VoIP Mailing List Archives

[bord at staff.onthenet...]

Hello,

I’m having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends ‘@CUBE’ onto the end of the dialled number, as per the following examples;

Asterisk log;
app_dial.c: Called SIP/test/0429123456@CUBE
chan_sip.c: Got SIP response 500 "Internal Server Error" back from 172.22.4.12:5060

In the SIP SDP;
INVITE sip:0429920437%40CUBE@172.22.4.12 SIP/2.0.
To: <sip:0429920437%40CUBE@172.22.4.12>.

As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I’m really not sure where this is coming from, and why.

Here is my trunk configuration;

PEER
type=friend
qualify=yes
nat=no
insecure=port,invite
host=172.22.4.12
dtmfmode=rfc2833
context=from-trunk
allow=ulaw
disallow=all

USER
type=friend
qualify=yes
nat=no
host=172.22.4.12
dtmfmode=rfc2833
allow=ulaw
disallow=all
canreinvite=no

Thanks for any help J

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au

[bferrell at baywinds.org]

Brenden,

check the context, from-trunk, in the dialplan.  Thtat's where this is being added

 

On 8/17/15 5:33 PM, Brendan Ord wrote:

[bord at staff.onthenet...]

Hi Bruce,

At the risk of sounding dumb J And, realising that I mustn’t know how context’s work properly (I guess they aren’t like Calling Search Spaces in Cisco-land).

I tried changing the context to from-internal and from-pstn with no change to @CUBE being appended.

The from-trunk context looks pretty long, and includes a heap of other contexts as well – this is all default configuration in FreePBX. I assume they’re doing most of the context leg work for us already in their distro’s …

Is there something I can stick in an email which might give a hint to my problem, or somewhere I should be looking? I was looking in extensions.conf at the contexts defined in there, was I in the wrong place?

Thanks in advance,

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au


From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Bruce Ferrell
Sent: Tuesday, 18 August 2015 10:38 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number



Brenden,

check the context, from-trunk, in the dialplan. Thtat's where this is being added



On 8/17/15 5:33 PM, Brendan Ord wrote:

[bord at staff.onthenet...]

I recreated the trunk, and still get @CUBE added to the end of it.

I’m confused, nowhere in any configuration does this word appear anymore. How would Asterisk stick this on? I grepped the entire conf directory looking for a mention of it, and nothing ..

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au


From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brendan Ord
Sent: Tuesday, 18 August 2015 11:31 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number



Hi Bruce,

At the risk of sounding dumb J And, realising that I mustn’t know how context’s work properly (I guess they aren’t like Calling Search Spaces in Cisco-land).

I tried changing the context to from-internal and from-pstn with no change to @CUBE being appended.

The from-trunk context looks pretty long, and includes a heap of other contexts as well – this is all default configuration in FreePBX. I assume they’re doing most of the context leg work for us already in their distro’s …

Is there something I can stick in an email which might give a hint to my problem, or somewhere I should be looking? I was looking in extensions.conf at the contexts defined in there, was I in the wrong place?

Thanks in advance,

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au


From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Bruce Ferrell
Sent: Tuesday, 18 August 2015 10:38 AM
To: asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number



Brenden,

check the context, from-trunk, in the dialplan. Thtat's where this is being added



On 8/17/15 5:33 PM, Brendan Ord wrote:

[dcunningham at voisoni...]

Hi Brendan,


Can you attach an Asterisk log with "sip set debug on", "core set verbose 9" and "core set debug 9"?



On 18 August 2015 at 10:33, Brendan Ord <bord@staff.onthenet.com.au (bord@staff.onthenet.com.au)> wrote:

[bord at staff.onthenet...]

Hi David,

http://pastebin.com/R4bsnmX7

I’ll start going through this as well and see if I can see anything.

Thanks for your help,

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David Cunningham
Sent: Tuesday, 18 August 2015 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number

Hi Brendan,

Can you attach an Asterisk log with "sip set debug on", "core set verbose 9" and "core set debug 9"?


On 18 August 2015 at 10:33, Brendan Ord <bord@staff.onthenet.com.au (bord@staff.onthenet.com.au)> wrote:
Hello,

I’m having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends ‘@CUBE’ onto the end of the dialled number, as per the following examples;

Asterisk log;
app_dial.c: Called SIP/test/0429123456@CUBE
chan_sip.c: Got SIP response 500 "Internal Server Error" back from 172.22.4.12:5060

In the SIP SDP;
INVITE sip:0429920437%40CUBE@172.22.4.12 ([email]sip%3A0429920437%2540CUBE@172.22.4.12[/email]) SIP/2.0.
To: <sip:0429920437%40CUBE@172.22.4.12 ([email]sip%3A0429920437%2540CUBE@172.22.4.12[/email])>.

