Re: [asterisk-users] incoming call label
Le 16/02/2018 à 05:30, the...@sys-concept.com a écrit : On 02/15/2018 04:49 PM, Joshua Colp wrote: On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: Thanks again for the hint. Here is the output from asterisk. The call is coming on Audocodes gateway from: pstn- But asterisk display: Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 Why not loolking up "pstn-" in sip.conf? It found pstn- using 10.10.0.8:5060 - if the request always comes from the same IP address and port it has no other way built in to differentiate between the two except by matching based on username in the 'From' header. It didn't find "pstn- using 10.10.0.8:5060" The call came IN from PSTN line on audiocodes equipment to FXO port that is labelled "pstn-" so asterisk reported as such. And I think asterisk suppose to lookup this label in sip.conf to the registered entry but instead selected pstn-9998 entry; I don't know why. If the call came IN on pstn- and sip.conf has two entries: [pstn-] [pstn-9998] Why it can not distinguish between the two of them correctly? -- Thelma If your device supports SIP authentication, you can try to turn on the "match_auth_username" parameter in sip.conf. It is said to be experimental but has always worked well for me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On 02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn- >> >> But asterisk display: >> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >> >> Why not loolking up "pstn-" in sip.conf? > > It found pstn- using 10.10.0.8:5060 - if the request always comes from > the same IP address and port it has no other way built in to differentiate > between the two except by matching based on username in the 'From' header. It didn't find "pstn- using 10.10.0.8:5060" The call came IN from PSTN line on audiocodes equipment to FXO port that is labelled "pstn-" so asterisk reported as such. And I think asterisk suppose to lookup this label in sip.conf to the registered entry but instead selected pstn-9998 entry; I don't know why. If the call came IN on pstn- and sip.conf has two entries: [pstn-] [pstn-9998] Why it can not distinguish between the two of them correctly? -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
Thelma On 02/15/2018 07:16 PM, the...@sys-concept.com wrote: > > On 02/15/2018 04:49 PM, Joshua Colp wrote: >> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: >> >> >> >>> >>> Thanks again for the hint. >>> Here is the output from asterisk. >>> >>> The call is coming on Audocodes gateway from: pstn- >>> >>> But asterisk display: >>> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >>> >>> Why not loolking up "pstn-" in sip.conf? >> >> It found pstn- using 10.10.0.8:5060 - if the request always comes from >> the same IP address and port it has no other way built in to differentiate >> between the two except by matching based on username in the 'From' header. >> > > Call comes from same IP address always. > To comes form Audiocode: > > <--- SIP read from UDP:10.10.0.8:5060 ---> > INVITE sip:4@10.10.0.4 SIP/2.0 > Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844 > Max-Forwards: 70 > From: "Z" ;tag=1c766802762 > To: > Call-ID: 7668022781522018162620@10.10.0.8 > CSeq: 1 INVITE > Contact: > > Contact: "sip:pstn-" > > And it found in sip.conf only: > Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 > > Is perhaps the name effected by the special character "-" (dash) that is > why it only matches "pstn" and take the first one it found. Will it > make a difference if I rename the port to pstn_ in configuration files. > > -- > Thelma sip show peers Name/username HostDyn Forcerport ComediaACL Port Status Description pstn-/voice- 10.10.0.8D No No 5060 Unmonitored pstn-9998/fax-999810.10.0.8D No No 5060 Unmonitored -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On 02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn- >> >> But asterisk display: >> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >> >> Why not loolking up "pstn-" in sip.conf? > > It found pstn- using 10.10.0.8:5060 - if the request always comes from > the same IP address and port it has no other way built in to differentiate > between the two except by matching based on username in the 'From' header. > Call comes from same IP address always. To comes form Audiocode: <--- SIP read from UDP:10.10.0.8:5060 ---> INVITE sip:4@10.10.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844 Max-Forwards: 70 From: "Z" ;tag=1c766802762 To: Call-ID: 7668022781522018162620@10.10.0.8 CSeq: 1 INVITE Contact: Contact: "sip:pstn-" And it found in sip.conf only: Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 Is perhaps the name effected by the special character "-" (dash) that is why it only matches "pstn" and take the first one it found. Will it make a difference if I rename the port to pstn_ in configuration files. -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > Thanks again for the hint. > Here is the output from asterisk. > > The call is coming on Audocodes gateway from: pstn- > > But asterisk display: > Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 > > Why not loolking up "pstn-" in sip.conf? It found pstn- using 10.10.0.8:5060 - if the request always comes from the same IP address and port it has no other way built in to differentiate between the two except by matching based on username in the 'From' header. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On 02/15/2018 04:08 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
>> On 02/15/2018 03:44 PM, Joshua Colp wrote:
>>> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
IN audocodes setting I have:
"EndPoint Phone Number"
Channel: 3phone number: pstn-
Channel: 4phone number: pstn-9998
When I am calling " pstn-" the port number "Channel:3" lights up but
asterisk is showing that the call is coming on "pstn-9998"
-- Executing . Answer("SIP/pstn-9998
Asterisk should be showing "pstn-" (not pstn-9998)
Where is this label coming from?
