Re: [asterisk-users] How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?
Hi Alex, Am Dienstag, den 25.02.2014, 13:04 -0500 schrieb Alex Villacís Lasso: > El 25/02/14 08:30, Karsten Wemheuer escribió: > > Hi Alex, > > > > Am Donnerstag, den 20.02.2014, 13:48 -0500 schrieb Alex Villacís Lasso: > >> I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following > >> the setup guide at > >> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb > >> . I want to run asterisk and kamailio on the same server, with SIP > >> realtime configuration > >> (MySQL database) so that kamailio authenticates and then forwards the > >> registration to asterisk on localhost. The setup calls for asterisk to > >> be configured to listen for SIP traffic on all interfaces, on a > >> nonstandard port (I chose 5080). It also calls for > >> blanking of the password for the SIP peer (in my case, a softphone), > >> so that it will not request for authentication again. I have managed > >> to make a call with working audio from the softphone to an extension > >> on asterisk through kamailio. > >> > >> My concern is that asterisk is left listening for SIP through all > >> interfaces and with no SIP passwords. I want to secure the setup > >> against directed traffic to the asterisk UDP port (5080), that > >> bypasses the kamailio process. I tried setting > >> bindaddr=127.0.0.1 so asterisk will only listen for SIP traffic on > >> localhost, but this has the side effect of also removing audio - the > >> call appears to be successful on the softphone and on the asterisk > >> logs, but no audio is actually heard. My theory is > >> that the RTP traffic is being sent to kamailio instead of the > >> softphone. > >> > >> How can I set up asterisk so that it can send RTP anywhere but reject > >> any SIP traffic that does not come from the kamailio process on > >> localhost? > >> > > If You bind asterisk to 127.0.0.1 I think the media connection is set > > for this IP. Your Softphone can not reach the correct 127.0.0.1 > > (localhost is everywhere). > > > > I would suggest, You setup asterisk on eth0 address or 0.0.0.0. In the > > sip.conf You could secure Your setup with > > deny = 0.0.0.0/0.0.0.0 > > permit = Your-LAN-Adress > > This way asterisk accepts SIP from Your box only. > > > This might work, but would need to touch sip.conf every time the IP > address changes. It would be nice to have a configuration that can be > set up once and not modified again. That is why I wanted to set up > localhost. > It is the LAN address of Your Server, where asterisk and kamailio are running. The permit entry allows communication between kamailio and asterisk. Why would You change this address? Maybe I don't understand Your setup. Karsten -- _ -- 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-users
Re: [asterisk-users] Call transfer problem.
You have to use "attendant" transfer, not "blind". - A calls B - B answers on "line 1" (button 1) - B has to use "line 2" (push button 2) to call C, C sees call coming from B, the same does asterisk - while having "line 2" active, he pushes button "transfer" followed by button "line 1" - A speaks with C On Mon, Feb 24, 2014 at 7:45 PM, Mike Diehl wrote: > I'm sorry, I should have mentioned that he's doing a "phone-based" > transfer, not an "asterisk-based" transfer. > > Mike. > > On Mon, Feb 24, 2014 at 1:30 PM, Don Kelly wrote: > > Does he complete the call as a "supervised" transfer--waits for the > called > > party to answer before completing the transfer? > > > > --Don > > > > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl > > Sent: Monday, February 24, 2014 12:24 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [asterisk-users] Call transfer problem. > > > > Hi all, > > > > I have a user who is having trouble transferring calls, using a > Grandstream > > GXP2xxx. > > > > Here's the use case that I've seen: > > > > I call the user from phone A and he answers on phone B. > > > > Then, he hits the transfer button on his phone and dials an extension > that > > is reachable by him, but not by me, based on administrative policy. > > > > However, the Asterisk logs indicate that the new call is being initiated > by > > phone A, not phone B! Thus the call transfer fails. > > > > I have other users, with other phones, that are able to transfer just > fine. > > What could be different with this particular user? > > > > Any ideas? > > > > Mike. > > > > -- > > _ > > -- 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-users > > > > > > -- > > _ > > -- 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-users > > -- > _ > -- 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-users > -- _ -- 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-users
[asterisk-users] SIP 603 Declined error message
I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place
calls inbound, everything works fine. If I place calls outbound, originating
from the Asterisk box, everything works fine (I have done this with the use of
the .call files). If I setup an extension with the findme-followme feature and
have it try to hair-pin a call back out the same trunk to the Avaya, I get a
"SIP/2.0 603 Declined" message. Here is the output.
