[asterisk-users] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host

[Paul Hales]

The fact that all of the phones have the same 'host' is not a good sign.

Also - turn 'qualify' on. It really helps with phone status.

PaulH

On Sat, 2007-02-03 at 12:50 -0500, Erick Perez wrote:
> The following strange conditions is happening while I try to dial a
> SIP user from another SIp user.
> SIP to Zap dialing is fine, as all 4 users can call PSTN.
> I'm using Asterisk SVN-branch-1.2-r51359M
> 
> Example: extension 3210 calls extension 3213. They are all registered properly:
> chrom01*CLI> sip show peers
> Name/username              Host            Dyn Nat ACL Port     Status
> 3213/3213                  192.168.0.112    D          5060     Unmonitored
> 3212/3212                  192.168.0.112    D          5060     Unmonitored
> 3211/3211                  192.168.0.112    D          5060     Unmonitored
> 3210/3210                  192.168.0.112    D          5060     Unmonitored
> 4 sip peers [4 online , 0 offline]
> 
>     -- Executing Ringing("SIP/3210-084eaa80", "") in new stack
>     -- Executing AGI("SIP/3210-084eaa80",
> "agi://127.0.0.1:4577/call_log") in new stack
>     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
>     -- Executing Dial("SIP/3210-084eaa80", "SIP/3213)|30|to") in new stack
> Feb  3 12:42:25 WARNING[10368]: chan_sip.c:1994 create_addr: No such host: 3213)
> Feb  3 12:42:25 NOTICE[10368]: app_dial.c:1056 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 3 - No route to destination)
>   == Everyone is busy/congested at this time (1:0/0/1)
> 
> **sip.conf***
> **************
> i have 4 extensions, 3210,3211,3212 and 3213. they are all defined in
> sip.conf with the following parameters (just change 3212 for the next
> extension and so on).
> [3212]
> username=3212
> secret=3212
> type=friend
> context=default
> nat=no
> canreinvite=no
> mailbox=3212 at default
> disallow=all
> allow=ulaw
> host=dynamic
> language=en
> dtmfmode=inband
> 
> My dial plan is like this:
> The AGI is doing nothing more than simple call logging to MySQL
> **extensions.conf**
> **********************
> exten => _321[0123],1,Ringing
> exten => _321[0123],n,AGI(agi://127.0.0.1:4577/call_log)
> exten => _321[0123],n,Dial(SIP/${EXTEN}),30,to)
> exten => _321[0123],n,Voicemail,u${EXTEN}
> exten => _321[0123],n,Hangup
> 
> comments?
>