Alright I have a SIP phone located off premises with a very annoying
issue.
Well I say a sip phone it is actually two phones hooked to a Cisco Spa
2102
Link: http://www.cisco.com/en/US/products/ps10026/index.html
Each phone being a different line/extension.
Alright either line can ALWAYS make outbound calls no issue. The problem
is on the Inbound side. I'm completely stumped as to why. I could make
10 back to back out bound calls and then call inbound and watch the call
come in to * and try to be passed to the sip phone only to get "Error
Message 14: Not a Working Number." So it doesn't seem to be a matter of
the phones Sip Login "Timing out"
And when I check sip peers it shows the correct IP address of the box so
it doesn't appear to be that it can't find the Cisco box.
Here is what I use for the inbound context, replacing the _X_ with the
actual extension of course.
[to_ddwhome]
exten=> _X_,1,wait(1)
exten=> _X_,n,Dial(${ddwhome},21)
exten=> _X_,n,Goto(dial_inf,${EXTEN},1)
${ddwhome}=SIP/ddwhome
Now the odd thing is when it gets the Error 14 message then the third
step to dial_inf does not execute. Though when it rarely does connect
with the sip phone if no one answers in 21 seconds than it will roll
over to that step.
Any ideas?
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
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On Mon, 2009-12-28 at 12:11 -0600, James A. Shigley wrote:> Alright I have a SIP phone located off premises with a very annoying > issue. > > > > Well I say a sip phone it is actually two phones hooked to a Cisco Spa > 2102 > > Link: http://www.cisco.com/en/US/products/ps10026/index.html >Looks pretty much like the PAP2 which I have running flawlessly with 1.6 in and outbound - so don't despair, you can solve this.> > Each phone being a different line/extension. > > > > Alright either line can ALWAYS make outbound calls no issue. The > problem is on the Inbound side. I?m completely stumped as to why. I > could make 10 back to back out bound calls and then call inbound and > watch the call come in to * and try to be passed to the sip phone only > to get ?Error Message 14: Not a Working Number.? So it doesn?t seem to > be a matter of the phones Sip Login ?Timing out? > > > > And when I check sip peers it shows the correct IP address of the box > so it doesn?t appear to be that it can?t find the Cisco box. > > > > Here is what I use for the inbound context, replacing the _X_ with the > actual extension of course. > > > > [to_ddwhome] > > exten=> _X_,1,wait(1) > > exten=> _X_,n,Dial(${ddwhome},21) > > exten=> _X_,n,Goto(dial_inf,${EXTEN},1) > > > > ${ddwhome}=SIP/ddwhome > > > > Now the odd thing is when it gets the Error 14 message then the third > step to dial_inf does not execute. Though when it rarely does connect > with the sip phone if no one answers in 21 seconds than it will roll > over to that step. > > > > Any ideas? > > > > James Shigley >Probably be useful to see sip.conf as well and know the version of Asterisk you are running but in passing, you don't have any firewall rules that could stop your asterisk talking TO the Cisco when something comes in? The one minor issue I had with mine is my router has some NAT issues with signalling (It's a Draytek - they are known for it). In the end I shifted the PAP2 up to 5061/5062 and the problem was gone. None of this may be useful to you but I'll tell you this much. In my few weeks with Asterisk I've had times where I've asked myself why certain things would plain refuse to work and on every occasion it was *not* the fault of Asterisk. 50% my config, 40% my network, 10% different docs for different versions and missing info ;-)
Is ddwhome defined in global context?? If so, then you should use global function. Paste asterisk log to check. Saludos, Juan E. Rodr?guez -----Original Message----- From: "James A. Shigley" <jas at answeringserv.com> Date: Mon, 28 Dec 2009 12:11:35 To: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users at lists.digium.com> Subject: [asterisk-users] SIP Issue _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users