Any reason that this might be happening? It has been working up until now this
week. I rebooted the machine on Tuesday.
<--- SIP read from TCP:172.17.184.46:31285 --->
INVITE sip:51...@edj.devjones.com SIP/2.0
From: "Haley, Scott"
;tag=8066eb6f589ce3124b652973b4b00
To:
Call-ID: 8066eb6f589ce3125b652973b4b00
CSeq: 1 INVITE
Max-Forwards: 71
Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
Supported: 100rel,histinfo,join,replaces,sdp-anat,timer
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,PRACK,PUBLISH,UPDATE
User-Agent: Avaya CM/R016x.02.0.823.0
Contact: "Haley, Scott"
Route:
Accept-Language: en;q=1
Alert-Info: ;avaya-cm-alert-type=internal
History-Info: ;index=1
History-Info: "51104" ;index=1.1
Min-SE: 1200
P-Asserted-Identity: "Haley, Scott"
Record-Route:
Session-Expires: 1200;refresher=uac
Content-Type: application/sdp
Content-Length: 257
v=0
o=- 1393419743 1 IN IP4 172.17.184.46
s=-
c=IN IP4 172.17.184.93
b=AS:64
t=0 0
a=avf:avc=n prio=n
a=csup:avf-v0
m=audio 28196 RTP/AVP 0 18 127
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
<->
--- (23 headers 13 lines) ---
Sending to 172.17.184.46:31285 (NAT)
Using INVITE request as basis request - 8066eb6f589ce3125b652973b4b00
Found peer 'trunk503in' for '3145152244' from 172.17.184.46:31285
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 127
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 127
Capabilities: us - 0x100c (ulaw|alaw|g722), peer - audio=0x104
(ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1
(telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 172.17.184.93:28196
Looking for 51104 in from-trunk-sip-trunk503out (domain edj.devjones.com)
list_route: hop:
<--- Transmitting (NAT) to 172.17.184.46:31285 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP
172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285
Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
Record-Route:
From: "Haley, Scott"
;tag=8066eb6f589ce3124b652973b4b00
To:
Call-ID: 8066eb6f589ce3125b652973b4b00
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1200;refresher=uac
Contact:
Content-Length: 0
<>
-- Executing [51104@from-trunk-sip-trunk503out:1]
Set("SIP/trunk503in-010b", "GROUP()=OUT_1") in new stack
-- Executing [51104@from-trunk-sip-trunk503out:2]
Goto("SIP/trunk503in-010b", "from-trunk,51104,1") in new stack
-- Goto (from-trunk,51104,1)
-- Executing [51104@from-trunk:1] Set("SIP/trunk503in-010b",
"__FROM_DID=51104") in new stack
-- Executing [51104@from-trunk:2] Gosub("SIP/trunk503in-010b",
"app-blacklist-check,s,1") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/trunk503in-010b",
"0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/trunk503in-010b",
"CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/trunk503in-010b",
"") in new stack
-- Executing [51104@from-trunk:3] Gosub("SIP/trunk503in-010b",
"cidlookup,cidlookup_1,1") in new stack
-- Executing [cidlookup_1@cidlookup:1] GotoIf("SIP/trunk503in-010b",
"1?cidlookup,cidlookup_return,1") in new stack
-- Goto (cidlookup,cidlookup_return,1)
-- Executing [cidlookup_return@cidlookup:1]
ExecIf("SIP/trunk503in-010b", "0?Set(CALLERID(name)=)") in new stack
-- Executing [cidlookup_return@cidlookup:2]
Return("SIP/trunk503in-010b", "") in new stack
-- Executing [51104@from-trunk:4] ExecIf("SIP/trunk503in-010b", "0
?Set(CALLERID(name)=3145152244)") in new stack
-- Executing [51104@from-trunk:5] Set("SIP/trunk503in-010b",
