Re: [asterisk-users] DUNDI anyone?
DUNDI was a great idea and we saw it deployed, but we've watched clients struggle with it. And many eventually give up on it. I don't consider it overly complex, but I suspect it's proper (and safe) configuration is beyond a lot of tel admins. Im curious what the state of dundi deployments is. -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benoit Panizzon Sent: Tuesday, May 2, 2023 7:45 AM To: Asterisk Users Subject: [asterisk-users] DUNDI anyone? Hi Well it is well some time that my last DUNDI peer has become unreachable. I guess too many issues with spoofed numbers etc. But I am wondering, do people, especially larger entities like telcos, still use DUNDI? I know that in some Hamradio communities, DUNDI is used to interconnect PBXes, but that is with private phone number ranges, not connected to the public. Want some DUNDI peering? DM me :-) Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- 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
[asterisk-users] DUNDI anyone?
Hi Well it is well some time that my last DUNDI peer has become unreachable. I guess too many issues with spoofed numbers etc. But I am wondering, do people, especially larger entities like telcos, still use DUNDI? I know that in some Hamradio communities, DUNDI is used to interconnect PBXes, but that is with private phone number ranges, not connected to the public. Want some DUNDI peering? DM me :-) Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- 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] DUNDi peers disconnect after being connected for months or years, cannot reconnect again
It's falling down on the DUNDi qualify setting. If I change qualify=yes to qualify=no on the entry in dundi_peers_custom.conf Then it will properly route the calls. It's just a workaround, because I still don't know why the qualify process isn't working, and now I have no visibility on the connectivity of the peers. Thank you, Court Campbell IT Cybersecurity Manager Flex-N-Gate Direct Line: 705-749-4116 Office: 705-742-3534,22303 Cell: 905-252-1091 E-Mail: ccampb...@flexngate.com -Original Message- From: Court Campbell Sent: Wednesday, July 27, 2022 3:58 PM To: aster...@phreaknet.org Cc: Asterisk Users Subject: RE: [asterisk-users] DUNDi peers disconnect after being connected for months or years, cannot reconnect again This is what I get when I run "dundi set debug on". I've changed the IPs to DUNDi Server 1 and 2. DUNDi peers always show UNREACHABLE Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 26713 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 11263 DTrans: 0 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 26713 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 11263 DTrans: 0 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 26713 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 11263 DTrans: 0 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 26713 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 11263 DTrans: 0 [DUNDi Server 2:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 29321 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 001 Type: ACK (Response) Flags: 00 STrans: 22000 DTrans: 29321 [DUNDi Server 1:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 04905 DTrans: 0 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 001 Type: ACK (Response) Flags: 00 STrans: 20346 DTrans: 04905 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 26713 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 11263 DTrans: 0 [DUNDi Server 2:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 29321 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 001 Type: ACK (Response) Flags: 00 STrans: 12424 DTrans: 29321 [DUNDi Server 1:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 04905 DTrans: 0 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 001 Type: ACK (Response) Flags: 00 STrans: 02609 DTrans: 04905 [DUNDi Server 2:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 29321 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 001 Type: ACK (Response) Flags: 00 STrans: 03011 DTrans: 29321 [DUNDi Server 1:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Thank you, Court Campbell -Original Message- From: aster...@phreaknet.org Sent: Wednesday, July 27, 2022 12:55 PM To: Court Campbell Cc: Asterisk Users Subject: Re: [asterisk-users] DUNDi peers disconnect after being connected for months or years, cannot reconnect again WARNING: This email originated from outside of Flex-N-Gate, and it may contain dangerous attachments or links that may steal your credentials or infect your computer. Please be cautious when opening attachments, clicking links or responding to this email. On 7/27/2022 11:34 AM, Court Campbell wrote: > > I realize that the heyday of DUNDi was about 2008, and that there's > less and less information online about it and lots of people don't use > it anymore and use static IAX trunks instead. But we have 53 asterisk > phone systems connecting our locations, and so creating static IAX > trunks (even with a regional hub and spoke model) is a significant > undertaking. > > We've had issues with DUNDi peers disconnecting in the past and with > Elastix used to have to do an amportal restart on the PBX to have it > reconnect again. But the new issue that we're seeing is that the peer > refuses to reconn
Re: [asterisk-users] DUNDi peers disconnect after being connected for months or years, cannot reconnect again
This is what I get when I run "dundi set debug on". I've changed the IPs to DUNDi Server 1 and 2. DUNDi peers always show UNREACHABLE Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 26713 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 11263 DTrans: 0 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 26713 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 11263 DTrans: 0 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 26713 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 11263 DTrans: 0 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 26713 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 11263 DTrans: 0 [DUNDi Server 2:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 29321 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 001 Type: ACK (Response) Flags: 00 STrans: 22000 DTrans: 29321 [DUNDi Server 1:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 04905 DTrans: 0 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 001 Type: ACK (Response) Flags: 00 STrans: 20346 DTrans: 04905 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 26713 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 11263 DTrans: 0 [DUNDi Server 2:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 29321 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 001 Type: ACK (Response) Flags: 00 STrans: 12424 DTrans: 29321 [DUNDi Server 1:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 04905 DTrans: 0 [DUNDi Server 2:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 001 Type: ACK (Response) Flags: 00 STrans: 02609 DTrans: 04905 [DUNDi Server 2:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Flags: 00 STrans: 29321 DTrans: 0 [DUNDi Server 1:4520] (Final) Tx-Frame -- OSeqno: 000 ISeqno: 001 Type: ACK (Response) Flags: 00 STrans: 03011 DTrans: 29321 [DUNDi Server 1:4520] (Final) Rx-Frame -- OSeqno: 000 ISeqno: 000 Type: NULL (Command) Thank you, Court Campbell -Original Message- From: aster...@phreaknet.org Sent: Wednesday, July 27, 2022 12:55 PM To: Court Campbell Cc: Asterisk Users Subject: Re: [asterisk-users] DUNDi peers disconnect after being connected for months or years, cannot reconnect again WARNING: This email originated from outside of Flex-N-Gate, and it may contain dangerous attachments or links that may steal your credentials or infect your computer. Please be cautious when opening attachments, clicking links or responding to this email. On 7/27/2022 11:34 AM, Court Campbell wrote: > > I realize that the heyday of DUNDi was about 2008, and that there's > less and less information online about it and lots of people don't use > it anymore and use static IAX trunks instead. But we have 53 asterisk > phone systems connecting our locations, and so creating static IAX > trunks (even with a regional hub and spoke model) is a significant > undertaking. > > We've had issues with DUNDi peers disconnecting in the past and with > Elastix used to have to do an amportal restart on the PBX to have it > reconnect again. But the new issue that we're seeing is that the peer > refuses to reconnect again, even after changing the long secret key in > dundi_general, the MAC address of the VM the PBX is running on, the IP > of the PBX and recreating the public/private keypair. > Do you have any logs/debug demonstrating this? > > The PBX that disconnected will not reconnect to any DUNDi peers at > all, even to a completely new VM that we stand up at the same site on > the same virtual switch on the same host. We are only trying to do > inter-site extension dialing, not routing external calling between > sites using DUNDi. > > I'm also okay with giving up on DUNDi if anybody else has a less > labour intensive way of routing extension dialing between that many > PBXes than a web of static IAX tru
[asterisk-users] DUNDi peers disconnect after being connected for months or years, cannot reconnect again
I realize that the heyday of DUNDi was about 2008, and that there's less and
less information online about it and lots of people don't use it anymore and
use static IAX trunks instead. But we have 53 asterisk phone systems connecting
our locations, and so creating static IAX trunks (even with a regional hub and
spoke model) is a significant undertaking.
We've had issues with DUNDi peers disconnecting in the past and with Elastix
used to have to do an amportal restart on the PBX to have it reconnect again.
But the new issue that we're seeing is that the peer refuses to reconnect
again, even after changing the long secret key in dundi_general, the MAC
address of the VM the PBX is running on, the IP of the PBX and recreating the
public/private keypair. The PBX that disconnected will not reconnect to any
DUNDi peers at all, even to a completely new VM that we stand up at the same
site on the same virtual switch on the same host. We are only trying to do
inter-site extension dialing, not routing external calling between sites using
DUNDi.
I'm also okay with giving up on DUNDi if anybody else has a less labour
intensive way of routing extension dialing between that many PBXes than a web
of static IAX trunks.
Here are the config files. I removed all the ; commented lines in dundi.conf to
save space.
Iax_custom.conf
[dundi]
type=user
dbsecret=dundi/secret
context=ext-local
disallow=all
allow=ulaw
allow=g726
dundi.conf
[general]
#include dundi_general_custom.conf
ttl=32
autokill=yes
[mappings]
#include dundi_mappings_custom.conf
#include dundi_peers_custom.conf
dundi_mappings.conf
priv => dundi-priv-canonical,0,IAX2,dundi:${SECRET}@PBX_IP/${NUMBER},nopartial
priv => dundi-priv-customers,100,IAX2,dundi:${SECRET}@PBX_IP/${NUMBER},nopartial
priv => dundi-priv-via-pstn,400,IAX2,dundi:${SECRET}@PBX_IP/${NUMBER},nopartial
dundi_general.conf
organization=
locality=
stateprov=
country=
email=
phone=
department=
secret=secret key
entityid=MAC address
dundi_peers.conf
[server A MAC]
model=symmetric
host=
inkey=
outkey=
his_status=connected
include=priv
permit=priv
qualify=yes
order=primary
[server B MAC]
model=symmetric
host=
inkey=
outkey=
his_status=connected
include=priv
permit=priv
qualify=yes
order=primary
extensions_custom.conf
[from-internal]
include => from-internal-noxfer
include => from-internal-xfer
include => dundi-priv-lookup
include => bad-number ; auto-generated
exten => h,1,Macro(hangupcall)
;
; CONFIGURACION PARA DUNDi
[dundi-priv-canonical]
; Here we include the context that contains the extensions.
exten => _X,1,Macro(stdexten,${EXTEN})
;include => ext-local
; Here we include the context that contains the queues.
; include => ext-queues
[dundi-priv-customers]
; If you have customers (or resell services) we can list them here
[dundi-priv-via-pstn]
; Here we include the context with our trunk to the PSTN,
; if we want the other teams can use our trunks
;include => outbound-allroutes
[dundi-priv-local]
; In this context we unify the three contexts, we can use this as
; context of the trunks of dundi iax
include => dundi-priv-canonical
include => dundi-priv-customers
include => dundi-priv-via-pstn
[dundi-priv-lookup]
; This context is responsible for making the search for a number of dundi
; Before you do the search properly define our caller id.