As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I’m really not sure where this is coming from, and why.

Here is my trunk configuration;

PEER
type=friend
qualify=yes
nat=no
insecure=port,invite
host=172.22.4.12
dtmfmode=rfc2833
context=from-trunk
allow=ulaw
disallow=all

USER
type=friend
qualify=yes
nat=no
host=172.22.4.12
dtmfmode=rfc2833
allow=ulaw
disallow=all
canreinvite=no

Thanks for any help J

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au




--
_____________________________________________________________________
-- 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



--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019

[bord at staff.onthenet...]

David,

I should also note;

246 is my extension, it has IP 172.22.3.238.

172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway.

The trunk is called ‘testing’ at the moment. The route that selects this trunk uses a 9 prefix.

This system is in semi-production, so there might be fluff in the log from other active calls.

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David Cunningham
Sent: Tuesday, 18 August 2015 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number

Hi Brendan,

Can you attach an Asterisk log with "sip set debug on", "core set verbose 9" and "core set debug 9"?


On 18 August 2015 at 10:33, Brendan Ord <bord@staff.onthenet.com.au (bord@staff.onthenet.com.au)> wrote:
Hello,

I’m having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends ‘@CUBE’ onto the end of the dialled number, as per the following examples;

Asterisk log;
app_dial.c: Called SIP/test/0429123456@CUBE
chan_sip.c: Got SIP response 500 "Internal Server Error" back from 172.22.4.12:5060

In the SIP SDP;
INVITE sip:0429920437%40CUBE@172.22.4.12 ([email]sip%3A0429920437%2540CUBE@172.22.4.12[/email]) SIP/2.0.
To: <sip:0429920437%40CUBE@172.22.4.12 ([email]sip%3A0429920437%2540CUBE@172.22.4.12[/email])>.

As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I’m really not sure where this is coming from, and why.

Here is my trunk configuration;

PEER
type=friend
qualify=yes
nat=no
insecure=port,invite
host=172.22.4.12
dtmfmode=rfc2833
context=from-trunk
allow=ulaw
disallow=all

USER
type=friend
qualify=yes
nat=no
host=172.22.4.12
dtmfmode=rfc2833
allow=ulaw
disallow=all
canreinvite=no

Thanks for any help J

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au




--
_____________________________________________________________________
-- 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



--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019

[bord at staff.onthenet...]

Starting to make sense when I saw this line:

[2015-08-18 15:01:33] DEBUG[19366][C-00001cfc]: pbx.c:3785 ast_str_retrieve_variable: Result of 'OUT_3_SUFFIX' is '@CUBE'

But I can’t find where this is in configuration ..

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au


From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brendan Ord
Sent: Tuesday, 18 August 2015 3:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number



David,

I should also note;

246 is my extension, it has IP 172.22.3.238.

172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway.

The trunk is called ‘testing’ at the moment. The route that selects this trunk uses a 9 prefix.

This system is in semi-production, so there might be fluff in the log from other active calls.

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David Cunningham
Sent: Tuesday, 18 August 2015 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number

Hi Brendan,

Can you attach an Asterisk log with "sip set debug on", "core set verbose 9" and "core set debug 9"?


On 18 August 2015 at 10:33, Brendan Ord <bord@staff.onthenet.com.au (bord@staff.onthenet.com.au)> wrote:
Hello,

I’m having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends ‘@CUBE’ onto the end of the dialled number, as per the following examples;

Asterisk log;
app_dial.c: Called SIP/test/0429123456@CUBE
chan_sip.c: Got SIP response 500 "Internal Server Error" back from 172.22.4.12:5060

In the SIP SDP;
INVITE sip:0429920437%40CUBE@172.22.4.12 ([email]sip%3A0429920437%2540CUBE@172.22.4.12[/email]) SIP/2.0.
To: <sip:0429920437%40CUBE@172.22.4.12 ([email]sip%3A0429920437%2540CUBE@172.22.4.12[/email])>.

As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I’m really not sure where this is coming from, and why.

