Re: [asterisk-users] how to check Asterisk SIP registration
I appreciate everyone's feedback. I did not post the "sip show peers" output because I did not have time to save it but I'm fairly sure that qualify was "OK" and that IP addresses did show up. NAT/firewall is not an issue because Asterisk and the sip devices are on the same network (open LAN). Anyway, regardless of the "sip show peers" output, the fact that the SIP devices registered fine and communication was re-established after killing asterisk and starting it, demonstrates that the root cause is not the "network" but the Asterisk's SIP service. I am using an alias IP address on the SIP server. Usually it works fine but maybe this time something went wrong. At the time I had my issue, I checked that the alias IP address was defined. Maybe Asterisk's SIP service was not correctly bound/listening to that alias IP address... Maybe removing and adding the alias IP address would have magically solved the issue but I did not try that. Can the SIP service be restarted without affecting the rest of Asterisk? (I don't think "sip reload" does this) Thanks, Vieri --- On Sat, 12/26/09, Olle E. Johansson wrote: > You've unfortunately gotten a lot of > confused answers. To try to clear this up: > > 1. Only type=peer objects accept registrations. "sip show > users" or "sip show registry" has nothing to do with peers. > A peer might be part of a type=friend > 2. If you see IP addresses when you run "sip show peers" > then those objects have an active registration, Asterisk > knows where to reach them. > 3. Nat's or firewalls between the device and Asterisk might > cause issues with Asterisk sending messages to them or > devices sending messages to Asterisk > 4. Your output below indicates that Asterisk doesn't know > how to reach the device, that Asterisk has no IP and port > address to send messages to, thus the device is not > registered at all. > 5. Turning "qualify" on can help with keeping a NAT binding > open. > > To summarize, start with looking for IP address in "sip > show peers". If we have an IP address, check the result of > the Qualify option in the same output. If there's an IP, the > device could reach Asterisk. If the status is "unreachable" > Asterisk could not reach the device on the IP address. > Then go hunting in your network to find the issue. > > Best regards, > /Olle > > > 24 dec 2009 kl. 17.39 skrev Vieri: > > > Unfortunately, "sip show peers" did not "work" in my > case. The sip peers were apparently "online" and "OK" (I use > qualify=yes) but they weren't... > > The SIP clients could NOT register, so they were > offline but "sip show peers" stated that they were OK. > > > > I would prefer to perform an "automated" SIP > registration (via cron script). If it fails then I can spawn > a "rescue" script. > > Surely, a "real" sip registration is more reliable > then "sip show peers". > > > > Any ideas? > > > > Vieri > > > > > > --- On Wed, 12/23/09, Danny Nicholas > wrote: > > > >> "Sip show users" or "sip show peers" > >> should do the trick, but I'm not sure > >> about 1.2; all of my experience is in the > 1.4 > >> branch. > >> > >> -Original Message- > >> From: asterisk-users-boun...@lists.digium.com > >> [mailto:asterisk-users-boun...@lists.digium.com] > >> On Behalf Of Vieri > >> Sent: Wednesday, December 23, 2009 1:09 PM > >> To: asterisk-users@lists.digium.com > >> Subject: [asterisk-users] how to check Asterisk > SIP > >> registration > >> > >> Hi, > >> > >> This is the first time I experience this problem > with > >> Asterisk: > >> all of a sudden SIP registrations stopped working. > Active > >> calls kept working > >> but new calls could not be established (I did NOT > perform a > >> "graceful > >> restart"). > >> > >> Besides, would a "restart gracefully" actually > deny SIP > >> registration? > >> > >> I did not have a network issue because killing > asterisk and > >> starting it > >> again solved the problem. > >> > >> How can I diagnose what happened to the SIP > service (I > >> checked the log but > >> am quite lost)? > >> > >> Also, how can I do a simple command-line "check" > to see >
Re: [asterisk-users] how to check Asterisk SIP registration
You've unfortunately gotten a lot of confused answers. To try to clear this up:
1. Only type=peer objects accept registrations. "sip show users" or "sip show
registry" has nothing to do with peers. A peer might be part of a type=friend
2. If you see IP addresses when you run "sip show peers" then those objects
have an active registration, Asterisk knows where to reach them.
3. Nat's or firewalls between the device and Asterisk might cause issues with
Asterisk sending messages to them or devices sending messages to Asterisk
4. Your output below indicates that Asterisk doesn't know how to reach the
device, that Asterisk has no IP and port address to send messages to, thus the
device is not registered at all.