; because if not we have a caller id as 'device<>'.
exten => _X.,1,Macro(user-callerid)
exten => _X.,n,Macro(dundi-priv,${EXTEN})
exten => _X.,n,GotoIf($['${DIALSTATUS}' = 'BUSY']?100)
exten => _X.,n,Goto(bad-number,${EXTEN},1)
exten => _X.,100,Playtones(congestion)
exten => _X.,101,Congestion(10)
[macro-dundi-priv]
; This is the macro is called from the context [dundi-priv-lookup]
; It also avoids having loops in the consultations dundi.
exten => s,1,Goto(${ARG1},1)
switch => DUNDi/priv
;
Thank you,
Court Campbell
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Re: [asterisk-users] DUNDI with minimal features
You could use IMAP storage for your voicemail to solve this I think. Have both PBXes use the same storage. I don't know if this works for you or not, but you might consider a single PBX and just combine the two offices under one installation. We do this for a lot of our customers (actually we host their PBX, but all the offices' phones connect to it). If both offices have good connectivity, and especially if you have a QoS enabled VPN between them, this could work well. Cheers, j On 03/27/2019 12:43 PM, Janet wrote: Great document thank you! I will have to experiment with this on a couple of test systems. Something I'm not clear on, if a user receives a voicemail on one of the PBX's, does DUNDI handle retrieving the message from the right system? Or if the user tries to retrieve a voicemail on PBX A but the message was left on PBX B, they won't hear it. Janet -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JR Richardson Sent: Wednesday, March 27, 2019 2:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DUNDI with minimal features I have 2 PBX's, one in each office (say one in New York, one in Boston). I have mobile users that can show up at either office and connect their soft phones. Is there a very simple DUNDI config available which describes how to set this up? Also, can I have the same outbound trunks setup in each office, so that calls don't have to route across the NY-BOS connection to get out to the PSTN? Thanks, Janet Not sure how relevant on newer versions, but yes, pretty easy to setup. http://blog.sinologic.net/wp-content/uploads/2007/08/DUNDi_So_Easy.pdf Good luck!. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- 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 -- Jeff LaCoursiere StratusTalk, Inc. 703 496 4990 x108 815 546 6599 cell j...@stratustalk.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
Re: [asterisk-users] DUNDI with minimal features
Great document thank you! I will have to experiment with this on a couple of test systems. Something I'm not clear on, if a user receives a voicemail on one of the PBX's, does DUNDI handle retrieving the message from the right system? Or if the user tries to retrieve a voicemail on PBX A but the message was left on PBX B, they won't hear it. Janet -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JR Richardson Sent: Wednesday, March 27, 2019 2:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DUNDI with minimal features > I have 2 PBX's, one in each office (say one in New York, one in > Boston). I have mobile users that can show up at either office and > connect their soft phones. > > > > Is there a very simple DUNDI config available which describes how to > set this up? > > Also, can I have the same outbound trunks setup in each office, so > that calls don't have to route across the NY-BOS connection to get out > to the PSTN? > > > > Thanks, Janet Not sure how relevant on newer versions, but yes, pretty easy to setup. http://blog.sinologic.net/wp-content/uploads/2007/08/DUNDi_So_Easy.pdf Good luck!. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- 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] DUNDI with minimal features
> I have 2 PBX's, one in each office (say one in New York, one in Boston). I > have mobile users that can show up at either office and connect their soft > phones. > > > > Is there a very simple DUNDI config available which describes how to set > this up? > > Also, can I have the same outbound trunks setup in each office, so that > calls don't have to route across the NY-BOS connection to get out to the > PSTN? > > > > Thanks, Janet Not sure how relevant on newer versions, but yes, pretty easy to setup. http://blog.sinologic.net/wp-content/uploads/2007/08/DUNDi_So_Easy.pdf Good luck!. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- 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
[asterisk-users] DUNDI with minimal features
I have 2 PBX's, one in each office (say one in New York, one in Boston). I have mobile users that can show up at either office and connect their soft phones. Is there a very simple DUNDI config available which describes how to set this up? Also, can I have the same outbound trunks setup in each office, so that calls don't have to route across the NY-BOS connection to get out to the PSTN? Thanks, Janet -- _ -- 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] DUNDi with SIP Mapping
On Wed, Apr 16, 2014 at 10:20 AM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
You are a bit outside of what I have done, but this looks like it might be
what you want to do with SIP:
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP
I had looked at that guide before, but couldn't get it working. I could do
SIP without authentication. This would have worked if I only wanted to
terminate calls to extensions. For future purposes I wanted to include PSTN
routes. In the end I went with IAX and have it up and running. It was
actually simple to integrate with FreePBX. The important piece was setting
ttl to 1 to prevent DUNDi lookup loops, which would cause the box to
sometimes see its own DUNDi extensions.
The one FreePBX box with the PRI will try 10 digits numbers on DUNDi
private then go out the PRI. The other FreePBX boxes try to dial 10 digit
numbers on DUNDi private then use DUNDi to reach the PSTN. This allows me
to add additionally FreePBX boxes with PSTN connections and use weights.
Additionally providing a separate mapping for the PSTN allows toll free to
first try DUNDi private, then a VoIP provider, then the DUNDi PSTN.
cd /var/lib/asterisk/keys
astgenkey -n `hostname -f`
chown asterisk:asterisk *
share .pub keys between all servers
vim /etc/asterisk/dundi.conf
cachetime=60
ttl=1
priv = dundi-extens,0,IAX2,dundi:${
SECRET}@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial
priv = dundi-dids,100,IAX2,dundi:${
SECRET}@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial
pstn = dundi-via-pstn,400,IAX2,dundi:${
SECRET}@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial
;[EID of remote DUNDi peer]
;model = symmetric
;host = IP or FQDN of remote DUNDi peer
;inkey = public key of remote DUNDi peer, without .pub
;outkey = private key of local PBX, without .key
;include = all
;permit = all
;qualify = yes
vim /etc/asterisk/extensions_custom.conf
[dundi-local]
include = dundi-extens
include = dundi-dids
include = dundi-via-pstn
[dundi-local-keepcid]
exten = _X.,1,Set(KEEPCID=TRUE)
exten = _X.,n,Goto(dundi-local,${EXTEN},1)
[dundi-extens]
include = ext-queues
include = ext-findmefollow
include = ext-group
include = ext-local
[dundi-dids]
include = ext-did-0002
[dundi-via-pstn]
include = outbound-allroutes
FreePBX Trunks
Type: DUNDi
Trunk Name: DUNDi Private
DUNDi Mapping: priv
Type: DUNDi
Trunk Name: DUNDi Pstn
DUNDi Mapping: pstn
Type: IAX
Trunk Name: DUNDi
Outgoing Settings:
Trunk Name: dundi
PEER Details:
type=friend
dbsecret=dundi/secret
disallow=all
context=dundi-local-keepcid
allow=ulawg729
FreePBX Outbound Routes
Route Name: dundi
Route Type: Intra-Company
Dial Pattern: NXXX
Trunk: DUNDi Private
Route Name: outbound
Dial Pattern: 1NXXNXX
Dial Pattern: NXXNXX
Trunk: DUNDi Private
Trunk: PRI or DUNDi Pstn
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[asterisk-users] DUNDi with SIP Mapping
From the reading and testing I have done it doesn't look like SIP supports
a username and password in the Dial string. I currently have the following
mapping.
priv = dundi-extens,0,SIP,
dundi:pass@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial
On the sending side I see
NOTICE[31598] chan_sip.c: Conflicting extension values given. Using 'dundi'
and not '1001'
On the receiving side it will not match the SIP dundi user and tries to
call dundi instead of 1001.
-- Executing [dundi@from-sip-external:1] NoOp(SIP/1.1.1.2-,
Received incoming SIP connection from unknown peer to dundi) in new stack
Is there a way to configure DUNDi to use SIP or does it only work with IAX?
Thanks,
Ryan
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Re: [asterisk-users] DUNDi with SIP Mapping
From the reading and testing I have done it doesn't look like SIP
supports a username and password in the Dial string. I currently
have the following mapping.
priv = dundi-extens,0,SIP,dundi:pass@1.1.1.1/$
{NUMBER},nounsolicited,nocomunsolicit,nopartial
On the sending side I see
NOTICE[31598] chan_sip.c: Conflicting extension values given. Using
'dundi' and not '1001'
On the receiving side it will not match the SIP dundi user and tries
to call dundi instead of 1001.
-- Executing [dundi@from-sip-external:1] NoOp(SIP/1.1.1.
2-, Received incoming SIP connection from unknown peer to
dundi) in new stack
Is there a way to configure DUNDi to use SIP or does it only work with
IAX?
I am using DUNDi with SIP to do some least cost routing amongst my various
locations. My mapping is close to what you have:
priv = dundi-extens,0,SIP,trunk_name/number_to_dial
Where trunk_name is replaced with the actual name of my trunk as defined
in sip.conf and number_to_dial is the number they should dial on that
trunk. I have not tried to define the SIP username/password in the DUNDi
config itself, so I don't know if what you are trying to do is possible or
not.--
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Re: [asterisk-users] DUNDi with SIP Mapping
On Wed, Apr 16, 2014 at 9:06 AM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: I am using DUNDi with SIP to do some least cost routing amongst my various locations. My mapping is close to what you have: priv = dundi-extens,0,SIP,trunk_name/number_to_dial Where trunk_name is replaced with the actual name of my trunk as defined in sip.conf and number_to_dial is the number they should dial on that trunk. I have not tried to define the SIP username/password in the DUNDi config itself, so I don't know if what you are trying to do is possible or not. I was trying to avoid having to define the SIP trunks on all systems. I currently have three FreePBX systems connected by SIP trunks with 800 DIDs. Each system has SIP trunks defined to both other systems and routes defining the extensions / DIDs. As I add more DID blocks and FreePBX systems maintaining the trunks and routes is going to become cumbersome. I wanted to move to DUNDi to simplify the setup. It looks like I need to switch to IAX trunks to be able to do this. Thanks, Ryan -- _ -- 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] DUNDi with SIP Mapping
I wanted to move to DUNDi to simplify the setup. It looks like I need to switch to IAX trunks to be able to do this. You are a bit outside of what I have done, but this looks like it might be what you want to do with SIP: http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP-- _ -- 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] DUNDI or ENUM or ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 20/01/2014 12:03, Jean-Denis Girard a écrit : Hi list, I'm looking for the best / recommended solution for automatic discovery of phone numbers for a multiple Asterisk system. This would be for an administration, with many branches (~30), but a common infrastructure (DNS, LDAP). Most branches would have Asterisk servers for various reasons (location, administrative). All contacts would be in LDAP, and Asterisk servers would have DNS entries. The problem is contacting other Asterisk without setting static routes in dialplan. I think DUNDI would be ideal, but is it still recommended for new installations or is it deprecated? dundi.com is dead, and redirects to the profile page on Digium website (https://my.digium.com/en/users/viewprofile/). ENUM could be another solution. What would you suggest? No recommendation ? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlLep2QACgkQuu7Rv+oOo/h1YwCgnhs5Pioo0vr5wuWB4yZeDVuJ S+oAnj1GGr7JXtc3wDVyc4wOSN5GZZcw =0O+Z -END PGP SIGNATURE- -- _ -- 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] DUNDI or ENUM or ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, I'm looking for the best / recommended solution for automatic discovery of phone numbers for a multiple Asterisk system. This would be for an administration, with many branches (~30), but a common infrastructure (DNS, LDAP). Most branches would have Asterisk servers for various reasons (location, administrative). All contacts would be in LDAP, and Asterisk servers would have DNS entries. The problem is contacting other Asterisk without setting static routes in dialplan. I think DUNDI would be ideal, but is it still recommended for new installations or is it deprecated? dundi.com is dead, and redirects to the profile page on Digium website (https://my.digium.com/en/users/viewprofile/). ENUM could be another solution. What would you suggest? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlLdnUgACgkQuu7Rv+oOo/jd6QCffhNne0yiNnfrgcS+cRQziz1/ dL0Anipk2Qqj2pCWbLIorW+Z8qff3q4L =DzaL -END PGP SIGNATURE- -- _ -- 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] dundi
Dear is it possible to send ring(call) to all devices with same (sip_username) in all servers ? in this schematics, some bodies have shared lines. so all lines must be in service . Best -- Pezhman Lali -- _ -- 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] dundi
Dundi just give you location of extensions. For ring you should have capable dialplan and peering. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali Sent: Wednesday, August 03, 2011 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] dundi Dear is it possible to send ring(call) to all devices with same (sip_username) in all servers ? in this schematics, some bodies have shared lines. so all lines must be in service . Best -- Pezhman Lali -- _ -- 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] DUNDi and Lua dialplan
Hello,
I would like to known how to use DUNDi with a Lua dialplan ?
