Re: [asterisk-users] Unable to create channel DAHDI

2016-06-09 Thread Matt Fredrickson 
Looks like the hookstate is listed as offhook.  I don't think
chan_dahdi will attempt to make a call out a device that is offhook.

Hope that helps,
Matthew Fredrickson

On Tue, Jun 7, 2016 at 1:36 PM, Brent Davidson
 wrote:
> In trying to troubleshoot the Delay after Answer problem I had before (which
> seems to be fixed), I have somehow created a new problem:
>
> Outgoing calls are now failing with the following message:
>
> [Jun  7 13:28:09] WARNING[9247][C-]: app_dial.c:2429 dial_exec_full:
> Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
>
> But I DO have working dahdi as incoming calls are working correctly.
>
> CLI> dahdi show channels
>Chan Extension   Context Language   MOH Interpret
> BlockedIn Service Description
>  pseudo defaultdefault
> Yes
>   3 mainmenu   default
> Yes
>   4 mainmenu   default
> Yes
> CLI> dahdi show status
> Description  Alarms  IRQbpviol CRCFra
> Codi Options  LBO
> Wildcard AEX410  OK  0  0  0  CAS
> Unk   0 db (CSU)/0-133 feet (DSX-1)
> CLI> dahdi show channel 3
> Channel: 3
> Description:
> File Descriptor: 14
> Span: 1
> Extension:
> Dialing: no
> Context: mainmenu
> Caller ID:
> Calling TON: 0
> Caller ID name:
> Mailbox: none
> Destroy: 0
> InAlarm: 0
> Signalling Type: FXS Kewlstart
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Busy Detection: yes
> Busy Count: 8
> Busy Pattern: 0,0,0,0
> TDD: no
> Relax DTMF: yes
> Dialing/CallwaitCAS: 0/0
> Default law: ulaw
> Fax Handled: no
> Pulse phone: no
> HW Gains (RX/TX): Disabled/Disabled
> SW Gains (RX/TX): 0.00/0.00
> Dynamic Range Compression (RX/TX): 0.00/0.00
> DND: no
> Echo Cancellation:
> 128 taps
> (unless TDM bridged) currently OFF
> Wait for dialtone: 0ms
> Actual Confinfo: Num/0, Mode/0x
> Actual Confmute: No
> Hookstate (FXS only): Offhook
> CLI>
>
> dahdi show channel 4
> Channel: 4
> Description:
> File Descriptor: 15
> Span: 1
> Extension:
> Dialing: no
> Context: mainmenu
> Caller ID:
> Calling TON: 0
> Caller ID name:
> Mailbox: none
> Destroy: 0
> InAlarm: 0
> Signalling Type: FXS Kewlstart
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Busy Detection: yes
> Busy Count: 8
> Busy Pattern: 0,0,0,0
> TDD: no
> Relax DTMF: yes
> Dialing/CallwaitCAS: 0/0
> Default law: ulaw
> Fax Handled: no
> Pulse phone: no
> HW Gains (RX/TX): Disabled/Disabled
> SW Gains (RX/TX): 0.00/0.00
> Dynamic Range Compression (RX/TX): 0.00/0.00
> DND: no
> Echo Cancellation:
> 128 taps
> (unless TDM bridged) currently OFF
> Wait for dialtone: 0ms
> Actual Confinfo: Num/0, Mode/0x
> Actual Confmute: No
> Hookstate (FXS only): Offhook
>
> The Hookstates always say offhook for some reason, though I'm not sure why.
>
> My setup:
>
> Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
> Server is CentOS 7
> Quad core CPU with 16GB Ram
> 2 Snom 300 phones.
> NO NAT.  Server and phone are on the same subnet with only a gigabit switch
> between them.
> Digium AEX410P analog card with 2 incoming analog PSTN lines
>
> Any ideas?
>
>
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] PJSIP: P-Asserted-Identity and Privacy headers are missing when CALLERID(num)=prohib

2016-06-09 Thread Richard Mudgett 
On Thu, Jun 9, 2016 at 11:40 AM, Olivier  wrote:

> Hello,
>
> My ITSP provides me with a SIP trunk which requires a CallerID value for
> any outbound call.
> Though a CallerID is required, anonymous calls are allowed.
> See extracts from a successfull anonymous call:
>
> From: "Anonymous" ;tag=438b284694b5b3de
>
> Privacy: id
>
> P-Asserted-Identity: "FooBar" 
>
> I'm trying to mimic this on a 13.8.0-enabled system.
>
> Whenever I set CALLERID(num-pres)=prohib in my dialplan, it seems
> P-Asserted-Identity not not present in outbound INVITE.
>
> I would expect to see it there along with a "Privacy: id" header.
>
> Do you agree with my expectation ?
> How can I work around this, keeping PJSIP stack ?
>

Do you have the following options enabled in pjsip.conf?
;trust_id_inbound=no; Accept identification information received from
this
; endpoint (default: "no")
;trust_id_outbound=no   ; Send private identification details to the
endpoint
; (default: "no")

Richard
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[asterisk-users] PJSIP: P-Asserted-Identity and Privacy headers are missing when CALLERID(num)=prohib

2016-06-09 Thread Olivier 
Hello,

My ITSP provides me with a SIP trunk which requires a CallerID value for
any outbound call.
Though a CallerID is required, anonymous calls are allowed.
See extracts from a successfull anonymous call:

From: "Anonymous" ;tag=438b284694b5b3de

Privacy: id

P-Asserted-Identity: "FooBar" >

I'm trying to mimic this on a 13.8.0-enabled system.