>>>
>>> It is from the SIP entry in sip.conf that it was matched against.
>>>
>>
>> Thanks for the input.
>>
>> In sip.conf I have relevant entries.
>>
>> [pstn-] ; incoming/outgoing calls on FXO port
>> type=friend
>> secret=spa354
>> username=voice-
>> mailbox=622 ; just for audiocodes error complain
>> host=dynamic
>> canreinvite=no ; (dtmf not wroking correctly without this one)
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> nat=no
>> context=incoming
>> callgroup=1
>> pickupgroup=1
>> insecure=invite
>>
>> [pstn-9998]
>> type=friend
>> secret=158567
>> username=fax-9998
>> insecure=invite
>> mailbox=622 ; just for audiocodes error complain
>> host=dynamic
>> canreinvite=no ; (dtmf not wroking correctly without this one)
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> nat=no
>> context=incoming
>> callgroup=1
>> pickupgroup=
>>
>> My asterisk registration is correct as well:
>> sip show users
>> Username Secret Accountcode Def.Context
>> ACL Forcerport
>> pstn-9998 158567 incoming
>> No No
>> pstn- spa354 incoming
>> No No
>>
>> Caller display ID from PSTN on FXO ports are working OK.
>> The [pstn-] is channel: 4
>> The [pstn-9998] is channel: 3
>>
>> If the call on Audocode is lighting UP "channel:3" the sip.conf should
>> associate that call with [pstn-] (and not [pstn-9998])
>
> Not necessarily. You appear to be doing IP+port based matching. If requests
> always come from the same source IP address and port, then it would match
> only one. Turning on sip debug using "sip set debug on" and verbosity using
> "core set debug 9" would give you more information about each packet
> (including where it is from) and what was actually matched based on it.
Thanks again for the hint.
Here is the output from asterisk.
The call is coming on Audocodes gateway from: pstn-
But asterisk display:
Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
Why not loolking up "pstn-" in sip.conf?
<--- SIP read from UDP:10.10.0.8:5060 --->
INVITE sip:4@10.10.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844
Max-Forwards: 70
From: "Z" ;tag=1c766802762
To:
Call-ID: 7668022781522018162620@10.10.0.8
CSeq: 1 INVITE
Contact:
Supported:
em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
v=0
o=AudiocodesGW 766797875 766797759 IN IP4 10.10.0.8
s=Phone-Call
c=IN IP4 10.10.0.8
t=0 0
m=audio 6000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<->
--- (14 headers 12 lines) ---
Sending to 10.10.0.8:5060 (no NAT)
Sending to 10.10.0.8:5060 (no NAT)
Using INVITE request as basis request - 7668022781522018162620@10.10.0.8
Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.0.8:6000
Looking for 4 in incoming (domain 10.10.0.4)
list_route: hop:
--
Thelma
--
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Re: [asterisk-users] incoming call label
On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
> On 02/15/2018 03:44 PM, Joshua Colp wrote:
> > On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
> >>
> >> IN audocodes setting I have:
> >> "EndPoint Phone Number"
> >>
> >> Channel: 3phone number: pstn-
> >> Channel: 4phone number: pstn-9998
> >>
> >> When I am calling " pstn-" the port number "Channel:3" lights up but
> >> asterisk is showing that the call is coming on "pstn-9998"
> >>
> >> -- Executing . Answer("SIP/pstn-9998
> >>
> >> Asterisk should be showing "pstn-" (not pstn-9998)
> >> Where is this label coming from?