In extensions.conf, we should do like these:
|[lookupdundi]
switch = DUNDi/priv
[internal]
include = dundiextens
include = lookupdundi
exten = _,2,NoOp(calling ${EXTEN})
exten = _,n,Dial(SIP/${EXTEN})
exten = _,n,Hangup()|
priority 1 is either defined in dundiextens (local registered devices)
or lookupdundi (remote)
But as in Lua there is no priority, we can't to this.
I found the following method working:
|extensions = {
internal = {
[_] = function(c,e)
app.noop('lua:: dialing exten ' .. e)
-- Goto is not working, I need to use a Local
channel
app.dial('Local/'..e..'@lookupdundi')
app.dial('SIP/'..e)
app.hangup()
end;
};
}|
But is this correct/the best one ?
Regards,
Guillaume
--
Guillaume Bourgb...@proformatique.com - proformatique
10 bis, rue Lucien VOILIN - 92800 Puteaux
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Re: [asterisk-users] DUNDi questions
On 7/30/2010 11:37 AM, unsero...@aol.com wrote: Hi all, I have two questions regarding DUNDi and Asterisk Realtime. I have successfully set up DUNDi on my two Asterisk boxes, which means dundi show peers on each box shows the other box as known and dialplan show dundiextens shows the extensions on each box configured in sip.conf. 1. But when i switch my config to use sip in realtime, my extensions are only visible to DUNDi if i set rtcachefriends in sip.conf to yes. Am I forced to set rtcachefriends to use DUNDi with realtime or do I miss something, maybe an additional column in my database table? 2. How can I use DUNDi within my dialplan to determine if an extension is reachable and then establish a call to it and if not, pass the call to my PSTN device? This sounds like you need to enable regcontext and regexten in sip.conf and for your peers. This will cause a line of dialplan to be added to the regcontext upon registration of your peer, which you can then use as the lookup context for your DUNDi mapping. The dialplan is dynamically created and will add and remove the information assigned to regexten for the peer upon (de)registration. Leif Madsen. -- _ -- 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] DUNDi questions
Hi all, I have two questions regarding DUNDi and Asterisk Realtime. I have successfully set up DUNDi on my two Asterisk boxes, which means dundi show peers on each box shows the other box as known and dialplan show dundiextens shows the extensions on each box configured in sip.conf. 1. But when i switch my config to use sip in realtime, my extensions are only visible to DUNDi if i set rtcachefriends in sip.conf to yes. Am I forced to set rtcachefriends to use DUNDi with realtime or do I miss something, maybe an additional column in my database table? 2. How can I use DUNDi within my dialplan to determine if an extension is reachable and then establish a call to it and if not, pass the call to my PSTN device? This sounds like you need to enable regcontext and regexten in sip.conf and for your peers. This will cause a line of dialplan to be added to the regcontext upon registration of your peer, which you can then use as the lookup context for your DUNDi mapping. The dialplan is dynamically created and will add and remove the information assigned to regexten for the peer upon (de)registration. Leif Madsen. -- I've set regcontext in the general section in sip.conf but not explicitly for the peers. Also I did not set regexten yet, I guess this could be causing my problems. Will research what regexten is used for and try it again. So DUNDi will only advertise extensions that are actually registered? Is that correct? As soon as an extension is unregistered (will say offline) it is not advertised by DUNDi anymore to route a call to e.g. the extensions voicemail? Oh, i thought i could do something like a DUNDILOOKUP inside of my dialplan to see if the extension is known and if not pass it to my PSTN gateway. Thanks so far. Oliver -- _ -- 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] DUNDi questions
Hi all, I have two questions regarding DUNDi and Asterisk Realtime. I have successfully set up DUNDi on my two Asterisk boxes, which means dundi show peers on each box shows the other box as known and dialplan show dundiextens shows the extensions on each box configured in sip.conf. 1. But when i switch my config to use sip in realtime, my extensions are only visible to DUNDi if i set rtcachefriends in sip.conf to yes. Am I forced to set rtcachefriends to use DUNDi with realtime or do I miss something, maybe an additional column in my database table? 2. How can I use DUNDi within my dialplan to determine if an extension is reachable and then establish a call to it and if not, pass the call to my PSTN device? Thanks in advance, Oliver -- _ -- 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] DUNDi Confusion
Dear community,
Please help. I've been looking around the internet (and in this great forum)
for help with DUNDi setup between servers (I'm using Elastix) and while I
can get my servers to lookup extensions on each other very well, I have not
been able to successfully make calls between servers. For my test
environment, I have 3 servers setup for now, and these are the steps I've
followed:
1. I edited dundi.conf on each server to have the following info:
(this listing is for all servers)
[mappings]
priv = ext-dundi,0,IAX2,priv:${SECRET}@
'server-hostname'/${NUMBER},nopartial
[00:1C:C0:65:34:04]
model = symmetric
host = 192.168.1.128
inkey = priv
outkey = priv
include = all
permit = all
qualify = yes
order = primary
dynamic = yes
[08:00:27:57:6E:0E]
model = symmetric
host = elastix-1
inkey = priv
outkey = priv
include = all
permit = all
qualify = yes
[08:00:27:15:0E:F1]
model = symmetric
host = elastix-2
inkey = priv
outkey = priv
include = all
permit = all
qualify = yes
order = primary
dynamic = yes
2. I also edited extensions_custom.conf in each server to have:
[ext-dundi]
include = ext-local
include = ext-paging
include = ext-intercom-users
include = ext-group
include = ext-meetme
3. I also created an IAX2 Trunk called 'priv' using FreePBX (placing
information below only within the PEER Details(this trunk shows up as
'IAX2/priv' in FreePBX/Elastix web configurator):
[priv]
type=friend
dbsecret=dundi/secret
context=from-internal
trunk=yes
4. I also created a DUNDi Trunk called 'priv' as well in FreePBX and edited
only the DUNDI Mapping in there. This too shows up as 'DUNDi/priv' in the
FreePBX/Elastix web configurator.
The next steps to do is what confuses me. My DUNDi lookups and queries work
fine, and I have no firewalls between the boxes.
I have created a route called dundi-outside in each server's FreePBX that
references the DUNDi/priv route, and subsequently deleted it, because
whenever i try to make calls i get either an 'all-circuits-are-busy' error
msg, or i get a
'call-cannot-be-completed-as-dialled-please-check-the-number-and-dial-again'
error.
I'm really confused as what is going wrong. Am I (surely) missing something?
Any help will be greatly appreciated.
Hope to hear from you soon.
--
best regards,
Sina Owolabi
2348034022578
23417203257
23417420690
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[asterisk-users] DUNDI Sip authentication failure
Hi all, I'm new in asterisk and I got to set up a dundi config for my work.
I have 2 PBX for the test, the two PBX are in the same local network
PBX A : 192.168.199.23
PBX B : 192.168.199.21
my config files : (on PBX B , the config files on PBX A looks like it)
/etc/asterisk/dundi.conf
[general]
bind=192.168.199.21
port=4520
cachetime=5
ttl=32
autokill=yes
entityid=00:30:18:4C:33:53
[mappings]
;dundi-test =
dundi-local,0,IAX2,dundi:${secr...@toronto.example.com/${NUMBER},nounsolicited,nocomunsolicit,nopartial
priv =
dundi-priv-canonical,0,SIP,dundi:${secr...@192.168.199.21/${NUMBER},nounsolicited,nocomunsolicit,nopartial
;priv = dundi-priv-canonical,0,SIP,192.168.199.21/${NUMBER},nopartial
priv =
dundi-priv-customers,100,SIP,dundi:${secr...@192.168.199.21/${NUMBER},nounsolicited,nocomunsolicit,nopartial
;priv = dundi-priv-customers,100,SIP,192.168.199.21/${NUMBER},nopartial
priv =
dundi-priv-customers,400,SIP,dundi:${secr...@192.168.199.21/${NUMBER},nounsolicited,nocomunsolicit,nopartial
;priv = dundi-priv-via-pstn,400,SIP,192.168.199.21/${NUMBER},nopartial
[00:40:48:B2:78:6B]
model = symmetric
host = 192.168.199.23
inkey = 192.168.199.23
outkey = 192.168.199.21
include = priv
permit = priv
qualify = yes
order = primary
*/etc/asterisk/sip_custom.conf
language=fr
nat=never
;Subscribecontext=ext-local
[priv]
type=friend
dbsecret=dundi/secret
context=dundi-priv-local
host=192.168.199.23
qualify=yes*
/etc/asterisk/extensions_custom.conf
[ext-local-custom]
;for Direct IVR dialing if IVR is installed on the PBX B
exten = _36X,1,Macro(dundi-priv,${EXTEN})
[dundi-priv-canonical]
; local number of the PBX A for dundi advertise
exten = _37X,1,Goto(ext-local,${EXTEN},1)
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-lookup
[trydundi]
exten = _.,1,Macro(dundi-priv,${EXTEN})
exten = _.,2,Congestion
What works : if I use (on PBX B)
dundi lookup 3...@priv
asterisk respond :
1. 0 SIP/dundi:+wxatxxjxspp8mrpal3mr...@192.168.199.23/360
(EXISTS|NOUNSLCTD|NOCOMUNSLTD)
from 00:40:48:b2:78:6b, expires in 5 s
DUNDi lookup completed in 7 ms
but if I try to call from 360 to 370 or from 370 to 360 the call fails
So it seems that I have a SIP authentication failure.
but I don't know how to find the real problem.
Can you help me ?