Whenever I set CALLERID(num-pres)=prohib in my dialplan, it seems
P-Asserted-Identity not not present in outbound INVITE.

I would expect to see it there along with a "Privacy: id" header.

Do you agree with my expectation ?
How can I work around this, keeping PJSIP stack ?

Best regards
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[asterisk-users] Fedora GLIBC 2.22 warning

2016-06-09 Thread George Joseph 
A recent update to the glibc-headers package (2.22-17) changed the order of
members in the sockaddr_storage structure which will cause an Asterisk
compile failure.  We shouldn't have been relying on the order and therefore
patches are up on gerrit for  the 11, 13 and master branches.

master: https://gerrit.asterisk.org/#/c/2980/2
13: https://gerrit.asterisk.org/#/c/2979/2
11: https://gerrit.asterisk.org/#/c/2981/2

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Digium, Inc. | 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] asterisk 13.9 with PJSIP -rejects with 488 Not Acceptable Here on invite with SRTP

2016-06-09 Thread Yaron Nachum 
Hi Everyone,
I am trying to setup an Audio Call from firefox WebRTC to Asterisk. The
Flow is:
PC -> SIPoWS -> KAMAILIO -> SIPoUDP -> ASTERISK
Regular call (no srtp)  works fine. However when I setup SRTP the asterisk
replies with 488 Not Acceptable Here.
I followed the Secure Calling Tutorial, but nothing seems to solve the
issue.

Below you can see the endpoint configuration + debug output.
Any help would be appreciated.

[acme]
type=endpoint
transport=transport-udp
context=app-router
disallow=all
allow=alaw
allow=ulaw
aors=acme
direct_media=no
media_encryption=dtls
dtls_verify=no
dtls_cert_file = /etc/asterisk/keys/asterisk.pem
dtls_private_key = /etc/asterisk/keys/asterisk.pem
dtls_setup=actpass
use_avpf=yes
ice_support=yes
media_use_received_transport=yes



<--- Received SIP request (2019 bytes) from UDP:10.25.133.209:5064 --->
INVITE sip:001...@ipcentrex.bezeq.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.25.133.209:5064;branch=z9hG4bK258keq2010n19mk406d1.1
Via: SIP/2.0/TCP 147.235.160.240:443
;branch=z9hG4bKd954.785f69fdfe198df7cabc66c0132a6b4c.0
Max-Forwards: 68
To: 
From: "1000" ;tag=1800vui39b
Call-ID: abars16ecm7s7icq4asq
CSeq: 3488 INVITE
Contact: 
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY
Content-Type: application/sdp
Supported: outbound
User-Agent: SIP.js/0.7.3
Content-Length: 1366

v=0
o=mozilla...THIS_IS_SDPARTA-46.0.1 4701336699161149943 0 IN IP4
10.25.133.241
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256
A8:CD:3D:44:3E:98:38:4F:3C:92:B7:05:B0:2B:91:0F:0F:39:7A:49:1F:8B:FB:26:18:1B:26:16:6B:2A:9C:03
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 20582 UDP/TLS/RTP/SAVPF 109 9 0 8
c=IN IP4 10.25.133.241
a=candidate:0 1 UDP 2122187007 147.235.159.2 58553 typ host
a=candidate:2 1 UDP 2122121471 2002:93eb:9f02::93eb:9f02 58554 typ host
a=candidate:4 1 UDP 2122252543 10.2.0.15 58555 typ host
a=candidate:0 2 UDP 2122187006 147.235.159.2 58556 typ host
a=candidate:2 2 UDP 2122121470 2002:93eb:9f02::93eb:9f02 58557 typ host
a=candidate:4 2 UDP 2122252542 10.2.0.15 58558 typ host
a=candidate:5 1 UDP 1686052863 62.219.92.9 58555 typ srflx raddr 10.2.0.15
rport 58555
a=candidate:5 2 UDP 1686052862 62.219.92.9 58558 typ srflx raddr 10.2.0.15
rport 58558
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:109 maxplaybackrate=48000;stereo=1
a=ice-pwd:446b46366a44a35b757dbec6a85a06a7
a=ice-ufrag:8d8969e7
a=mid:sdparta_0
a=msid:{60af682c-6e6d-4ad9-a165-02ce89d3ca8a}
{37d6a8a7-78c0-42b4-b52e-cdec5b076b1f}
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:2150808732 cname:{4c9490b5-d183-4f6f-954a-56b87f361da2}