> >
> > It is from the SIP entry in sip.conf that it was matched against.
> >
>
> Thanks for the input.
>
> In sip.conf I have relevant entries.
>
> [pstn-] ; incoming/outgoing calls on FXO port
> type=friend
> secret=spa354
> username=voice-
> mailbox=622 ; just for audiocodes error complain
> host=dynamic
> canreinvite=no ; (dtmf not wroking correctly without this one)
> disallow=all
> allow=ulaw
> allow=alaw
> nat=no
> context=incoming
> callgroup=1
> pickupgroup=1
> insecure=invite
>
> [pstn-9998]
> type=friend
> secret=158567
> username=fax-9998
> insecure=invite
> mailbox=622 ; just for audiocodes error complain
> host=dynamic
> canreinvite=no ; (dtmf not wroking correctly without this one)
> disallow=all
> allow=ulaw
> allow=alaw
> nat=no
> context=incoming
> callgroup=1
> pickupgroup=
>
> My asterisk registration is correct as well:
> sip show users
> Username Secret Accountcode Def.Context
> ACL Forcerport
> pstn-9998 158567 incoming
> No No
> pstn- spa354 incoming
> No No
>
> Caller display ID from PSTN on FXO ports are working OK.
> The [pstn-] is channel: 4
> The [pstn-9998] is channel: 3
>
> If the call on Audocode is lighting UP "channel:3" the sip.conf should
> associate that call with [pstn-] (and not [pstn-9998])
Not necessarily. You appear to be doing IP+port based matching. If requests
always come from the same source IP address and port, then it would match only
one. Turning on sip debug using "sip set debug on" and verbosity using "core
set debug 9" would give you more information about each packet (including where
it is from) and what was actually matched based on it.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
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asterisk-users mailing list
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Re: [asterisk-users] incoming call label
On 02/15/2018 03:44 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>>
>> IN audocodes setting I have:
>> "EndPoint Phone Number"
>>
>> Channel: 3phone number: pstn-
>> Channel: 4phone number: pstn-9998
>>
>> When I am calling " pstn-" the port number "Channel:3" lights up but
>> asterisk is showing that the call is coming on "pstn-9998"
>>
>> -- Executing . Answer("SIP/pstn-9998
>>
>> Asterisk should be showing "pstn-" (not pstn-9998)
>> Where is this label coming from?
>
> It is from the SIP entry in sip.conf that it was matched against.
>
Thanks for the input.
In sip.conf I have relevant entries.
[pstn-] ; incoming/outgoing calls on FXO port
type=friend
secret=spa354
username=voice-
mailbox=622 ; just for audiocodes error complain
host=dynamic
canreinvite=no ; (dtmf not wroking correctly without this one)
disallow=all
allow=ulaw
allow=alaw
nat=no
context=incoming
callgroup=1
pickupgroup=1
insecure=invite
[pstn-9998]
type=friend
secret=158567
username=fax-9998
insecure=invite
mailbox=622 ; just for audiocodes error complain
host=dynamic
canreinvite=no ; (dtmf not wroking correctly without this one)
disallow=all
allow=ulaw
allow=alaw
nat=no
context=incoming
callgroup=1
pickupgroup=
My asterisk registration is correct as well:
sip show users
Username Secret Accountcode Def.Context
ACL Forcerport
pstn-9998 158567 incoming
No No
pstn- spa354 incoming
No No
Caller display ID from PSTN on FXO ports are working OK.