Here are some logs :
On the CLI prompt :
-- Executing [...@from-internal:1] ResetCDR(SIP/360-08dfe0a0, )
in new stack
-- Executing [...@from-internal:2] NoCDR(SIP/360-08dfe0a0, ) in
new stack
-- Executing [...@from-internal:3] Wait(SIP/360-08dfe0a0, 1) in
new stack
-- Executing [...@from-internal:4] Playback(SIP/360-08dfe0a0,
silence/1cannot-complete-as-dialedcheck-number-dial-again|noanswer)
in new stack
-- SIP/360-08dfe0a0 Playing 'silence/1' (language 'fr')
-- SIP/360-08dfe0a0 Playing 'cannot-complete-as-dialed' (language
'fr')
-- SIP/360-08dfe0a0 Playing 'check-number-dial-again' (language 'fr')
-- Executing [...@from-internal:5] Wait(SIP/360-08dfe0a0, 1) in
new stack
== Spawn extension (from-internal, 370, 5) exited non-zero on
'SIP/360-08dfe0a0'
-- Executing [...@from-internal:1] Macro(SIP/360-08dfe0a0,
hangupcall) in new stack
-- Executing [...@macro-hangupcall:1] ResetCDR(SIP/360-08dfe0a0, w)
in new stack
-- Executing [...@macro-hangupcall:2] NoCDR(SIP/360-08dfe0a0, ) in
new stack
-- Executing [...@macro-hangupcall:3] GotoIf(SIP/360-08dfe0a0,
1?skiprg) in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [...@macro-hangupcall:6] GotoIf(SIP/360-08dfe0a0,
1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [...@macro-hangupcall:9] GotoIf(SIP/360-08dfe0a0,
1?theend) in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [...@macro-hangupcall:11] Hangup(SIP/360-08dfe0a0, )
in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/360-08dfe0a0' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/360-08dfe0a0'
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: NULL (Command)
Flags: 00 STrans: 29219 DTrans: 0 [192.168.199.21:4520] (Final)
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 08363 DTrans: 29219 [192.168.199.21:4520] (Final)
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: NULL (Command)
Flags: 00 STrans: 12520 DTrans: 0 [192.168.199.21:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 09513
Re: [asterisk-users] DUNDi + SIP Realtime
JR - couldn't find your whitepaper from astricon06 online, links are
broken would it be possible for you to email it to me?
I have not tried setting up DUNDi yet, but from the sound of it, seems
like it would be pretty handy.
I have sip phones registering to two asterisk servers [primary and
backup] and have a macro setup where incoming call checks if sip phone
is available via ${SIPPEER}(${EXTEN}:status and if it is not on the
primary server, then sends the call to the backup server in a separate
context which attempts to dial the sip phone once, and if the phone is
offline / unavailable there as well, then to send it to voicemail or
followme as the case may be.
However, DUNDi sounds better in terms of scalability as I'm fast
outgrowing two servers.
:)
Thanks,
Neeraj
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Re: [asterisk-users] DUNDi + SIP Realtime
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP extensions
to the others via DUNDi. It works like a charm. SIP extensions can only
register at one server, the one they belong to. In case one extension
wants to call other that is registered in another server, DUNDi takes care
of that by calling the other server using IAX2 and G.729 codec. It's
important to clarify that each server today works as a completely
independent PBX, talking to each other using IAX2 that is routed via a
MPLS network. Also, the servers are very distant from each other
physically.
Today the extensions are being mapped in DUNDi like this:
[dundi-internal]
exten = _70XX,1,Noop()
And all extensions configurations are done in sip.conf. No realtime is
being used, yet.
Now the customer wants to take a step further and make it possible that
*every* SIP extension could be able to register in *every* server. That
would make possible for them to use DNS to automatically find the
closest PBX and make the extension register on that one.
So far I considered the following for this project:
- Moving all SIP extensions from individual sip.confs to one MySQL
database, and point all servers to that one
- Configure sip.conf on each machine like this:
regcontext=dundi-internal
rtcachefriends=yes
rtsavesysname=yes
rtupdate=no
rtautoclear=yes
ignoreregexpire=no
That way each time an extension registers, Asterisk would add an extension
to the dundi-internal context, which as you guessed, is the one being
mapped to the other servers. So instead of mapping extensions using
wildcards, the extensions will be mapped individually.
extensions.conf would be something like this:
[internal]
;Tries to make the call using SIP, in the case
;the extension is registered in this server
;If it's not, switches to DUNDi
exten = _,1,Dial(SIP/${EXTEN},60)
exten = _,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten = _,n,NoOp(FROM_DUNDI = ${FROM_DUNDI})
exten = _,n,GotoIf($[${FROM_DUNDI} = 1]?end:start)
exten = _,n(start),Answer()
exten = _,n,Playback(vm-dialout)
exten = _,n,Goto(dundi-internal-helper,${EXTEN},1)
exten = _,n(end),Noop(Loop detected. Hanging up.)
exten = _,n,Hangup()
[dundi-internal-helper]
switch = DUNDi/dundi_internal
[from-dundi]
exten = _,1,Set(FROM_DUNDI=1)
exten = _,n,Dial(SIP/${EXTEN},60)
So far it's working fine in a test lab with 2 servers running Asterisk
1.6.0.15.
For the gurus out there: is there something that I'm doing terribly wrong,
that would break everything and make the universe collapse into itself
when I apply the same principle on production?
I'll be happy to provide more details in case there are any doubts. I
really appreciate your feedback, no matter what is it. :)
Vin?cius Fontes
www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e
telefonia IP
[JR Richardson]
I used to do this exact thing a few years ago, wrote a couple of papers
about it. Realtime + DINDi works great for this, I would add in MySQL
replication to the mix so each server writes the SIP cache info to a Master
database that is replicated out to all the servers. Each server will have a
copy of the same database and be able to contact the phones if DUNDi queries
become unavailable.
The tricky problem you may run into, if you haven't figured it out yet, is
what to do about voicemail and where the storage will be, distributed
voicemail will be problematic in a dynamic sip ua registration environment
across multiple servers. Centralize voicemail using DUNDi can help this out
as well.
I'll send you some papers off line
Hope this helps.
JR
Engineering for the Masses
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Re: [asterisk-users] DUNDi + SIP Realtime
- JR Richardson jmr.richard...@gmail.com escreveu:
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP
extensions
to the others via DUNDi. It works like a charm. SIP extensions can
only
register at one server, the one they belong to. In case one
extension
wants to call other that is registered in another server, DUNDi
takes care
of that by calling the other server using IAX2 and G.729 codec.
It's
important to clarify that each server today works as a completely
independent PBX, talking to each other using IAX2 that is routed via
a
MPLS network. Also, the servers are very distant from each other
physically.
Today the extensions are being mapped in DUNDi like this:
[dundi-internal]
exten = _70XX,1,Noop()
And all extensions configurations are done in sip.conf. No realtime
is
being used, yet.
Now the customer wants to take a step further and make it possible
that
*every* SIP extension could be able to register in *every* server.
That
would make possible for them to use DNS to automatically find the
closest PBX and make the extension register on that one.
So far I considered the following for this project:
- Moving all SIP extensions from individual sip.confs to one MySQL
database, and point all servers to that one
- Configure sip.conf on each machine like this:
regcontext=dundi-internal
rtcachefriends=yes
rtsavesysname=yes
rtupdate=no
rtautoclear=yes
ignoreregexpire=no
That way each time an extension registers, Asterisk would add an
extension
to the dundi-internal context, which as you guessed, is the one
being
mapped to the other servers. So instead of mapping extensions using
wildcards, the extensions will be mapped individually.
extensions.conf would be something like this:
[internal]
;Tries to make the call using SIP, in the case
;the extension is registered in this server
;If it's not, switches to DUNDi
exten = _,1,Dial(SIP/${EXTEN},60)
exten = _,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten = _,n,NoOp(FROM_DUNDI = ${FROM_DUNDI})
exten = _,n,GotoIf($[${FROM_DUNDI} = 1]?end:start)
exten = _,n(start),Answer()
exten = _,n,Playback(vm-dialout)
exten = _,n,Goto(dundi-internal-helper,${EXTEN},1)
exten = _,n(end),Noop(Loop detected. Hanging up.)
exten = _,n,Hangup()
[dundi-internal-helper]
switch = DUNDi/dundi_internal
[from-dundi]
exten = _,1,Set(FROM_DUNDI=1)
exten = _,n,Dial(SIP/${EXTEN},60)
So far it's working fine in a test lab with 2 servers running
Asterisk
1.6.0.15.
For the gurus out there: is there something that I'm doing terribly
wrong,
that would break everything and make the universe collapse into
itself
when I apply the same principle on production?
I'll be happy to provide more details in case there are any doubts.
I
really appreciate your feedback, no matter what is it. :)
Vin?cius Fontes
www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e
telefonia IP
[JR Richardson]
I used to do this exact thing a few years ago, wrote a couple of
papers
about it. Realtime + DINDi works great for this, I would add in
MySQL
replication to the mix so each server writes the SIP cache info to a
Master
database that is replicated out to all the servers. Each server will
have a
copy of the same database and be able to contact the phones if DUNDi
queries
become unavailable.
The tricky problem you may run into, if you haven't figured it out
yet, is
what to do about voicemail and where the storage will be, distributed
voicemail will be problematic in a dynamic sip ua registration
environment
across multiple servers. Centralize voicemail using DUNDi can help
this out
as well.
I'll send you some papers off line
Hope this helps.
JR
Engineering for the Masses
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Thank you so much for your answers. I really wasn't aware of the problem with
voicemail, but storing it on IMAP or even MySQL will certainly help.
Your case is really close to my own, with a few differences. The main objective
in my case is not redundancy, but saving bandwidth. That because if a user of
pbxA is physically in the same network of pbxB, it will register directly to
pbxA. And as the dialplan uses the options tT on Dial(), if this user calls an
extension registered at pbxB, the audio needs to go to pbxA and come back to
pbxB. To make things worst, in this case the codec used is alaw.
So my idea is to force all inter-server communication to be done
[asterisk-users] DUNDi + SIP Realtime
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP extensions to
the others via DUNDi. It works like a charm. SIP extensions can only register
at one server, the one they belong to. In case one extension wants to call
other that is registered in another server, DUNDi takes care of that by calling
the other server using IAX2 and G.729 codec. It's important to clarify that
each server today works as a completely independent PBX, talking to each other
using IAX2 that is routed via a MPLS network. Also, the servers are very
distant from each other physically.
Today the extensions are being mapped in DUNDi like this:
[dundi-internal]
exten = _70XX,1,Noop()
And all extensions configurations are done in sip.conf. No realtime is being
used, yet.
Now the customer wants to take a step further and make it possible that *every*
SIP extension could be able to register in *every* server. That would make
possible for them to use DNS to automatically find the closest PBX and make
the extension register on that one.