"__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [51104@from-trunk:6] Set("SIP/trunk503in-010b",
"CALLERPRES()=allowed_not_screened") in new stack
-- Executing [51104@from-trunk:7] Goto("SIP/trunk503in-010b",
"app-blackhole,hangup,1") in new stack
-- Goto (app-blackhole,hangup,1)
-- Execu
Re: [asterisk-users] SIP 603 Declined error message
On Wed, Feb 26, 2014 at 8:10 AM, Haley,Scott A
wrote:
> I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place
> calls inbound, everything works fine. If I place calls outbound, originating
> from the Asterisk box, everything works fine (I have done this with the use
> of the .call files). If I setup an extension with the findme-followme
> feature and have it try to hair-pin a call back out the same trunk to the
> Avaya, I get a "SIP/2.0 603 Declined" message. Here is the output.
>
>
>
> Any reason that this might be happening? It has been working up until now
> this week. I rebooted the machine on Tuesday.
>
>
>
> <--- SIP read from TCP:172.17.184.46:31285 --->
>
> INVITE sip:51...@edj.devjones.com SIP/2.0
>
> From: "Haley, Scott"
> ;tag=8066eb6f589ce3124b652973b4b00
>
> To:
>
> Call-ID: 8066eb6f589ce3125b652973b4b00
>
> CSeq: 1 INVITE
>
> Max-Forwards: 71
>
> Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
>
> Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
>
> Supported: 100rel,histinfo,join,replaces,sdp-anat,timer
>
> Allow:
> INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,PRACK,PUBLISH,UPDATE
>
> User-Agent: Avaya CM/R016x.02.0.823.0
>
> Contact: "Haley, Scott"
>
> Route:
>
> Accept-Language: en;q=1
>
> Alert-Info: ;avaya-cm-alert-type=internal
>
> History-Info: ;index=1
>
> History-Info: "51104" ;index=1.1
>
> Min-SE: 1200
>
> P-Asserted-Identity: "Haley, Scott"
>
> Record-Route:
>
> Session-Expires: 1200;refresher=uac
>
> Content-Type: application/sdp
>
> Content-Length: 257
>
>
>
> v=0
>
> o=- 1393419743 1 IN IP4 172.17.184.46
>
> s=-
>
> c=IN IP4 172.17.184.93
>
> b=AS:64
>
> t=0 0
>
> a=avf:avc=n prio=n
>
> a=csup:avf-v0
>
> m=audio 28196 RTP/AVP 0 18 127
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:18 G729/8000
>
> a=fmtp:18 annexb=no
>
> a=rtpmap:127 telephone-event/8000
>
> <->
>
> --- (23 headers 13 lines) ---
>
> Sending to 172.17.184.46:31285 (NAT)
>
> Using INVITE request as basis request - 8066eb6f589ce3125b652973b4b00
>
> Found peer 'trunk503in' for '3145152244' from 172.17.184.46:31285
>
> == Using SIP RTP TOS bits 184
>
> == Using SIP RTP CoS mark 5
>
> Found RTP audio format 0
>
> Found RTP audio format 18
>
> Found RTP audio format 127
>
> Found audio description format PCMU for ID 0
>
> Found audio description format G729 for ID 18
>
> Found audio description format telephone-event for ID 127
>
> Capabilities: us - 0x100c (ulaw|alaw|g722), peer - audio=0x104
> (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
>
> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1
> (telephone-event|), combined - 0x0 (nothing)
>
> Peer audio RTP is at port 172.17.184.93:28196
>
> Looking for 51104 in from-trunk-sip-trunk503out (domain edj.devjones.com)
>
> list_route: hop:
>
>
>
> <--- Transmitting (NAT) to 172.17.184.46:31285 --->
>
> SIP/2.0 100 Trying
>
> Via: SIP/2.0/TCP
> 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285
>
> Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
>
> Record-Route:
>
> From: "Haley, Scott"
> ;tag=8066eb6f589ce3124b652973b4b00
>
> To:
>
> Call-ID: 8066eb6f589ce3125b652973b4b00
>
> CSeq: 1 INVITE
>
> Server: FPBX-2.8.1(1.8.13.0)