The [pstn-] is channel: 4
The [pstn-9998] is channel: 3
If the call on Audocode is lighting UP "channel:3" the sip.conf should
associate that call with [pstn-] (and not [pstn-9998])
--
Thelma
--
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Re: [asterisk-users] incoming call label
On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>
> IN audocodes setting I have:
> "EndPoint Phone Number"
>
> Channel: 3phone number: pstn-
> Channel: 4phone number: pstn-9998
>
> When I am calling " pstn-" the port number "Channel:3" lights up but
> asterisk is showing that the call is coming on "pstn-9998"
>
> -- Executing . Answer("SIP/pstn-9998
>
> Asterisk should be showing "pstn-" (not pstn-9998)
> Where is this label coming from?
It is from the SIP entry in sip.conf that it was matched against.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
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asterisk-users mailing list
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[asterisk-users] incoming call label
I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
IN audocodes setting I have:
"EndPoint Phone Number"
Channel: 3phone number: pstn-
Channel: 4phone number: pstn-9998
When I am calling " pstn-" the port number "Channel:3" lights up but
asterisk is showing that the call is coming on "pstn-9998"
-- Executing . Answer("SIP/pstn-9998
Asterisk should be showing "pstn-" (not pstn-9998)
Where is this label coming from?
--
Thelma
--
_
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Re: [asterisk-users] [OT] Gigaset N510IP provisionning
Hi Olivier, Yes, it's possible, we have a provisioning like that on production. I recommend you to follow the Wiki of Gigaset and contact their support in case of misunderstanding: They help us a lot for the integration. You only have one XML file to provide, with an example in the wiki, if I remember correctly. Regards. -- Ludovic Gasc (GMLudo) 2018-02-12 16:03 GMT+01:00 Olivier : > Hello, > > Has someone met success in Gigaset N510IP DECT base station provisionning ? > If positive, could you describe a bit which files you had to create on > (HTTP) provsionning server ? > > > Best regards > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with DAHDI
Luca Bertoncello schrieb: > But if I try to call another VoIP-phone it rings but no voice will be > transferred... Got it! A "little" firewall problem... :( Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with DAHDI
Tzafrir Cohen schrieb:
> This means that you have configured a dahdi channel in
> /etc/asterisk/chan_dahdi.conf . The default configuration does not
> include one. Do you have any DAHDI device on the system?
I think not...
> If /dev/dahdi/channel itself does not exist, it means that the
> kernel-level support is not loaded (or not even configured). It is
> generally from the kernel module dahdi, which is an out-of-tree one.
>
> If you really want it, you may need to run:
>
> m-a a-i dahdi
>
> But do you really have a DAHDI device?
No, I haven't...
I remove the configuration
dahdichan = 1
for every user and I don't have any error anymore.
But if I try to call another VoIP-phone it rings but no voice will be
transferred...
Any idea?
I tryed to enable the debugging and I see that:
> 0xaa90a380 -- Strict RTP learning after remote address set to:
192.168.200.10:41000
-- Executing [00493517654321@default:1] Dial("SIP/00493511234567-",
"SIP/00493517654321") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/00493517654321
-- SIP/00493517654321-0001 is ringing
> 0xb4606f38 -- Strict RTP learning after remote address set to:
192.168.200.11:41000
-- SIP/00493517654321-0001 answered SIP/00493511234567-
-- Channel SIP/00493517654321-0001 joined 'simple_bridge' basic-bridge
<0ef9a447-b1b3-45af-a4af-7c4ac4d10546>
-- Channel SIP/00493511234567- joined 'simple_bridge' basic-bridge
<0ef9a447-b1b3-45af-a4af-7c4ac4d10546>
-- Channel SIP/00493517654321-0001 left 'simple_bridge' basic-bridge
<0ef9a447-b1b3-45af-a4af-7c4ac4d10546>
-- Channel SIP/00493511234567- left 'simple_bridge' basic-bridge
<0ef9a447-b1b3-45af-a4af-7c4ac4d10546>
== Spawn extension (default, 00493517654321, 1) exited non-zero on
'SIP/00493511234567-'
Where is the error?!?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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Re: [asterisk-users] Problem with DAHDI
On Thu, Feb 15, 2018 at 06:55:16PM +0100, Luca Bertoncello wrote: > Hi again! > > I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI > with Armbian/Debian 9. > > First test was to call a test service that say the time. Works! > Second test was to record my voice and play it again. Works! > Third test was to call the other VoIP-phone. It does NOT work... :( > > Then I noticed that, by starting, Asterisk says the following messages: > > [Feb 15 18:42:54] NOTICE[3428]: loader.c:1446 load_modules: 319 modules will > be loaded. > [Feb 15 18:43:01] NOTICE[3428]: res_odbc.c:1089 load_module: res_odbc loaded. > [Feb 15 18:43:12] WARNING[3428]: res_phoneprov.c:1231 get_defaults: Unable to > find a valid server address or name. > [Feb 15 18:43:15] NOTICE[3428]: pbx_lua.c:1640 load_or_reload_lua_stuff: Lua > PBX Switch loaded. > SIP channel loading... > [Feb 15 18:43:24] NOTICE[3428]: chan_skinny.c:8429 config_load: Configuring > skinny from skinny.conf > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open > '/dev/dahdi/channel': No such