5. Turning "qualify" on can help with keeping a NAT binding open.
To summarize, start with looking for IP address in "sip show peers". If we have
an IP address, check the result of the Qualify option in the same output. If
there's an IP, the device could reach Asterisk. If the status is "unreachable"
Asterisk could not reach the device on the IP address.
Then go hunting in your network to find the issue.
Best regards,
/Olle
24 dec 2009 kl. 17.39 skrev Vieri:
> Unfortunately, "sip show peers" did not "work" in my case. The sip peers were
> apparently "online" and "OK" (I use qualify=yes) but they weren't...
> The SIP clients could NOT register, so they were offline but "sip show peers"
> stated that they were OK.
>
> I would prefer to perform an "automated" SIP registration (via cron script).
> If it fails then I can spawn a "rescue" script.
> Surely, a "real" sip registration is more reliable then "sip show peers".
>
> Any ideas?
>
> Vieri
>
>
> --- On Wed, 12/23/09, Danny Nicholas wrote:
>
>> "Sip show users" or "sip show peers"
>> should do the trick, but I'm not sure
>> about 1.2; all of my experience is in the 1.4
>> branch.
>>
>> -----Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com]
>> On Behalf Of Vieri
>> Sent: Wednesday, December 23, 2009 1:09 PM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] how to check Asterisk SIP
>> registration
>>
>> Hi,
>>
>> This is the first time I experience this problem with
>> Asterisk:
>> all of a sudden SIP registrations stopped working. Active
>> calls kept working
>> but new calls could not be established (I did NOT perform a
>> "graceful
>> restart").
>>
>> Besides, would a "restart gracefully" actually deny SIP
>> registration?
>>
>> I did not have a network issue because killing asterisk and
>> starting it
>> again solved the problem.
>>
>> How can I diagnose what happened to the SIP service (I
>> checked the log but
>> am quite lost)?
>>
>> Also, how can I do a simple command-line "check" to see
>> that SIP
>> registrations are OK? I would like to use a SIP client
>> (like sipsak) to
>> perform a simple registration from a custom bash script so
>> I can quickly
>> detect if this problem occurs again and "auto-kill+restart"
>> the asterisk
>> process. I know this sounds ugly but on my production
>> server, it's better to
>> bring the whole system down and back up in as little time
>> as possible.
>>
>> Any suggestions?
>>
>> Asterisk is 1.2.31.1
>>
>> Some log lines:
>>
>> Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial
>> deadlock for
>> 'SIP/4053-b4520e98'
>> Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial
>> deadlock for
>> '0xb4302278', 9 retries!
>>
>> Dec 23 13:13:43 VERBOSE[18837] logger.c:
>>-- Executing
>> Dial("SIP/6174-b456d828", "SIP/4062|20|tTwWM(auto-blkvm)")
>> in new stack
>> Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create
>> channel of type
>> 'SIP' (cause 3 - No route to destination)
>> Dec 23 13:13:43 VERBOSE[18837]
>> logger.c: == Everyone is busy/congested at
>> this time (1:0/0/1)
>> Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with
>> DIALSTATUS=CHANUNAVAIL.
>>
>> Thanks,
>>
>> Vieri
>
>
>
>
>
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---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
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Re: [asterisk-users] how to check Asterisk SIP registration
I wrote a script to check clients and restart asterisk if registrations died
(external IAX)...but you could modify for your needs. Check it out on
www.generationd.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Thursday, December 24, 2009 12:06 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] how to check Asterisk SIP registration
Thanks but "sip show registry" yields nothing.
--- On Thu, 12/24/09, Danny Nicholas wrote:
> "sip show registry" might be more
> helpful.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Vieri
> Sent: Thursday, December 24, 2009 10:39 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] how to check Asterisk SIP registration
>
> Unfortunately, "sip show peers" did not "work" in my case.
> The sip peers
> were apparently "online" and "OK" (I use qualify=yes) but they
> weren't...
> The SIP clients could NOT register, so they were offline but "sip show
> peers" stated that they were OK.
>
> I would prefer to perform an "automated" SIP registration (via cron
> script).
> If it fails then I can spawn a "rescue" script.
> Surely, a "real" sip registration is more reliable then "sip show
> peers".
>
> Any ideas?
>
> Vieri
>
>
> --- On Wed, 12/23/09, Danny Nicholas
> wrote:
>
> > "Sip show users" or "sip show peers"
> > should do the trick, but I'm not sure about 1.2; all of my
> > experience is in the 1.4 branch.