So far I considered the following for this project:
- Moving all SIP extensions from individual sip.confs to one MySQL database,
and point all servers to that one
- Configure sip.conf on each machine like this:
regcontext=dundi-internal
rtcachefriends=yes
rtsavesysname=yes
rtupdate=no
rtautoclear=yes
ignoreregexpire=no
That way each time an extension registers, Asterisk would add an extension to
the dundi-internal context, which as you guessed, is the one being mapped to
the other servers. So instead of mapping extensions using wildcards, the
extensions will be mapped individually.
extensions.conf would be something like this:
[internal]
;Tries to make the call using SIP, in the case
;the extension is registered in this server
;If it's not, switches to DUNDi
exten = _,1,Dial(SIP/${EXTEN},60)
exten = _,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten = _,n,NoOp(FROM_DUNDI = ${FROM_DUNDI})
exten = _,n,GotoIf($[${FROM_DUNDI} = 1]?end:start)
exten = _,n(start),Answer()
exten = _,n,Playback(vm-dialout)
exten = _,n,Goto(dundi-internal-helper,${EXTEN},1)
exten = _,n(end),Noop(Loop detected. Hanging up.)
exten = _,n,Hangup()
[dundi-internal-helper]
switch = DUNDi/dundi_internal
[from-dundi]
exten = _,1,Set(FROM_DUNDI=1)
exten = _,n,Dial(SIP/${EXTEN},60)
So far it's working fine in a test lab with 2 servers running Asterisk 1.6.0.15.
For the gurus out there: is there something that I'm doing terribly wrong, that
would break everything and make the universe collapse into itself when I apply
the same principle on production?
I'll be happy to provide more details in case there are any doubts. I really
appreciate your feedback, no matter what is it. :)
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
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Re: [asterisk-users] DUNDi Errors (ENCREJ)
On Tuesday 30 June 2009 05:01:42 am srinivas Antarvedi wrote: - To resolve this i tried to remove all keys in all servers and once again created and distributed the loaded in each system with keys init command but stilll i am getting the same error can anybody help me out??? Thanks and regards srinivas antarvedi try module reload res_crypto.so or restart your asterisk servers. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi Errors (ENCREJ)
Hello users.
i am planning to implement the dundi protocol among 3 servers
where the real channels residing in 2 servers and the remaining one
is only for routing purpose..
here is how my config files
#Routing_server
routing server -192.168.1.11
node1-192.168.1.21
node2-192.168.1.31
i)dundi.conf
dundi=dundicontext,0,IAX2,priv:${secr...@192.168.1.11/${NUMBER},nopartial
[MACaddress of node1]
model=symmetric
host = 192.168.1.21
inkey = priv
outkey = priv
include = priv
permit = priv
qualify = yes
order=primary
;[MAC oF system node2];
;model=symmetric
;host = 192.168.1.31
;inkey = priv
;outkey = priv
;include = priv
;permit = priv
;qualify = yes
;order=secondary
2)extension.conf
[dundicontext]
include = lookupdundi
[lookupdundi]
switch = DUNDi/dundi
3)iax.conf
[priv]
dbsecret=dundi/secret
type=friend
context=dundicontext
- when i tested the dundi show peers in my server the 2 nodes
information i was able to see
- when i used dundi lookup 2...@dundi
i am getting this error
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ENCREJ (Response)
Flags: 00 STrans: 16791 DTrans: 30106 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 30106 DTrans: 16791 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: ENCRYPT (Command)
Flags: 00 STrans: 26692 DTrans: 0 [192.168.1.11:4520]
ENTITY IDENT: 00:23:7d:93:f7:5e
KEYCRC32: 4234245369
ENCDATA : [IV 11dc622321b7c86ae1e7016c4fe4101d] 4 encrypted blocks
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ENCREJ (Response)
Flags: 00 STrans: 22476 DTrans: 26692 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 26692 DTrans: 22476 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: ENCRYPT (Command)
Flags: 00 STrans: 00299 DTrans: 0 [192.168.1.11:4520]
ENTITY IDENT: 00:23:7d:93:f7:5e
SHAREDKEY : [ 66 df 0a c5 75 59 2f 75 bc 48 bd c8 39 6c 33 df
73 37 85 10 86 ed b5 da 4c 88 a7 c5 00 f0 ab d0 2a 9b b3 71 86 7a c4
53 dd dc 4f 29 fb 43 b4 17 d9 91 e1 df 5b 6f 6d c2 b0 f7 d2 f9 f0 b8
3b 0c 0e 7d af ef 8e 4f cf 9f 7e ca 50 b2 04 97 60 2b cb df fd 97 82
d4 bf a0 cf 9a 66 60 11 19 bc 6b 63 30 a5 05 2f 9e a7 63 1d 90 f6 ac
13 23 39 30 33 1d 29 7a 0 6 da 52 5d b0 d7 e7 3f e7 ef 2d a1 ]
SIGNATURE : [ 65 da f2 7a 0f e1 ea 40 73 56 bc 78 d0 05 c0 c3
ec 4c 97 53 cc 2f 2f 97 01 1a 0d ee 8f 21 8f 5a c2 65 91 6d 32 16 dc
27 75 f6 12 9f 3e f3 bd 34 29 9e c9 af 8d 03 ef 43 7c f4 4d 48 e6 cc
70 af 86 89 ef 24 78 3e c3 71 be cb 55 2c e3 79 19 61 2b 34 d4 8f 62
f6 99 8d 27 9f af 56 a3 8b 30 c6 a1 42 de e5 92 4b f0 8f a2 90 91 86
27 fd 0f 7f 1d b6 4a f7 7 2 53 95 d9 d2 14 03 c6 fd b9 9e 5a ]
ENCDATA : [IV 11dc622321b7c86ae1e7016c4fe4101d] 4 encrypted blocks
- To resolve this i tried to remove all keys in all servers and once
again created and
distributed the loaded in each system with keys init command but
stilll i am
getting the same error
can anybody help me out???
Thanks and regards
srinivas antarvedi
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[asterisk-users] dundi show peers - UNREACHABLE but I can ping it!
Hi guys!! This is something that have always bother me, hope you can help me... :) I've 8 server connected using IAX / DUNDi, it works just fine. However, sometimes when some of our links goes down the server takes forever to appear back as OK at DUNDi's list and people can't call the other Box. It's happening right now: CLI dundi show peers 00:14:22:16:54:c5200.X.X.6(S) Symmetric Unavail UNREACHABLE # ping 200.X.X.6 PING 200.X.x.6 (200.X.X.6) 56(84) bytes of data. 64 bytes from 200.X.X.6: icmp_seq=1 ttl=52 time=13.8 ms 64 bytes from 200.X.X.6: icmp_seq=2 ttl=52 time=17.7 ms --- 200.X.X.6 ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 1001ms rtt min/avg/max/mdev = 13.874/15.813/17.752/1.939 ms I've tried to do module reload pbx_dundi.so, sometimes it seems like it works... But doesn't seems right... There is a right way to force DUNDi to re-check the peers? How does it check the peers? BTW, my DUNDi configuration is based on that guide DUNDi so easy a caveman could do it... none of my Asterisk is a trixbox, elastix, etc... Well, thats it.. Thanks in advance!! Best regards, -- Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi show peers - UNREACHABLE but I can ping it!
Sorry if this is a top-post, I'm using MS Outlook. Does DSP work when all is well? What about dundi flush or dundi show trans? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Durante Sent: Tuesday, March 31, 2009 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] dundi show peers - UNREACHABLE but I can ping it! Hi guys!! This is something that have always bother me, hope you can help me... :) I've 8 server connected using IAX / DUNDi, it works just fine. However, sometimes when some of our links goes down the server takes forever to appear back as OK at DUNDi's list and people can't call the other Box. It's happening right now: CLI dundi show peers 00:14:22:16:54:c5200.X.X.6(S) Symmetric Unavail UNREACHABLE # ping 200.X.X.6 PING 200.X.x.6 (200.X.X.6) 56(84) bytes of data. 64 bytes from 200.X.X.6: icmp_seq=1 ttl=52 time=13.8 ms 64 bytes from 200.X.X.6: icmp_seq=2 ttl=52 time=17.7 ms --- 200.X.X.6 ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 1001ms rtt min/avg/max/mdev = 13.874/15.813/17.752/1.939 ms I've tried to do module reload pbx_dundi.so, sometimes it seems like it works... But doesn't seems right... There is a right way to force DUNDi to re-check the peers? How does it check the peers? BTW, my DUNDi configuration is based on that guide DUNDi so easy a caveman could do it... none of my Asterisk is a trixbox, elastix, etc... Well, thats it.. Thanks in advance!! Best regards, -- Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi show peers - UNREACHABLE but I can ping it!
Hi Danny, On Tue, Mar 31, 2009 at 3:32 PM, Danny Nicholas da...@debsinc.com wrote: Sorry if this is a top-post, I'm using MS Outlook. Does DSP work when all is well? What about dundi flush or dundi show trans? Yes, when everything is OK all the calls goes just fine! Perfectly actually... I tried 'dundi flush', well it said cache flushed but didn't change anything... 'dundi show trans' showed: CLI dundi show trans Remote Src Dst Tx Rx Ack 200.X.X.6: 4520 11066 0 001 000 000 190.X.X.34 : 4520 29803 0 001 000 000 200.X.X.6: 4520 06203 11498 000 000 000 200.X.X.6: 4520 31969 11498 000 000 000 Well... how do I read this?! lol... I've google a couple of times for DUNDi documentation, but I never could find anything really good about it... If anyone has a link to share... :) Any ideas? Thank you ! Best regards, -- Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi show peers - UNREACHABLE but I can ping it!
Hi , On Tue, Mar 31, 2009 at 4:41 PM, Tiago Durante tiagodura...@gmail.com wrote: Hi Danny, On Tue, Mar 31, 2009 at 3:32 PM, Danny Nicholas da...@debsinc.com wrote: Sorry if this is a top-post, I'm using MS Outlook. Does DSP work when all is well? What about dundi flush or dundi show trans? Yes, when everything is OK all the calls goes just fine! Perfectly actually... I tried 'dundi flush', well it said cache flushed but didn't change anything... 'dundi show trans' showed: CLI dundi show trans Remote Src Dst Tx Rx Ack 200.X.X.6 : 4520 11066 0 001 000 000 190.X.X.34 : 4520 29803 0 001 000 000 200.X.X.6 : 4520 06203 11498 000 000 000 200.X.X.6 : 4520 31969 11498 000 000 000 Well... how do I read this?! lol... I've google a couple of times for DUNDi documentation, but I never could find anything really good about it... If anyone has a link to share... :) Any ideas? Now I saw there was no channels UP and I could restart the Asterisk. I did 'dundi show trans' and it has nothing... My peers can 'see' each other now... But how do I do it without restarting the service.. And there is anyway to force the Asterisk to do it, lets say, once per day? thank you !!! regards, -- Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi broken in asterisk 1.4-svn?
