[asterisk-users] Audiocodes MP-114 2FXO/2FXS help registering with Asterisk
I have AudioCodes MP-2FXO/2FXS but have a problem registering it with Asterisk. Any links or pointers to configuration how it is done? -- Joseph ___ -- 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
Re: [asterisk-users] TDM 400 hardware(?) issue
On Tue, 2009-12-22 at 11:53 -0700, Greg Woods wrote: > > the machine will lock up because the TDM board or the Dahdi > > driver goes south. /var/log/messages starts filling up with repeated > > messages: > > > > kernel: TDM PCI Master abort > > it's entirely possible that a flaky video card > is the whole problem. So I replaced it. Unfortunately this has not fixed the problem, although it does seem to occur much less frequently now. Back to the drawing board. --Greg ___ -- 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
Re: [asterisk-users] compile issues.
Aditya Kumar wrote: > but when I do make install: > I still get the same error... > > tar: vm-undeleted.gsm: Cannot change ownership to uid 1000, gid 1000: > Operation not permitted > tar: vm-unknown-caller.gsm: Cannot change ownership to uid 1000, gid > 1000: Operation not permitted > tar: vm-whichbox.gsm: Cannot change ownership to uid 1000, gid 1000: > Operation not permitted > tar: vm-youhave.gsm: Cannot change ownership to uid 1000, gid 1000: > Operation not permitted > tar: Error exit delayed from previous errors > make[1]: *** > [/home/aditya/asterisk/var/lib/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.16] > Error 2 > make[1]: Leaving directory `/home/aditya/asterisk-1.6.2.0/sounds' > make: *** [datafiles] Error 2 This is being caused by a packaging error in our asterisk-core-sounds tar files; they contain non-zero uid/gid values and they should not. If you edit sounds/Makefile in the Asterisk source tree and change the CORE_SOUNDS version to 1.4.17, then try the installation again, this should be fixed. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- 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
Re: [asterisk-users] pattern matching
You do not need to use pattern matching if you know the extension you are going to receive. Check the spelling on the dialplan if it does not work. You can start at the duplicated comma of the 34102. --Mensaje original-- De: Thomas Perron Remitente: asterisk-users-boun...@lists.digium.com Para: asterisk-users@lists.digium.com Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] pattern matching Enviado: 26 Dic, 2009 09:36 I want to ensure that only this extension is executed. But, I have others that are similar. I want: exten => 34101,1,Answer() exten => 34101,n,Record(34101:gsm) ; 34101 test zip code exten => 34101,n,Playback(34101) exten => 34101,n,Hangup Is this correct or do I need to have each of the four statements lead with an underscore (_) to make an exact match? Other code looks similar so I don't want the 102 to connect when I am dialing 101 exten => 34102,1,Answer() exten => 34102,,n,Record(34102:gsm) ; 34102 test zip code exten => 34102,n,Playback(34102) exten => 34102,n,Hangup ___ -- 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 Saludos, Juan E. Rodríguez ___ -- 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
Re: [asterisk-users] pattern matching
Thomas Perron wrote: > exten => 34101,1,Answer() > > Is this correct or do I need to have each of the four statements lead > with an underscore (_) to make an exact match? > Without the underscore, an exact match is required. The underscore, denotes a pattern. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- 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
[asterisk-users] pattern matching
I want to ensure that only this extension is executed. But, I have others that are similar. I want: exten => 34101,1,Answer() exten => 34101,n,Record(34101:gsm) ; 34101 test zip code exten => 34101,n,Playback(34101) exten => 34101,n,Hangup Is this correct or do I need to have each of the four statements lead with an underscore (_) to make an exact match? Other code looks similar so I don't want the 102 to connect when I am dialing 101 exten => 34102,1,Answer() exten => 34102,,n,Record(34102:gsm) ; 34102 test zip code exten => 34102,n,Playback(34102) exten => 34102,n,Hangup ___ -- 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
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
> >> 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
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
>
>
>
>
>
> ___
> -- 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
---
* 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] Tel uri Support
24 dec 2009 kl. 10.30 skrev Shelvananda, Ramananda Arkalgud: > Hi All, > > Is someone implemented Tel uri support in the latest asterisk ? If yes, can > you guys share some info on it > No. But I am very interested in why you ask? Do you have devices that support Tel: uri's? DO you have an idea on why Asterisk should support it? Regards, /O ___ -- 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
Re: [asterisk-users] 1.6 Troubleshooting help
24 dec 2009 kl. 08.18 skrev listu...@spamomania.co.uk: > Hi, > > How would I go about troubleshooting this: > > [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum > retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for > seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. Did you actually read the message? "See doc/sip-retransmit.txt." /O ___ -- 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
Re: [asterisk-users] How to exchange/get $variables from/to each channel on cmd Dial
23 dec 2009 kl. 16.00 skrev didier.cuffaut: > I apologize for my poor English. > So, i don't really understand 'how to' realize thus > > When you use the cmd Dial and want to get $ from caller channel to callee (or > callee channel from caller), which way is the right way ? > If you prefix a variable with an underscore, it will be copied to the outbound channel without the underscore. If you prefix with two underscores, it will be copied to the outbound channel with two underscores, thus will be inherited once again if that channel opens another (which happens if you're using chan_local). Regards, /olle --- o...@edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP >From PBX to large scale implementations for carriers. Contact us today! ___ -- 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
Re: [asterisk-users] Core show function?
23 dec 2009 kl. 19.52 skrev Ira: > Someone posted a message suggesting someone try sendtext() and so I > thought I'd see if it was useful. Much searching through help at the > CLI has failed to find any help for sendtext, but I did find that: > > "core show function vmcount" fails but: > > "core show function VMCOUNT" works. > > Is that a bug and if so, has it been reported? All functions are uppercase only, even though I personally think the CLI could be more helpful. /O ___ -- 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