> >
> > -----Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com]
> > On Behalf Of Vieri
> > Sent: Wednesday, December 23, 2009 1:09 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] how to check Asterisk SIP registration
> >
> > Hi,
> >
> > This is the first time I experience this problem with
> > Asterisk:
> > all of a sudden SIP registrations stopped working.
> Active
> > calls kept working
> > but new calls could not be established (I did NOT
> perform a
> > "graceful
> > restart").
> >
> > Besides, would a "restart gracefully" actually deny
> SIP
> > registration?
> >
> > I did not have a network issue because killing
> asterisk and
> > starting it
> > again solved the problem.
> >
> > How can I diagnose what happened to the SIP service
> (I
> > checked the log but
> > am quite lost)?
> >
> > Also, how can I do a simple command-line "check" to
> see
> > that SIP
> > registrations are OK? I would like to use a SIP
> client
> > (like sipsak) to
> > perform a simple registration from a custom bash
> script so
> > I can quickly
> > detect if this problem occurs again and
> "auto-kill+restart"
> > the asterisk
> > process. I know this sounds ugly but on my production server, it's
> > better to bring the whole system down and back up in as little
> time
> > as possible.
> >
> > Any suggestions?
> >
> > Asterisk is 1.2.31.1
> >
> > Some log lines:
> >
> > Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding
> initial
> > deadlock for
> > 'SIP/4053-b4520e98'
> > Dec 23 13:13:16 WARNING[11482] channel.c: Avoided
> initial
> > deadlock for
> > '0xb4302278', 9 retries!
> >
> > Dec 23 13:13:43 VERBOSE[18837] logger.c:
> > -- Executing
> > Dial("SIP/6174-b456d828",
> "SIP/4062|20|tTwWM(auto-blkvm)")
> > in new stack
> > Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to
> create
> > channel of type
> > 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43
> > VERBOSE[18837]
> > logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec
> > 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with
> > DIALSTATUS=CHANUNAVAIL.
> >
> > Thanks,
> >
> > Vieri
>
>
>
>
>
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Re: [asterisk-users] how to check Asterisk SIP registration
Thanks but "sip show registry" yields nothing.
--- On Thu, 12/24/09, Danny Nicholas wrote:
> "sip show registry" might be more
> helpful.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Vieri
> Sent: Thursday, December 24, 2009 10:39 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] how to check Asterisk SIP
> registration
>
> Unfortunately, "sip show peers" did not "work" in my case.
> The sip peers
> were apparently "online" and "OK" (I use qualify=yes) but
> they weren't...
> The SIP clients could NOT register, so they were offline
> but "sip show
> peers" stated that they were OK.
>
> I would prefer to perform an "automated" SIP registration
> (via cron script).
> If it fails then I can spawn a "rescue" script.
> Surely, a "real" sip registration is more reliable then
> "sip show peers".
>
> Any ideas?
>
> Vieri
>
>
> --- On Wed, 12/23/09, Danny Nicholas
> wrote:
>
> > "Sip show users" or "sip show peers"
> > should do the trick, but I'm not sure
> > about 1.2; all of my experience is in the 1.4
> > branch.
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com]
> > On Behalf Of Vieri
> > Sent: Wednesday, December 23, 2009 1:09 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] how to check Asterisk SIP
> > registration
> >
> > Hi,
> >
> > This is the first time I experience this problem with
> > Asterisk:
> > all of a sudden SIP registrations stopped working.
> Active
> > calls kept working
> > but new calls could not be established (I did NOT
> perform a
> > "graceful
> > restart").
> >
> > Besides, would a "restart gracefully" actually deny
> SIP
> > registration?
> >
> > I did not have a network issue because killing
> asterisk and
> > starting it
> > again solved the problem.
> >
> > How can I diagnose what happened to the SIP service
> (I
> > checked the log but
> > am quite lost)?
> >
> > Also, how can I do a simple command-line "check" to
> see
> > that SIP
> > registrations are OK? I would like to use a SIP
> client
> > (like sipsak) to
> > perform a simple registration from a custom bash
> script so
> > I can quickly
> > detect if this problem occurs again and
> "auto-kill+restart"
> > the asterisk
> > process. I know this sounds ugly but on my production
> > server, it's better to
> > bring the whole system down and back up in as little
> time
> > as possible.