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
>
> Supported: replaces, timer
>
> Session-Expires: 1200;refresher=uac
>
> Contact:
>
> Content-Length: 0
>
>
>
>
>
> <>
>
> -- Executing [51104@from-trunk-sip-trunk503out:1]
> Set("SIP/trunk503in-010b", "GROUP()=OUT_1") in new stack
>
> -- Executing [51104@from-trunk-sip-trunk503out:2]
> Goto("SIP/trunk503in-010b", "from-trunk,51104,1") in new stack
>
> -- Goto (from-trunk,51104,1)
>
> -- Executing [51104@from-trunk:1] Set("SIP/trunk503in-010b",
> "__FROM_DID=51104") in new stack
>
> -- Executing [51104@from-trunk:2] Gosub("SIP/trunk503in-010b",
> "app-blacklist-check,s,1") in new stack
>
> -- Executing [s@app-blacklist-check:1] GotoIf("SIP/trunk503in-010b",
> "0?blacklisted") in new stack
>
> -- Executing [s@app-blacklist-check:2] Set("SIP/trunk503in-010b",
> "CALLED_BLACKLIST=1") in new stack
>
> -- Executing [s@app-blacklist-check:3] Return("SIP/trunk503in-010b",
> "") in new stack
>
> -- Executing [51104@from-trunk:3] Gosub("SIP/trunk503in-010b",
> "cidlookup,cidlookup_1,1") in new stack
>
> -- Executing [cidlookup_1@cidlookup:1] GotoIf("SIP/trunk503in-010b",
> "1?cidlookup,cidlookup_return,1") in new stack
>
> -- Goto (cidlookup,cidlookup_return,1)
>
> -- Executing [cidlookup_return@cidlookup:1]
> ExecIf("SIP/trunk503in-010b", "0?Set(CALLERID(name)=)") in new stack
>
> -- Executing [cidlookup_return@cidlookup:2]
> Return("SIP/trunk503in-010b", "") in new stack
>
> -- Executing [51104@from-trunk:4] ExecIf("SIP/trunk503in-010b", "0
> ?Set(
Re: [asterisk-users] DAHDI compile problem
On Tue, Feb 25, 2014 at 11:50:41PM -0800, Doug wrote: > I am having trouble compiling dahdi-linux-complete-2.9.0+2.9.0.1 > on a Raspbien 3.10.25+ kernel. I get the following error - > > /usr/src/dahdi-linux-complete-2.9.0+2.9.0.1/linux/drivers/dahdi/dahdi-base.c:570:2: > error: implicit declaration of function kzalloc > [-Werror=implicit-function-declaration] > > > /usr/src/dahdi-linux-complete-2.9.0+2.9.0.1/linux/drivers/dahdi/dahdi-base.c:1391:2: > error: implicit declaration of function kmalloc > [-Werror=implicit-function-declaration] > > as well as a lot of warnings like - > > /usr/src/dahdi-linux-complete-2.9.0+2.9.0.1/linux/drivers/dahdi/dahdi-base.c:570:6: > warning: assignment makes pointer from integer without a cast [enabled by > default] > > I am obviously missing something. Can anyone help? I have the > kernel source and I have configured it and actually compiled it > successfully. The /lib/modules/3.10.25+/build and source links > are pointing to the kernel source. Odd...if it is only kzalloc, kmalloc, etc.. I wonder if it is resolved if you include linux/slab.h in include/dahdi/kernel.h. Something like: diff --git a/include/dahdi/kernel.h b/include/dahdi/kernel.h index f2f9ec5..c19aec5 100644 --- a/include/dahdi/kernel.h +++ b/include/dahdi/kernel.h @@ -43,6 +43,7 @@ #include #include #include +#include #ifdef CONFIG_DAHDI_NET #include Does that resolve it for you? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- 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-users
Re: [asterisk-users] DAHDI compile problem