Andreas Anderson wrote: Hi Guys, since about two weeks pbx_dundi.so from svn segfaults when i load it, 1.4.24 release works fine on the same box. Can someone tell me if that's something weird with my Fedora8 system or a possible bug in svn? Program terminated with signal 11, Segmentation fault. #0 0x in ?? () (gdb) bt #0 0x in ?? () #1 0x0881d00c in dundi_encrypt (trans=0x985e6e0, pack=0x985b5e0) at pbx_dundi.c:1298 #2 0x0881d607 in dundi_send (trans=0x985e6e0, cmdresp=10, flags=value optimized out, final=0, ied=0x1924210) at pbx_dundi.c:3081 #3 0x08824834 in do_register (data=0x985bf60) at pbx_dundi.c:4093 #4 0x080f0e31 in ast_sched_runq (con=0x9855780) at sched.c:363 #5 0x0881c89a in network_thread (ignore=0x0) at pbx_dundi.c:2164 #6 0x080fd7ab in dummy_start (data=0x9840ce0) at utils.c:856 #7 0x0013b50b in start_thread () from /lib/libpthread.so.0 #8 0x00263b2e in clone () from /lib/libc.so.6 Can you open a bug on http://bugs.digium.com with the steps of how to reproduce, along with enabling DONT_OPTIMIZE in the Compiler Flags section of menuselect? After that, then do a 'make install' so that Asterisk is recompiled with DONT_OPTIMIZE in order to generate a useful backtrace. Check out the backtrace.txt file in the doc/ subdirectory of your Asterisk source for more information. -- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi broken in asterisk 1.4-svn?
Hi Guys, since about two weeks pbx_dundi.so from svn segfaults when i load it, 1.4.24 release works fine on the same box. Can someone tell me if that's something weird with my Fedora8 system or a possible bug in svn? Program terminated with signal 11, Segmentation fault. #0 0x in ?? () (gdb) bt #0 0x in ?? () #1 0x0881d00c in dundi_encrypt (trans=0x985e6e0, pack=0x985b5e0) at pbx_dundi.c:1298 #2 0x0881d607 in dundi_send (trans=0x985e6e0, cmdresp=10, flags=value optimized out, final=0, ied=0x1924210) at pbx_dundi.c:3081 #3 0x08824834 in do_register (data=0x985bf60) at pbx_dundi.c:4093 #4 0x080f0e31 in ast_sched_runq (con=0x9855780) at sched.c:363 #5 0x0881c89a in network_thread (ignore=0x0) at pbx_dundi.c:2164 #6 0x080fd7ab in dummy_start (data=0x9840ce0) at utils.c:856 #7 0x0013b50b in start_thread () from /lib/libpthread.so.0 #8 0x00263b2e in clone () from /lib/libc.so.6 Regards Andreas _ Find a way to cure that travel bug with MSN NZ Travel http://travel.msn.co.nz/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dundi negative caching
Hi! Is it possible to configure a negative TTL (number was not found in Dundi) for DUNDI? regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dundi Issues
I'm getting the following error over and over on the console: pbx_dundi.c:2975 dundi_rexmit: Max retries exceeded to host Any idea how to troubleshoot this? My network latency is roughly 40-50ms between all hosts in my dundi cloud. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dundi Issues
I don't know if it's related, but when doing a packet sniff with wireshark, I see UDP checksum incorrect messages: 0.058230 source - destination UDP Source port: 4520 Destination port: 4520 [UDP CHECKSUM INCORRECT] The reply packet however does not have this warning: 9.199240 destination - source UDP Source port: 4520 Destination port: 4520 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Wednesday, November 05, 2008 8:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Dundi Issues I'm getting the following error over and over on the console: pbx_dundi.c:2975 dundi_rexmit: Max retries exceeded to host Any idea how to troubleshoot this? My network latency is roughly 40-50ms between all hosts in my dundi cloud. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dundi Issues
Jeremy Mann wrote: I don’t know if it’s related, but when doing a packet sniff with wireshark, I see UDP checksum incorrect messages: 0.058230 source - destination UDP Source port: 4520 Destination port: 4520 [UDP CHECKSUM INCORRECT] Be careful with this error, some network cards that can do IP Offload processing will show up with bad checksums in Wireshark. Check the specs for your NIC, this may be a Red Herring (or it might not! ). regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dundi Issues
I'm not aware of any offloading done on this particular box, it's an HP ML110 G5 using the onboard NIC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, November 05, 2008 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dundi Issues Jeremy Mann wrote: I don't know if it's related, but when doing a packet sniff with wireshark, I see UDP checksum incorrect messages: 0.058230 source - destination UDP Source port: 4520 Destination port: 4520 [UDP CHECKSUM INCORRECT] Be careful with this error, some network cards that can do IP Offload processing will show up with bad checksums in Wireshark. Check the specs for your NIC, this may be a Red Herring (or it might not! ). regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dundi Issues
Jeremy Mann wrote: I'm not aware of any offloading done on this particular box, it's an HP ML110 G5 using the onboard NIC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, November 05, 2008 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dundi Issues Jeremy Mann wrote: I don't know if it's related, but when doing a packet sniff with wireshark, I see UDP checksum incorrect messages: 0.058230 source - destination UDP Source port: 4520 Destination port: 4520 [UDP CHECKSUM INCORRECT] Be careful with this error, some network cards that can do IP Offload processing will show up with bad checksums in Wireshark. Check the specs for your NIC, this may be a Red Herring (or it might not! ). The HP docs indicate a setting for enable/disable checksum offload, try looking at the packets on-the-wire rather than on the server itself. http://h2.www2.hp.com/bc/docs/support/SupportManual/c00846707/c00846707.pdf regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dundi and regcontext
hi, when a user register on my asterisk i can see it adding Noop for that extension, but after awhile i won't see it anymore: what are the reasons for it being removed on the dynamic context? one thing i found when i unregister it's removed. dialplan show myregcontext [ Context 'myregcontext' created by 'SIP' ] '100500' = 1. Noop(100500) [SIP] '112802' = 1. Noop(112802) [SIP] -= 2 extensions (2 priorities) in 1 context. =- [ Context 'pfingobizsip' created by 'SIP' ] -= 0 extensions (0 priorities) in 1 context. =- my prob is when it's removed dundi cant find it anymore so a user calling from server 1 cannot call user that is in server 2. i've set re-registration to very low (1 minute) to monitor if my phone re-register and to see if it will be added again on the regcontext. but i don't even see it unregistering after 1 minute i only unregistering when i am using x-lite and closing x-lite, i dont see x-lite re-registering if i just leave the softphone open. any idea? regards, ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi and regcontext
According to Your description this is a phone problem. Asterisk behaves as its expected. post your dundi.conf to dig more in to this. regards rama On Wed, Sep 24, 2008 at 9:52 PM, ronald ramos [EMAIL PROTECTED]wrote: hi, when a user register on my asterisk i can see it adding Noop for that extension, but after awhile i won't see it anymore: what are the reasons for it being removed on the dynamic context? one thing i found when i unregister it's removed. dialplan show myregcontext [ Context 'myregcontext' created by 'SIP' ] '100500' = 1. Noop(100500) [SIP] '112802' = 1. Noop(112802) [SIP] -= 2 extensions (2 priorities) in 1 context. =- [ Context 'pfingobizsip' created by 'SIP' ] -= 0 extensions (0 priorities) in 1 context. =- my prob is when it's removed dundi cant find it anymore so a user calling from server 1 cannot call user that is in server 2. i've set re-registration to very low (1 minute) to monitor if my phone re-register and to see if it will be added again on the regcontext. but i don't even see it unregistering after 1 minute i only unregistering when i am using x-lite and closing x-lite, i dont see x-lite re-registering if i just leave the softphone open. any idea? regards, ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dundi and zap devices
Hi, is it possible to use zap devices (es: old analog phones) with dundi? It seems that a sort of zapregistration like sipregistration and iaxregistrations to include in the extensions.conf is missing... If yeshow? Thank you. Giorgio. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dundi Help
Hi tecnocrat,
I 'm trying to setup a Dundi system like yours (one lookup server and 2
pbx servers). I searched on internet for some docs but found a lot of
stuff explaining only a part of the problem and no good example at all
(there's a Richardson doc in internet which can help to start).
I tried to do it myself so I generated the two keys (pri and pub) for
each server with their own hostname then I copied:
- .121 keys to the other two servers (.137 and .204)
- .137 keys to .121
- .204 keys to .121
Let me know how if it works.
Giorgio Incantalupo
technocrat voip wrote:
Hello All,
Iam trying to achive a simple load balancing with dundi.
Here i have three asterisk boxes like below.
*.*.*.121 which is the dundi server
*.*.*.137 A Peer which has the 1000 phone registerd to it
*.*.*.204 B Peer which has the 200 phone registered to it.
The expected behavior of my setup is once i dial from 1000 phone it
has to goto B peer using the .121 dundi server.
Iam getting confused with the public key / private key stuff here.
Iam using the astgenkey -n command to generate them .
Can any body help me by explaining , what keys i have to generate on
each server and which keys i need to copy to which server .
regards
All the conf file stuff is is like below
*.*.*.137
iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi
dundi.conf
[mappings]
priv =
sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:0F:E2:76:4B:33]
model = symmetric
host = *.*.*.21
inkey = dundincomingseven
outkey = dundiseven
include = priv
permit = priv
qualify = yes
order = primary
*.*.*.204
iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi
dundi.conf
[mappings]
priv =
sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
;
[00:0F:E2:76:4B:33]
model = symmetric
host = *.*.*.21
inkey = dundiincomingfour
outkey = dundifour
include = priv
permit = priv
qualify = yes
order = primary
*.*.*.21
iax.conf
[priv]
type=user
context=local-custom
disallow=all
allow=ulaw
allow=alaw
allow=gsm
dundi.conf
[mappings]
priv = dundi-priv-localcustom,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER}
[00:11:0A:34:29:57]
model = symmetric
host = *.*.*.137
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
[00:11:0A:34:29:43]
model = symmetric
host = *.*.*.204
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
.
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FGA srl - http://www.fgasoftware.com -
[EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172
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Re: [asterisk-users] Dundi Help
hi gior,
If i understand correctly your setup would be like below.
A is the dundi serer
B is one pbx
C is one pbx
B and C dundi.conf contain the entity detials of A.
Either for C or B we can place calls to the extensions registered on the
other server.
When C extension make call to B extension call go through A and reach the B.
If you can provide the details of dundi.conf and the extension.conf it would
be very help full.
to dig into the issue.
could you able to do the dundi lookup ..?
regards
On Wed, Sep 10, 2008 at 3:36 PM, Giorgio Incantalupo
[EMAIL PROTECTED] wrote:
Hi tecnocrat,
I 'm trying to setup a Dundi system like yours (one lookup server and 2
pbx servers). I searched on internet for some docs but found a lot of
stuff explaining only a part of the problem and no good example at all
(there's a Richardson doc in internet which can help to start).
I tried to do it myself so I generated the two keys (pri and pub) for
each server with their own hostname then I copied:
- .121 keys to the other two servers (.137 and .204)
- .137 keys to .121
- .204 keys to .121
Let me know how if it works.
Giorgio Incantalupo
technocrat voip wrote:
Hello All,
Iam trying to achive a simple load balancing with dundi.
Here i have three asterisk boxes like below.
*.*.*.121 which is the dundi server
*.*.*.137 A Peer which has the 1000 phone registerd to it
*.*.*.204 B Peer which has the 200 phone registered to it.
The expected behavior of my setup is once i dial from 1000 phone it
has to goto B peer using the .121 dundi server.
Iam getting confused with the public key / private key stuff here.
Iam using the astgenkey -n command to generate them .