> >
> > Any suggestions?
> >
> > Asterisk is 1.2.31.1
> >
> > Some log lines:
> >
> > Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding
> initial
> > deadlock for
> > 'SIP/4053-b4520e98'
> > Dec 23 13:13:16 WARNING[11482] channel.c: Avoided
> initial
> > deadlock for
> > '0xb4302278', 9 retries!
> >
> > Dec 23 13:13:43 VERBOSE[18837] logger.c:
> > -- Executing
> > Dial("SIP/6174-b456d828",
> "SIP/4062|20|tTwWM(auto-blkvm)")
> > in new stack
> > Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to
> create
> > channel of type
> > 'SIP' (cause 3 - No route to destination)
> > Dec 23 13:13:43 VERBOSE[18837]
> > logger.c: == Everyone is busy/congested at
> > this time (1:0/0/1)
> > Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with
> > DIALSTATUS=CHANUNAVAIL.
> >
> > Thanks,
> >
> > Vieri
>
>
>
>
>
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Re: [asterisk-users] how to check Asterisk SIP registration
"sip show registry" might be more helpful.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Thursday, December 24, 2009 10:39 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] how to check Asterisk SIP registration
Unfortunately, "sip show peers" did not "work" in my case. The sip peers
were apparently "online" and "OK" (I use qualify=yes) but they weren't...
The SIP clients could NOT register, so they were offline but "sip show
peers" stated that they were OK.
I would prefer to perform an "automated" SIP registration (via cron script).
If it fails then I can spawn a "rescue" script.
Surely, a "real" sip registration is more reliable then "sip show peers".
Any ideas?
Vieri
--- On Wed, 12/23/09, Danny Nicholas wrote:
> "Sip show users" or "sip show peers"
> should do the trick, but I'm not sure
> about 1.2; all of my experience is in the 1.4
> branch.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Vieri
> Sent: Wednesday, December 23, 2009 1:09 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] how to check Asterisk SIP
> registration
>
> Hi,
>
> This is the first time I experience this problem with
> Asterisk:
> all of a sudden SIP registrations stopped working. Active
> calls kept working
> but new calls could not be established (I did NOT perform a
> "graceful
> restart").
>
> Besides, would a "restart gracefully" actually deny SIP
> registration?
>
> I did not have a network issue because killing asterisk and
> starting it
> again solved the problem.
>
> How can I diagnose what happened to the SIP service (I
> checked the log but
> am quite lost)?
>
> Also, how can I do a simple command-line "check" to see
> that SIP
> registrations are OK? I would like to use a SIP client
> (like sipsak) to
> perform a simple registration from a custom bash script so
> I can quickly
> detect if this problem occurs again and "auto-kill+restart"
> the asterisk
> process. I know this sounds ugly but on my production
> server, it's better to
> bring the whole system down and back up in as little time
> as possible.
>
> Any suggestions?
>
> Asterisk is 1.2.31.1
>
> Some log lines:
>
> Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial
> deadlock for
> 'SIP/4053-b4520e98'
> Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial
> deadlock for
> '0xb4302278', 9 retries!
>
> Dec 23 13:13:43 VERBOSE[18837] logger.c:
> -- Executing
> Dial("SIP/6174-b456d828", "SIP/4062|20|tTwWM(auto-blkvm)")
> in new stack
> Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create
> channel of type
> 'SIP' (cause 3 - No route to destination)
> Dec 23 13:13:43 VERBOSE[18837]
> logger.c: == Everyone is busy/congested at
> this time (1:0/0/1)
> Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with
> DIALSTATUS=CHANUNAVAIL.
>
> Thanks,
>
> Vieri
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Re: [asterisk-users] how to check Asterisk SIP registration
Unfortunately, "sip show peers" did not "work" in my case. The sip peers were
apparently "online" and "OK" (I use qualify=yes) but they weren't...
The SIP clients could NOT register, so they were offline but "sip show peers"
stated that they were OK.
I would prefer to perform an "automated" SIP registration (via cron script). If
it fails then I can spawn a "rescue" script.
Surely, a "real" sip registration is more reliable then "sip show peers".
Any ideas?
Vieri
--- On Wed, 12/23/09, Danny Nicholas wrote:
> "Sip show users" or "sip show peers"
> should do the trick, but I'm not sure
> about 1.2; all of my experience is in the 1.4
> branch.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Vieri
> Sent: Wednesday, December 23, 2009 1:09 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] how to check Asterisk SIP
> registration
>
> Hi,
>
> This is the first time I experience this problem with
> Asterisk:
> all of a sudden SIP registrations stopped working. Active
> calls kept working
> but new calls could not be established (I did NOT perform a
> "graceful
> restart").