Yes it did. Compiled all the way through. I guess that change should go in the release! Thank you, Doug On Wednesday, February 26, 2014 11:19 AM, Shaun Ruffell wrote: On Tue, Feb 25, 2014 at 11:50:41PM -0800, Doug wrote: >> I am having trouble compiling dahdi-linux-complete-2.9.0+2.9.0.1 >> on a Raspbien 3.10.25+ kernel. I get the following error - >> >> /usr/src/dahdi-linux-complete-2.9.0+2.9.0.1/linux/drivers/dahdi/dahdi-base.c:570:2: >> error: implicit declaration of function kzalloc >> [-Werror=implicit-function-declaration] >> >> >> /usr/src/dahdi-linux-complete-2.9.0+2.9.0.1/linux/drivers/dahdi/dahdi-base.c:1391:2: >> error: implicit declaration of function kmalloc >> [-Werror=implicit-function-declaration] >> >> as well as a lot of warnings like - >> >> /usr/src/dahdi-linux-complete-2.9.0+2.9.0.1/linux/drivers/dahdi/dahdi-base.c:570:6: >> warning: assignment makes pointer from integer without a cast [enabled by >> default] >> >> I am obviously missing something. Can anyone help? I have the >> kernel source and I have configured it and actually compiled it >> successfully. The /lib/modules/3.10.25+/build and source links >> are pointing to the kernel source. > >Odd...if it is only kzalloc, kmalloc, etc.. I wonder if it is >resolved if you include linux/slab.h in include/dahdi/kernel.h. >Something like: > > diff --git a/include/dahdi/kernel.h b/include/dahdi/kernel.h > index f2f9ec5..c19aec5 100644 > --- a/include/dahdi/kernel.h > +++ b/include/dahdi/kernel.h > @@ -43,6 +43,7 @@ > #include > #include > #include > +#include > > #ifdef CONFIG_DAHDI_NET > #include > >Does that resolve it for you? > >-- >Shaun Ruffell >Digium, Inc. | Linux Kernel Developer >445 Jan Davis Drive NW - Huntsville, AL 35806 - USA >Check us out at: www.digium.com & www.asterisk.org > > >-- _ -- 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-users
Re: [asterisk-users] Problem with "pri show channels" on rasterisk (11.7)
Sorry to insist. Maybe I should submit this as a bug? Thanks. On Tue, Feb 25, 2014 at 7:57 PM, Rodrigo Borges Pereira < rodrigoborgespere...@gmail.com> wrote: > Hi, > > I'm running a test system with Ast 11.7 and DAHDI 2.9.0. I loaded a TE205P > card. Then I enter asterisk console, and once I do a "pri show channels", > the console no longer works correctly. There's no output for any command > after that. If I type "pri show" and hit tab for completion, console > freezes there, I need to kill session. > > Any hints? new bug? known issue? > > Tks > -- _ -- 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-users
Re: [asterisk-users] Problem with "pri show channels" on rasterisk (11.7)
On Wed, Feb 26, 2014 at 05:34:02PM +, Rodrigo Borges Pereira wrote: > Sorry to insist. Maybe I should submit this as a bug? > > Thanks. As the owner of a TE205P, it might be easiest to contact Digium's technical support for installation assistance. I personally am not aware of any general issues when running "pri show channels" > On Tue, Feb 25, 2014 at 7:57 PM, Rodrigo Borges Pereira < > rodrigoborgespere...@gmail.com> wrote: > > > Hi, > > > > I'm running a test system with Ast 11.7 and DAHDI 2.9.0. I loaded a TE205P > > card. Then I enter asterisk console, and once I do a "pri show channels", > > the console no longer works correctly. There's no output for any command > > after that. If I type "pri show" and hit tab for completion, console > > freezes there, I need to kill session. > > > > Any hints? new bug? known issue? > > > > Tks > > -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- 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-users
Re: [asterisk-users] Problem with "pri show channels" on rasterisk (11.7)