Can any body help me by explaining , what keys i have to generate on
each server and which keys i need to copy to which server .
regards
All the conf file stuff is is like below
*.*.*.137
iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi
dundi.conf
[mappings]
priv =
sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:0F:E2:76:4B:33]
model = symmetric
host = *.*.*.21
inkey = dundincomingseven
outkey = dundiseven
include = priv
permit = priv
qualify = yes
order = primary
*.*.*.204
iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi
dundi.conf
[mappings]
priv =
sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
;
[00:0F:E2:76:4B:33]
model = symmetric
host = *.*.*.21
inkey = dundiincomingfour
outkey = dundifour
include = priv
permit = priv
qualify = yes
order = primary
*.*.*.21
iax.conf
[priv]
type=user
context=local-custom
disallow=all
allow=ulaw
allow=alaw
allow=gsm
dundi.conf
[mappings]
priv = dundi-priv-localcustom,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER}
[00:11:0A:34:29:57]
model = symmetric
host = *.*.*.137
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
[00:11:0A:34:29:43]
model = symmetric
host = *.*.*.204
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
.
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FGA srl - http://www.fgasoftware.com -
[EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172
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Re: [asterisk-users] Dundi Help
technocrat voip wrote:
hi gior,
If i understand correctly your setup would be like below.
A is the dundi serer
B is one pbx
C is one pbx
yes
B and C dundi.conf contain the entity detials of A.
yes
Either for C or B we can place calls to the extensions registered on
the other server.
tried only from B to C just for test. Sure it will work from C to B too.
When C extension make call to B extension call go through A and reach
the B.
In theory it should, but the A console shows nothing when calling...do
not know why...
If you can provide the details of dundi.conf and the extension.conf it
would be very help full.
to dig into the issue.
Sure but it is a lot of stuff (I post it below...it is a bit long ::)
Hope may help.)
could you able to do the dundi lookup ..?
The dundi lookup works fine (not locally, only from one machine to
others according to wiki), but I've got problems to understand some part
of the files (es: default and peer sections).
regards
::)
---
- Server DundiLookup -
---
DUNDI.CONF
[general]
department=dundio
organization=dundio
locality=Rho
stateprov=MI
country=IT
[EMAIL PROTECTED]
phone=0123456789
entityid=00:30:84:7A:5B:FA
cachetime=5
ttl=2
autokill=yes
[mappings]
dundi-priv = sipregistration,0,SIP,[EMAIL PROTECTED],nopartial
;per chiamare via sip
[00:30:84:7A:55:9F]
model = symmetric
host = 192.168.0.51
inkey = 51
outkey = 71
include = dundi-priv
permit = dundi-priv
qualify = yes
dynamic= yes
order = primary
[00:30:84:41:0C:1C]
model = symmetric
host = 192.168.0.55
inkey = 55
outkey = 71
include = dundi-priv
permit = dundi-priv
qualify = yes
dynamic= yes
order = primary
EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[default]
exten = _XXX,1,Macro(dundi-priv,${EXTEN})
exten = _XXX,2,Playback(invalid)
exten = _XXX,3,Hangup
[iax-clt]
include = sipregistration ;include i sip dell'extensions locale
exten = _XXX,2,Answer
exten = _XXX,3,Dial(SIP/${EXTEN})
exten = _XXX,4,Hangup
[dundi-priv-local]
include = iax-clt
[dundi-priv-switch]
switch = dundi/dundi-priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten= s,1,goto(${ARG1},1)
include = dundi-priv-lookup
SIP.CONF
[general]
regcontext=sipregistration
[peer]
username=peer
context=default
context=default; Default context for incoming calls
bindport=5060; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes; Enable DNS SRV lookups on outbound calls
---
- PBX server -
---
DUNDI.CONF
[general]
department=test
blah blah
entityid=00:30:84:7A:55:9F
cachetime=5
ttl=2
autokill=yes
[mappings]
dundi-priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
;call via sip
[00:30:84:7A:5B:FA]
model = symmetric
host = 192.168.0.71
inkey = 71
outkey = 51
include = dundi-priv
permit = dundi-priv
qualify = yes
dynamic= yes
order = primary
EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
autofallthrough=no
[default]
exten = _XXX,1,Macro(dundi-priv,${EXTEN})
exten = _XXX,2,Playback(invalid)
exten = _XXX,3,Hangup
[iax-clt]
include = sipregistration ;including local extensions
exten = _XXX,2,Answer
exten = _XXX,3,Dial(SIP/${EXTEN})
exten = _XXX,4,Hangup
[dundi-priv-local]
include = iax-clt
[dundi-priv-switch]
switch = dundi/dundi-priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten= s,1,goto(${ARG1},1)
include = dundi-priv-lookup
[general]
regcontext=sipregistration ;mandatory
[peer] ; mandatory
username=peer
context=default
[100]
type = friend
secret=blah
context=default
host=dynamic
qualify = no
username=100
fromuser=100
dtmfmode = rfc2833
language = it
SIP.CONF
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[asterisk-users] Dundi Help
Hello All,
Iam trying to achive a simple load balancing with dundi.
Here i have three asterisk boxes like below.
*.*.*.121 which is the dundi server
*.*.*.137 A Peer which has the 1000 phone registerd to it
*.*.*.204 B Peer which has the 200 phone registered to it.
The expected behavior of my setup is once i dial from 1000 phone it has to
goto B peer using the .121 dundi server.
Iam getting confused with the public key / private key stuff here.
Iam using the astgenkey -n command to generate them .
Can any body help me by explaining , what keys i have to generate on each
server and which keys i need to copy to which server .
regards
All the conf file stuff is is like below
*.*.*.137
iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi
dundi.conf
[mappings]
priv = sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:0F:E2:76:4B:33]
model = symmetric
host = *.*.*.21
inkey = dundincomingseven
outkey = dundiseven
include = priv
permit = priv
qualify = yes
order = primary
*.*.*.204
iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi
dundi.conf
[mappings]
priv = sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
;
[00:0F:E2:76:4B:33]
model = symmetric
host = *.*.*.21
inkey = dundiincomingfour
outkey = dundifour
include = priv
permit = priv
qualify = yes
order = primary
*.*.*.21
iax.conf
[priv]
type=user
context=local-custom
disallow=all
allow=ulaw
allow=alaw
allow=gsm
dundi.conf
[mappings]
priv = dundi-priv-localcustom,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER}
[00:11:0A:34:29:57]
model = symmetric
host = *.*.*.137
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
[00:11:0A:34:29:43]
model = symmetric
host = *.*.*.204
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
.
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Re: [asterisk-users] DUNDI Help
Hi,
I have been testing dundi setup, one thing i am having problem with is that
extensions are getting remove from the regcontext.
does it get removed when registration expires? how can i make sure it's added
back without power cycling the phone? which would be better, making expiration
higher? or lowering it so it will re-register fast? also i am using pap2 and
sipura, is there a settings to make re-register faster?
did you experience this as well before? how were you able to fix it? thank you
regards,
ron
--- On Wed, 8/27/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Wednesday, August 27, 2008, 1:06 PM
Sure, let me show you how I setup dundi on systems.
extensions.conf
exten = _1X,1,Goto(lookupdundi,${EXTEN},1)
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Playback(invalid)
You can have the i do whatever you want, and you can use the same
option in the macro you are using.
That is it, I leave out all the other context in the examples, from
time to time I add a dundi-static context and put in specific numbers
or patterns I want to accept, maybe for pstn calling or phones that
don't register, but in those cases I have multiple mappings in
dundi.conf for each context. For example:
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
priv = dundi-static,0,SIP,[EMAIL PROTECTED],nopartial
On Wed, Aug 27, 2008 at 3:56 AM, ronald ramos [EMAIL PROTECTED]
wrote:
Hi Again,
Is there a way i can detect whether a user has been added into the
regcontext?
Currently i'm seeing this and just gives a fast busy.
[Aug 27 16:44:46] WARNING[17402]: pbx.c:2483 __ast_pbx_run: Channel
'SIP/10.10.10.10-b63101d0' sent into invalid extension
'141100' in context
'lookupdundi', but no invalid handler
can i detect it somehow, so i can inform user that the extensions is not
available?
i have tried ChanIsAvail, but since i am using realtime ChanIsAvail
thinks
it registered, since it really is registered on the other server. So
it's
trying to call it, tries it for 30 secs (i set it to timeout at 30),
after 30 secs then it will go to DUNDI/priv. Is there a way that i can
detect it first so it does not try to dial it on the local before askng
dundi? thank you
regards,
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 8:16 PM
It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.
On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos
[EMAIL PROTECTED]
wrote:
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question
on regcontext though, i set it to sipregistrations, how often
does
an extension be added to the context sipregistrations and for how long
will
it stay there? i'm looking at dialplan show sipregistration,
sometimes
i
only see one extension there. even though i know i have 4 ip phones
registered to the asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald
ramos
[EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2
asterisk.
I copied the config from DUNDI enterprise SIP with no password.
Only
thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime
Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1
ms)
i can see my extension being added
on sipregistrations context
Added extension '136101' priority 1 to
sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls
Re: [asterisk-users] DUNDI Help
Hi Again,
Is there a way i can detect whether a user has been added into the regcontext?
Currently i'm seeing this and just gives a fast busy.
[Aug 27 16:44:46] WARNING[17402]: pbx.c:2483 __ast_pbx_run: Channel
'SIP/10..10.10.10-b63101d0' sent into invalid extension '141100' in context
'lookupdundi', but no invalid handler
can i detect it somehow, so i can inform user that the extensions is not
available?
i have tried ChanIsAvail, but since i am using realtime ChanIsAvail thinks it
registered, since it really is registered on the other server. So it's trying
to call it, tries it for 30 secs (i set it to timeout at 30), after 30 secs
then it will go to DUNDI/priv. Is there a way that i can detect it first so it
does not try to dial it on the local before askng dundi? thank you
regards,
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 8:16 PM
It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.
On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED]
wrote:
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question on regcontext though, i set it to sipregistrations, how often
does
an extension be added to the context sipregistrations and for how long
will
it stay there? i'm looking at dialplan show sipregistration, sometimes
i
only see one extension there. even though i know i have 4 ip phones
registered to the asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos
[EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2
asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only
thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added
on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include =
dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101)
in
new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new
stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel
'SIP/138100-08269548'
status is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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Re: [asterisk-users] DUNDI Help
Sure, let me show you how I setup dundi on systems.
extensions.conf
exten = _1X,1,Goto(lookupdundi,${EXTEN},1)
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Playback(invalid)
You can have the i do whatever you want, and you can use the same
option in the macro you are using.
That is it, I leave out all the other context in the examples, from
time to time I add a dundi-static context and put in specific numbers
or patterns I want to accept, maybe for pstn calling or phones that
don't register, but in those cases I have multiple mappings in
dundi.conf for each context. For example:
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
priv = dundi-static,0,SIP,[EMAIL PROTECTED],nopartial
On Wed, Aug 27, 2008 at 3:56 AM, ronald ramos [EMAIL PROTECTED] wrote:
Hi Again,
Is there a way i can detect whether a user has been added into the
regcontext?
Currently i'm seeing this and just gives a fast busy.