>
> Besides, would a "restart gracefully" actually deny SIP
> registration?
>
> I did not have a network issue because killing asterisk and
> starting it
> again solved the problem.
>
> How can I diagnose what happened to the SIP service (I
> checked the log but
> am quite lost)?
>
> Also, how can I do a simple command-line "check" to see
> that SIP
> registrations are OK? I would like to use a SIP client
> (like sipsak) to
> perform a simple registration from a custom bash script so
> I can quickly
> detect if this problem occurs again and "auto-kill+restart"
> the asterisk
> process. I know this sounds ugly but on my production
> server, it's better to
> bring the whole system down and back up in as little time
> as possible.
>
> Any suggestions?
>
> Asterisk is 1.2.31.1
>
> Some log lines:
>
> Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial
> deadlock for
> 'SIP/4053-b4520e98'
> Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial
> deadlock for
> '0xb4302278', 9 retries!
>
> Dec 23 13:13:43 VERBOSE[18837] logger.c:
> -- Executing
> Dial("SIP/6174-b456d828", "SIP/4062|20|tTwWM(auto-blkvm)")
> in new stack
> Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create
> channel of type
> 'SIP' (cause 3 - No route to destination)
> Dec 23 13:13:43 VERBOSE[18837]
> logger.c: == Everyone is busy/congested at
> this time (1:0/0/1)
> Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with
> DIALSTATUS=CHANUNAVAIL.
>
> Thanks,
>
> Vieri
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Re: [asterisk-users] how to check Asterisk SIP registration
"Sip show users" or "sip show peers" should do the trick, but I'm not sure
about 1.2; all of my experience is in the 1.4 branch.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Wednesday, December 23, 2009 1:09 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to check Asterisk SIP registration
Hi,
This is the first time I experience this problem with Asterisk:
all of a sudden SIP registrations stopped working. Active calls kept working
but new calls could not be established (I did NOT perform a "graceful
restart").
Besides, would a "restart gracefully" actually deny SIP registration?
I did not have a network issue because killing asterisk and starting it
again solved the problem.
How can I diagnose what happened to the SIP service (I checked the log but
am quite lost)?
Also, how can I do a simple command-line "check" to see that SIP
registrations are OK? I would like to use a SIP client (like sipsak) to
perform a simple registration from a custom bash script so I can quickly
detect if this problem occurs again and "auto-kill+restart" the asterisk
process. I know this sounds ugly but on my production server, it's better to
bring the whole system down and back up in as little time as possible.
Any suggestions?
Asterisk is 1.2.31.1
Some log lines:
Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for
'SIP/4053-b4520e98'
Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for
'0xb4302278', 9 retries!
Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing
Dial("SIP/6174-b456d828", "SIP/4062|20|tTwWM(auto-blkvm)") in new stack
Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type
'SIP' (cause 3 - No route to destination)
Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at
this time (1:0/0/1)
Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with
DIALSTATUS=CHANUNAVAIL.
Thanks,
Vieri
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[asterisk-users] how to check Asterisk SIP registration
Hi,
This is the first time I experience this problem with Asterisk:
all of a sudden SIP registrations stopped working. Active calls kept working
but new calls could not be established (I did NOT perform a "graceful
restart").
Besides, would a "restart gracefully" actually deny SIP registration?
I did not have a network issue because killing asterisk and starting it again
solved the problem.
How can I diagnose what happened to the SIP service (I checked the log but am
quite lost)?
Also, how can I do a simple command-line "check" to see that SIP registrations
are OK? I would like to use a SIP client (like sipsak) to perform a simple
registration from a custom bash script so I can quickly detect if this problem
occurs again and "auto-kill+restart" the asterisk process. I know this sounds
ugly but on my production server, it's better to bring the whole system down
and back up in as little time as possible.
Any suggestions?
Asterisk is 1.2.31.1
Some log lines:
Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for
'SIP/4053-b4520e98'
Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for
'0xb4302278', 9 retries!
Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing
Dial("SIP/6174-b456d828", "SIP/4062|20|tTwWM(auto-blkvm)") in new stack
Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type
'SIP' (cause 3 - No route to destination)
Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at
this time (1:0/0/1)
Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.
Thanks,
Vieri
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