file or directory > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open > channel 1: No such file or directory > here = 0, tmp->channel = 1, channel = 1 This means that you have configured a dahdi channel in /etc/asterisk/chan_dahdi.conf . The default configuration does not include one. Do you have any DAHDI device on the system? > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to > register channel '1' > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan > '1' failure ignored: ignore_failed_channels. > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open > '/dev/dahdi/channel': No such file or directory > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open > channel 1: No such file or directory > here = 0, tmp->channel = 1, channel = 1 > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to > register channel '1' > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan > '1' failure ignored: ignore_failed_channels. > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open > '/dev/dahdi/channel': No such file or directory > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open > channel 1: No such file or directory > here = 0, tmp->channel = 1, channel = 1 > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to > register channel '1' > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan > '1' failure ignored: ignore_failed_channels. > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open > '/dev/dahdi/channel': No such file or directory > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open > channel 1: No such file or directory > here = 0, tmp->channel = 1, channel = 1 > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to > register channel '1' > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan > '1' failure ignored: ignore_failed_channels. > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open > '/dev/dahdi/channel': No such file or directory > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open > channel 1: No such file or directory > here = 0, tmp->channel = 1, channel = 1 > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to > register channel '1' > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan > '1' failure ignored: ignore_failed_channels. > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open > '/dev/dahdi/channel': No such file or directory > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open > channel 1: No such file or directory > here = 0, tmp->channel = 1, channel = 1 > [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to > register channel '1' > [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan > '1' failure ignored: ignore_failed_channels. > [Feb 15 18:43:28] NOTICE[3428]: confbridge/conf_config_parser.c:2094 > verify_default_profiles: Adding default_menu menu to app_confbridge > [Feb 15 18:43:28] NOTICE[3428]: cel_tds.c:452 tds_load_module: cel_tds has no > global category, nothing to configure. > [Feb 15 18:43:28] WARNING[3428]: cel_tds.c:557 load_module: cel_tds module > had config problems; declining load > [Feb 15 18:43:28] NOTICE[3428]: cel_custom.c:97 load_config: No mappings > found in cel_custom.conf. Not logging CEL to custom CSVs. > [Feb 15 18:43:29] ERROR[3428]: codec_dahdi.c:820 find_transcoders: Failed to > open /dev/dahdi/transcode: No such file or directory > Asterisk Ready. > > it does not seems to be normal, but I can't understa
[asterisk-users] Problem with DAHDI
Hi again! I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI with Armbian/Debian 9. First test was to call a test service that say the time. Works! Second test was to record my voice and play it again. Works! Third test was to call the other VoIP-phone. It does NOT work... :( Then I noticed that, by starting, Asterisk says the following messages: [Feb 15 18:42:54] NOTICE[3428]: loader.c:1446 load_modules: 319 modules will be loaded. [Feb 15 18:43:01] NOTICE[3428]: res_odbc.c:1089 load_module: res_odbc loaded. [Feb 15 18:43:12] WARNING[3428]: res_phoneprov.c:1231 get_defaults: Unable to find a valid server address or name. [Feb 15 18:43:15] NOTICE[3428]: pbx_lua.c:1640 load_or_reload_lua_stuff: Lua PBX Switch loaded. SIP channel loading... [Feb 15 18:43:24] NOTICE[3428]: chan_skinny.c:8429 config_load: Configuring skinny from skinny.conf [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:28] NOTICE[3428]: confbridge/conf_config_parser.c:2094 verify_default_profiles: Adding default_menu menu to app_confbridge [Feb 15 18:43:28] NOTICE[3428]: cel_tds.c:452 tds_load_module: cel_tds has no global category, nothing to configure. [Feb 15 18:43:28] WARNING[3428]: cel_tds.c:557 load_module: cel_tds module had config problems; declining load [Feb 15 18:43:28] NOTICE[3428]: cel_custom.c:97 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. [Feb 15 18:43:29] ERROR[3428]: codec_dahdi.c:820 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory Asterisk Ready. it does not seems to be normal, but I can't understand why /dev/dahdi/channel does not exists... I installed the Paket asterisk-dahdi, of course... Other question: what does the error about res_phoneprov.c means? Can someone help me? Thank you very much! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk comm
Re: [asterisk-users] chan_oss.c: Unable to register channel type 'OSS'
Zitat von Tzafrir Cohen :
Yes. It is useful if you want to call using a local sound device.
