I have a simple problem with sialing a SIP device. I'm SURE
it's a syntax problem, but I dunno what it might be.
Here are the debug messages:
== Accepting call on 'Zap/1-1' ("PENSACOLA, FL"
<8503846785>)
-- Executing Goto("Zap/1-1", "2111|1") in new stack
-- Goto (default,2111,1)
-- Executing Dial("Zap/1-1", "SIP/18504844535 at
18504844535)") in new stack
WARNING[12299]: File chan_sip.c, Line 411 (create_addr): No such host:
18504844535)
NOTICE[12299]: File app_dial.c, Line 437 (dial_exec): Unable to create channel
of type 'SIP'
== Everyone is busy at this time
-- Executing Answer("Zap/1-1", "") in new stack
DEBUG[12299]: File chan_zap.c, Line 1654 (zt_answer): Took Zap/1-1 off hook
-- Executing Congestion("Zap/1-1", "5") in new stack
Here part of extentions.conf
exten => 2111,1,Dial,SIP/18504844535 at 18504844535
exten => 2111,2,Answer
exten => 2111,3,Congestion
Here is part of sip.conf
[18504844535]
type=friend
host=dynamic
context=default
Does anyone have any suggestions?
--Eric
I think your extensions.conf entry should look like this.> exten => 2111,1,Dial,SIP/18504844535That is how I do it on my server. On Thursday, March 6, 2003, at 12:33 PM, Eric Wieling wrote:> I have a simple problem with sialing a SIP device. I'm SURE > it's a syntax problem, but I dunno what it might be. > > Here are the debug messages: > > == Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>) > -- Executing Goto("Zap/1-1", "2111|1") in new stack > -- Goto (default,2111,1) > -- Executing Dial("Zap/1-1", "SIP/18504844535 at 18504844535)") in > new stack > WARNING[12299]: File chan_sip.c, Line 411 (create_addr): No such host: > 18504844535) > NOTICE[12299]: File app_dial.c, Line 437 (dial_exec): Unable to create > channel of type 'SIP' > == Everyone is busy at this time > -- Executing Answer("Zap/1-1", "") in new stack > DEBUG[12299]: File chan_zap.c, Line 1654 (zt_answer): Took Zap/1-1 off > hook > -- Executing Congestion("Zap/1-1", "5") in new stack > > Here part of extentions.conf > > exten => 2111,1,Dial,SIP/18504844535 at 18504844535 > exten => 2111,2,Answer > exten => 2111,3,Congestion > > Here is part of sip.conf > > [18504844535] > type=friend > host=dynamic > context=default > > > Does anyone have any suggestions? > > --Eric > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users > >Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017
I have some really weird happening with my asterisk setup. I have a
TNT-MAX taking call and forwarding via SIP to an asterisk box (box A).
That asterisk box (box A) forwards the call to another asterisk box (box
B) via iax2. Box B handles all the calls and voice mail and stuff.
When I call into the system and and dial a person extension with
Dial({extension},20,Ttr)
I don't get the ringing, just silence. I want the person to hear a
ringing until the person or the voice mail takes the call.
If I load up my soft phone and call through box A via iax2, I get
relayed to box B and I dial the extension. I hear the ringing, but when
the person answers the call I can talk to them, they can talk to me, but
it is still ringing.
But if I call straight into box B with my soft phone and call the
extension, I hear the ringing and and when the person answers the
ringing goes away.
Peter