Indeed, I tested two more cards on this system, and I don't get the same problem with them. Unfortunately, different models (TE122P and TE110P), but still it seems to suggest a problem with the TE205P. Weird manifestation though, because everything else seems ok with, even a patlooptest is just fine. Too bad it's OOW by now :) Thanks. On Wed, Feb 26, 2014 at 7:21 PM, Shaun Ruffell wrote: > On Wed, Feb 26, 2014 at 05:34:02PM +, Rodrigo Borges Pereira wrote: > > Sorry to insist. Maybe I should submit this as a bug? > > > > Thanks. > > As the owner of a TE205P, it might be easiest to contact Digium's > technical support for installation assistance. > > I personally am not aware of any general issues when running "pri > show channels" > > > On Tue, Feb 25, 2014 at 7:57 PM, Rodrigo Borges Pereira < > > rodrigoborgespere...@gmail.com> wrote: > > > > > Hi, > > > > > > I'm running a test system with Ast 11.7 and DAHDI 2.9.0. I loaded a > TE205P > > > card. Then I enter asterisk console, and once I do a "pri show > channels", > > > the console no longer works correctly. There's no output for any > command > > > after that. If I type "pri show" and hit tab for completion, console > > > freezes there, I need to kill session. > > > > > > Any hints? new bug? known issue? > > > > > > Tks > > > > > -- > Shaun Ruffell > Digium, Inc. | Linux Kernel Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- 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-users > -- _ -- 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-users
Re: [asterisk-users] Problem with "pri show channels" on rasterisk (11.7)
I actually meant that card specifically, not a general issue with the model. But I defer more considerations to when I have the chance to test another TE205. > On 26/02/2014, at 22:22, Shaun Ruffell wrote: > >> On Wed, Feb 26, 2014 at 09:03:49PM +, Rodrigo Borges Pereira wrote: >> Indeed, I tested two more cards on this system, and I don't get the same >> problem with them. Unfortunately, different models (TE122P and TE110P), but >> still it seems to suggest a problem with the TE205P. Weird manifestation >> though, because everything else seems ok with, even a patlooptest is just >> fine. Too bad it's OOW by now :) >> >> Thanks. > > I would be very surprised this has anything to do with the TE205P > specifically. The command "pri show channels" does not result in any > calls to the drivers. My guess is this has more to do with the state > / configuration of Asterisk. > > -- > Shaun Ruffell > Digium, Inc. | Linux Kernel Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- 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-users -- _ -- 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-users
Re: [asterisk-users] Problem with "pri show channels" on rasterisk (11.7)
On Wed, Feb 26, 2014 at 09:03:49PM +, Rodrigo Borges Pereira wrote: > Indeed, I tested two more cards on this system, and I don't get the same > problem with them. Unfortunately, different models (TE122P and TE110P), but > still it seems to suggest a problem with the TE205P. Weird manifestation > though, because everything else seems ok with, even a patlooptest is just > fine. Too bad it's OOW by now :) > > Thanks. I would be very surprised this has anything to do with the TE205P specifically. The command "pri show channels" does not result in any calls to the drivers. My guess is this has more to do with the state / configuration of Asterisk. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- 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-users