On a Banana PI? ;)
Consider editing /etc/asterisk/modules.conf and disable ('noload =>')
chan_oss.so .
So I did...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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Re: [asterisk-users] chan_oss.c: Unable to register channel type 'OSS'
On Thu, Feb 15, 2018 at 08:38:03AM +, Luca Bertoncello wrote:
> Zitat von Tzafrir Cohen :
>
> Hi,
>
> > Off-topic: any reason you don't use chan_alsa?
>
> This was the "Armbian installation", I didn't configured it extra...
>
> > Are you sure you quote the error message right?
>
> Copy+Paste... ;)
>
> But I searched a little bit and I really don't think, I need this module...
> As I undestand, I just need it, if I want to call/answer call using the
> console, and I really don't need this...
>
> Or I understood wrong?
Yes. It is useful if you want to call using a local sound device.
Consider editing /etc/asterisk/modules.conf and disable ('noload =>')
chan_oss.so .
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Tzafrir Cohen
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com
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Re: [asterisk-users] chan_oss.c: Unable to register channel type 'OSS'
Zitat von Tzafrir Cohen : Hi, Off-topic: any reason you don't use chan_alsa? This was the "Armbian installation", I didn't configured it extra... Are you sure you quote the error message right? Copy+Paste... ;) But I searched a little bit and I really don't think, I need this module... As I undestand, I just need it, if I want to call/answer call using the console, and I really don't need this... Or I understood wrong? Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_oss.c: Unable to register channel type 'OSS'
Hi, On Thu, Feb 15, 2018 at 07:45:00AM +, Luca Bertoncello wrote: > Hi list! > > Currently I use Asterisk 1.8.30.0 on an OpenWRT-Switch. > Now I want to change to Asterisk 13.14.1 on a Banana PI (with Armbian/Debian > 9). > Well, I copied the configuration and changed what needed, so basically, it > works, at least with my tests. Off-topic: any reason you don't use chan_alsa? > > But when Asterisk will be started, in the message log I get this error: > > [Feb 15 08:40:15] ERROR[3971] chan_oss.c: Unable to register channel type > 'OSS' > > Unfortunately I cannot find WHY Asterisk was unable to register a channel > type "OSS". > And then: do I need this? On the old Asterisk I didn't had that... Huh? chan_oss registers the channel type 'Console' . I just tried it on a rpi with a somewhat similar deb, and chan_oss had no problem loading and registering channel type Console. I'm not aware of a patch to do so in Debian asterisk package and I don't suppose Armbian maintain their own Asterisk package. Are you sure you quote the error message right? Maybe the error message is: Unable to register channel type 'Console' which may be because either chan_alsa or chan_console was already loaded? Of those three: * I suppose chan_alsa should work better than chan_oss * chan_console has nice support of multiple devices, but is completely broken in your specific version[1] [1] https://issues.asterisk.org/jira/browse/ASTERISK-27426 , and thanks Sean Bright -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