[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :sip_endpoint.c
Distributing rdata to modules: Request msg INVITE/cseq=3488
(rdata0x7f0908008b28)
[Jun  9 17:04:18] DEBUG[13281]: netsock2.c:172 ast_sockaddr_split_hostport:
Splitting '10.25.133.209' into...
[Jun  9 17:04:18] DEBUG[13281]: netsock2.c:226 ast_sockaddr_split_hostport:
...host '10.25.133.209' and port ''.
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:113
ip_identify_match_check: Source address 10.25.133.209:5064 does not match
identify 'KAMnet_TST'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:113
ip_identify_match_check: Source address 10.25.133.209:5064 does not match
identify 'KAMnet_CBS'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:113
ip_identify_match_check: Source address 10.25.133.209:5064 does not match
identify 'clacli_j5'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:113
ip_identify_match_check: Source address 10.25.133.209:5064 does not match
identify 'clacli_j6'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:113
ip_identify_match_check: Source address 10.25.133.209:5064 does not match
identify 'clacli_j7'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:108
ip_identify_match_check: Source address 10.25.133.209:5064 matches identify
'acme'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:143
ip_identify: Retrieved endpoint acme
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :tsx0x7f090001f
..Transaction created for Request msg INVITE/cseq=3488 (rdata0x7f0908008b28)
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :tsx0x7f090001f
.Incoming Request msg INVITE/cseq=3488 (rdata0x7f0908008b28) in state Null
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :tsx0x7f090001f
..State changed from Null to Trying, event=RX_MSG
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :dlg0x7f0900010
...Transaction tsx0x7f090001f8e8 state changed to Trying
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :dlg0x7f0900010 .UAS
dialog created
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :dlg0x7f0900010
.Module mod-invite added

[asterisk-users] asterisk pam authentication support

2016-06-09 Thread Willy Offermans 
Dear asterisk friends,

Can someone tell me whether asterisk supports PAM authentication or not?

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Mit freundlichen Gruessen,

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Re: [asterisk-users] Want to detect sound

2016-06-09 Thread Mamadou NGOM 

Hello,When  i use mixmonitor() I manage not to detect still the silence. what a local channel. My asterisk is connected to an operator among whom the channel it is for example  "SIP/from provider 0048".In fact, I do not want to record the silence.I want that when caller  do not speak, I can ask for more  to record  again his message.exten => 0XX,n(erreur),Record(${filename:wav,5,5)After this line i want  to test a condition: if  the caller say nothing, asterisk go to back in the line recording again (erreur).Can i use this following line ?exten => 0X,n,GotoIf($["${STAT(e,${RECORDED_FILE}"="0"]?erreur).If simeone can help me to resolve this problem.Regards!!Le 7 juin 2016 à 12:59, Faheem Muhammad  a écrit :Try MixMonitor. Land the call to a local channel and answer it.This code will record the silence as well.exten => _X.,1,MixMonitor()exten => _X.,n,Dial(Local/100@context1)[context1]exten => _X.,1,Answer()exten => _X.,n,Dial(SIP/${EXTEN}On Tue, Jun 7, 2016 at 2:16 PM, Mamadou NGOM  wrote:Hello everybody,I manage not to detect one silence with record () when I make as follows:Exten = > 0178900271, n, Record ($ ${ link_recorded_pseudos_clients } pseudo_ Client_Id} wav, 5,5) exten = > 0178900271, n, GotoIf ($ [" $ {STAT (e, RECORDED_FILE} " = "0"]? Erreur_enregistrement_PPX17_1)When I say nothing, it do not return to the stage "erreur_enregistrement_PPX17_1"If you can help me?Mamadou NGOMIngénieur Télécommunications & RéseauxMobile: 06-47-02-67-86Skype: Mamadou NumericapNumeriCap – SAS au capital de 30.000,00€ - RCS de Toulon N° 530188432 – TVA FR 485301188432 – APE6110Z - ARCEP N°13/0015. siège social : « le Galaxie C » 526 avenue Maréchal de Lattre de Tassigny 83000 Toulon. mail: fina...@numericap.comCentre d’exploitation : « Résidence les Coquières » 11 avenue Joseph Fallen - 13400 Aubagne – Tel :04.42.73.88.52 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:                http://www.asterisk.org/hello  asterisk-users mailing list To UNSUBSCRIBE or update options visit:    http://lists.digium.com/mailman/listinfo/asterisk-usersMamadou NGOMIngénieur Télécommunications & RéseauxMobile: 06-47-02-67-86Skype: Mamadou NumericapNumeriCap – SAS au capital de 30.000,00€ - RCS de Toulon N° 530188432 – TVA FR 485301188432 – APE6110Z - ARCEP N°13/0015. siège social : « le Galaxie C » 526 avenue Maréchal de Lattre de Tassigny 83000 Toulon. mail: fina...@numericap.comCentre d’exploitation : « Résidence les Coquières » 11 avenue Joseph Fallen - 13400 Aubagne – Tel :04.42.73.88.52 

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