Here is my trunk configuration;

PEER
type=friend
qualify=yes
nat=no
insecure=port,invite
host=172.22.4.12
dtmfmode=rfc2833
context=from-trunk
allow=ulaw
disallow=all

USER
type=friend
qualify=yes
nat=no
host=172.22.4.12
dtmfmode=rfc2833
allow=ulaw
disallow=all
canreinvite=no

Thanks for any help J

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au




--
_____________________________________________________________________
-- 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



--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019

[dcunningham at voisoni...]

Yes indeed.

Do you have the dialplan, eg from /etc/asterisk/extensions.conf?

Something is getting this OUT_3_SUFFIX variable and including it in a Dial to 172.22.4.12.



On 18 August 2015 at 16:21, Brendan Ord <bord@staff.onthenet.com.au (bord@staff.onthenet.com.au)> wrote:

[bferrell at baywinds.org]

just got back to my mail.

What I'd do is go into /etc/asterisk and grep for OUT_3_SUFFIX in all the files

once the file with that variable is located, we can figure out why it's adding it



On 08/17/2015 11:26 PM, David Cunningham wrote:

[bord at staff.onthenet...]

Hello,

So, I found this line under macro-dialout-trunk, in extensions_additional.conf (FreePBX, so it controls the conf files mostly);

exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS})

If I grep for OUT_3_SUFFIX in all files in /etc/asterisk I get nothing..

Here's a paste of a few things out of the two files that I thought were relevant to how FreePBX configured this trunk ...

http://pastebin.com/5fRy2Ai9


Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au

 

  
NOTICE:

This e-mail and any attachments are private and confidential and may contain privileged information. If you are not an authorised recipient, the copying or distribution of this e-mail and any attachments is prohibited and you must not read, print or act in reliance on this e-mail or attachments. Any pricing information supplied via email is an estimate or indicative only and may require a formal quotation to verify full terms and conditions.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Bruce Ferrell
Sent: Tuesday, 18 August 2015 4:38 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number

just got back to my mail.

What I'd do is go into /etc/asterisk and grep for OUT_3_SUFFIX in all the files

once the file with that variable is located, we can figure out why it's adding it



On 08/17/2015 11:26 PM, David Cunningham wrote:

[bord at staff.onthenet...]

Halt the wild goose chase ....


It was obviously something left over in the dial plan. Restarted Asterisk seeing as though we're now after-hours and I can do interruptive work, and it seems to have solved my @CUBE problem.

Interestingly, it persisted through a "dialplan reload" and the equivalent of a "core reload" too ..

[2015-08-18 17:04:30] VERBOSE[25543][C-00000000] app_dial.c: Called SIP/testing/0429920437
[2015-08-18 17:04:30] VERBOSE[25543][C-00000000] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)

This is expected, I need to review the dial-peer configurations on the Cisco GW. At least it isn't throwing the suffix on the end anymore it seems...

Thanks for the help and apologies for the goose chase ..

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au

 

  
NOTICE:

This e-mail and any attachments are private and confidential and may contain privileged information. If you are not an authorised recipient, the copying or distribution of this e-mail and any attachments is prohibited and you must not read, print or act in reliance on this e-mail or attachments. Any pricing information supplied via email is an estimate or indicative only and may require a formal quotation to verify full terms and conditions.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brendan Ord
Sent: Tuesday, 18 August 2015 4:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number

Hello,

So, I found this line under macro-dialout-trunk, in extensions_additional.conf (FreePBX, so it controls the conf files mostly);

exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS})

If I grep for OUT_3_SUFFIX in all files in /etc/asterisk I get nothing..

Here's a paste of a few things out of the two files that I thought were relevant to how FreePBX configured this trunk ...

http://pastebin.com/5fRy2Ai9


Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map) www.OntheNet.com.au

 

  
NOTICE:

This e-mail and any attachments are private and confidential and may contain privileged information. If you are not an authorised recipient, the copying or distribution of this e-mail and any attachments is prohibited and you must not read, print or act in reliance on this e-mail or attachments. Any pricing information supplied via email is an estimate or indicative only and may require a formal quotation to verify full terms and conditions.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Bruce Ferrell
Sent: Tuesday, 18 August 2015 4:38 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number

just got back to my mail.

What I'd do is go into /etc/asterisk and grep for OUT_3_SUFFIX in all the files

once the file with that variable is located, we can figure out why it's adding it



On 08/17/2015 11:26 PM, David Cunningham wrote:

[dcunningham at voisoni...]

Glad to hear it's sorted.



On 18 August 2015 at 17:08, Brendan Ord <bord@staff.onthenet.com.au (bord@staff.onthenet.com.au)> wrote: