!
-MC (IRC: mercutioviz)
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