[Aug 27 16:44:46] WARNING[17402]: pbx.c:2483 __ast_pbx_run: Channel
'SIP/10.10.10.10-b63101d0' sent into invalid extension '141100' in context
'lookupdundi', but no invalid handler
can i detect it somehow, so i can inform user that the extensions is not
available?
i have tried ChanIsAvail, but since i am using realtime ChanIsAvail thinks
it registered, since it really is registered on the other server. So it's
trying to call it, tries it for 30 secs (i set it to timeout at 30),
after 30 secs then it will go to DUNDI/priv. Is there a way that i can
detect it first so it does not try to dial it on the local before askng
dundi? thank you
regards,
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 8:16 PM
It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.
On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED]
wrote:
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question
on regcontext though, i set it to sipregistrations, how often
does
an extension be added to the context sipregistrations and for how long
will
it stay there? i'm looking at dialplan show sipregistration, sometimes
i
only see one extension there. even though i know i have 4 ip phones
registered to the asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald
ramos
[EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2
asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only
thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added
on sipregistrations context
Added extension '136101' priority 1 to
sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include =
dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101)
in
new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new
stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel
'SIP
[asterisk-users] DUNDI Help
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID Host Model AvgTime Status
00:8e:8c:8e:cb:53 10.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel 'SIP/138100-08269548' status
is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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Re: [asterisk-users] DUNDI Help
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED] wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel 'SIP/138100-08269548'
status is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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Re: [asterisk-users] DUNDI Help
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question on regcontext though, i set it to sipregistrations, how often does an
extension be added to the context sipregistrations and for how long will it
stay there? i'm looking at dialplan show sipregistration, sometimes i only see
one extension there. even though i know i have 4 ip phones registered to the
asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in
new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel
'SIP/138100-08269548'
status is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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Re: [asterisk-users] DUNDI Help
It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.
On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED] wrote:
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question on regcontext though, i set it to sipregistrations, how often does
an extension be added to the context sipregistrations and for how long will
it stay there? i'm looking at dialplan show sipregistration, sometimes i
only see one extension there. even though i know i have 4 ip phones
registered to the asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added
on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include =
dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in
new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel
'SIP/138100-08269548'
status is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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Re: [asterisk-users] [Dundi] Looking for new peers/limited time only
On Thursday 31 July 2008 11:36:18 am Anthony Messina wrote: For a limited time only, Messinet Secure Services (me) will be offering DUNDi E.164 termination to the entire +1 country code. I'd like to encourage more peering within the US, but peering is open to anyone. See http://messinet.com/?page_name=DUNDi for peering information. During the past 12 days, Messinet Secure Services routed over 2000 free VoIP calls to the +1 country code using DUNDi. We managed to log over 60 hours of call time served and many of our existing peers were able to find new peers willing to join the growing DUNDi cloud. Now that the free calls have come to an end, I want to thank all those who participed in this event. I hope that it has been a success for all involved. See http://messinet.com/?page_name=DUNDi for peering information. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Dundi] Looking for new peers/limited time only
On Thursday 31 July 2008 11:36:18 am Anthony Messina wrote: For a limited time only, Messinet Secure Services (me) will be offering DUNDi E.164 termination to the entire +1 country code. I'd like to encourage more peering within the US, but peering is open to anyone. See http://messinet.com/?page_name=DUNDi for peering information. Between 11AM yesterday and 11AM today, Messinet Secure Services serviced over 430 free calls totaling over 18 hour and 15 minutes of free calling time to any number using the +1 country code. All this was accomplished via DUNDi E.164 peering. Judging by the numbers for yesterday, today will probably be the last day this offer, so peer up now! See http://messinet.com/?page_name=DUNDi for peering information. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi question
Hello, I'm wondering about following DUNDI setup Suppose we have 2 Asterisks: astA and astB with DUNDI peering active between them and 2 SIP endpoints: sipA registered with astA and sipB regsitered on astB All this is on the same LAN now sipA call an number which corresponds to [EMAIL PROTECTED] , so astA lookups thru DUNDI at astB and forwards the call there. My question is how this fowarding is done ? Using SIP RE-INVITE, or REFER, or using SIP 301 responce with Contact pointing at [EMAIL PROTECTED] And does the final RTP stream traverse both Asterisks or only one of them or None of them? Thanks Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi network - redundancy / fault tolerance ?
On Thu, May 8, 2008 at 3:59 AM, Russell Bryant [EMAIL PROTECTED] wrote: Ex Vito wrote: Now, how to move on to acheive some kind of fault tolerance ? According to the docs we've studied, DUNDi does not like loops (which we assume one can limit with low enough TTLs). Which documentation are you referring to? You may have misunderstood something, or there may be some false information floating around the internet (*GASP*). Went back and reviewd the docs (essentially: Asterisk TFOT 2nd ed, wiki, the excellent docs by JR Richardson and dundi.com)... ...in short: nowhere is such statement written. I presume we self-inflicted such idea from the best practices mentioned in dundi.com and from the special attention that should be taken when creating looping topologies regarding TTLs. As I said before, don't worry about loops. Set your TTL to handle a worst case path for a query in your DUNDi topology. Great. That's now clear, thanks. - Assuming any of the above is possible as a means to acheive redundancy, which of the following topologies would your prefer ? ... #1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png #2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png Thanks in advance for review and feedback. I'm not necessarily up on my graph theory, either, but I would probably go with something like #1. After internal discussion and reviewing the final example in the DUNDi protocol draft, while agreeing that the differences are actually small, we are also targetting #1... Again, thanks for your quick feedback, Russel. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi call impossible in one direction
Ciao Matt, Are you using IAX2 as your transport between the 2 servers or SIP? If you are using IAX2, are you using Asterisk 1.4.18.1 or 1.4.19.1 on either machine? If so, you may be encountering the IAX2 bug that some have been discussing on the list recently you can read it here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211000.html No, I'm using Asterisk 1.2.25-bristuffed. I'll be trying Russell's suggestion. Thanks, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi call impossible in one direction
Andrea Spadaccini wrote: I've set up DUNDi between two asterisk boxes, and sometimes happens that calls from machine A can't reach peers in machine B, but calls from B to A work correctly. The strange thing is that the CLI command 'dundi show peers' shows correctly the registered peer in both servers, and in this situation if I make a call from B to A, suddenly peers in server A are able to call peers in machine B. Try using the DUNDi query CLI command to see what results your server is getting when you try to make calls. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dundi network - redundancy / fault tolerance ?
Hi list, I'm planning a private DUNDi network for a cross-country distributed PBX. Initially it will be composed of about 10 systems, growing to about 20. Current requirements point to a topology of two interconnected DUNDi hubs, each peering with half the PBXs... This would lead to two interconnected / inter-peered stars. Example: - Consider PBXs A to H - C and E will be hubs and peer with each other - A, B and D peer with C - F, G and H peer with E This leads to a maximum three hop lookup and will make good use of current network topology / bandwidths. Of course, should any of the hubs be unavailable and the lookup capability is severely compromised. Now, how to move on to acheive some kind of fault tolerance ? According to the docs we've studied, DUNDi does not like loops (which we assume one can limit with low enough TTLs). Our doubts are: - Should one use the order peer parameter to specify alternate lookup paths / peers ? Is that its purpose ? If not, what is it used for ? - Alternatively, should one create loops in the DUNDi topology and limit them via TTL ? - If both options are possible, which would be the trade-offs between them ? (Not clear at all to us!) - Assuming any of the above is possible as a means to acheive redundancy, which of the following topologies would your prefer ? (hmmm, maybe I need to refresh my graph theory...) ;-) #1 - Peer each PBX with both hubs #2 - Duplicate both hubs and peer each PBX with its hub and its hub dup For better understanding, take a look at: #1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png #2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png Thanks in advance for review and feedback. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi network - redundancy / fault tolerance ?
Ex Vito wrote: Now, how to move on to acheive some kind of fault tolerance ? According to the docs we've studied, DUNDi does not like loops (which we assume one can limit with low enough TTLs). Which documentation are you referring to? You may have misunderstood something, or there may be some false information floating around the internet (*GASP*). The DUNDi protocol has built in handling for loops. It keeps track of which nodes have already been queried, so you don't have to worry about loops in your network. Every node can peer with every other node if you really wanted to. Of course, that's not necessarily the most efficient thing to do ... Our doubts are: - Should one use the order peer parameter to specify alternate lookup paths / peers ? Is that its purpose ? If not, what is it used for ? The order parameter is really a tool. There is not an exact situation that it is intended for. It depends on your network. Keep in mind that DUNDi caches results along the way. If you use the order option to have servers send queries through a primary server, you getter better caching performance. - Alternatively, should one create loops in the DUNDi topology and limit them via TTL ? As I said before, don't worry about loops. Set your TTL to handle a worst case path for a query in your DUNDi topology. - If both options are possible, which would be the trade-offs between them ? (Not clear at all to us!) I'm not sure what you mean. The best thing to do is to have multiple peers. Have every server have at least two peers. Setting a primary and secondary can be good for caching reasons. - Assuming any of the above is possible as a means to acheive redundancy, which of the following topologies would your prefer ? (hmmm, maybe I need to refresh my graph theory...) ;-) #1 - Peer each PBX with both hubs #2 - Duplicate both hubs and peer each PBX with its hub and its hub dup For better understanding, take a look at: #1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png #2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png Thanks in advance for review and feedback. I'm not necessarily up on my graph theory, either, but I would probably go with something like #1. A combination of having multiple peers and usage of the order option can give you good redundancy without hurting your performance. When you set primary, secondary, etc. peers, the server will attempt to contact them one at a time. If you have multiple peers, but do not set an order, they will all be contacted at once, which may (probably will) increase latency for call completion, will increase bandwidth consumption, among other things. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi network - redundancy / fault tolerance ?
I don;t have any answers for you... But I would love to hear about the results after you get this working and what road blocks you hit and how you overcame them. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Ex Vito [EMAIL PROTECTED] Sent: Wednesday, May 07, 2008 10:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] dundi network - redundancy / fault tolerance ? Hi list, I'm planning a private DUNDi network for a cross-country distributed PBX. Initially it will be composed of about 10 systems, growing to about 20. Current requirements point to a topology of two interconnected DUNDi hubs, each peering with half the PBXs... This would lead to two interconnected / inter-peered stars. Example: - Consider PBXs A to H - C and E will be hubs and peer with each other - A, B and D peer with C - F, G and H peer with E This leads to a maximum three hop lookup and will make good use of current network topology / bandwidths. Of course, should any of the hubs be unavailable and the lookup capability is severely compromised. Now, how to move on to acheive some kind of fault tolerance ? According to the docs we've studied, DUNDi does not like loops (which we assume one can limit with low enough TTLs). Our doubts are: - Should one use the order peer parameter to specify alternate lookup paths / peers ? Is that its purpose ? If not, what is it used for ? - Alternatively, should one create loops in the DUNDi topology and limit them via TTL ? - If both options are possible, which would be the trade-offs between them ? (Not clear at all to us!) - Assuming any of the above is possible as a means to acheive redundancy, which of the following topologies would your prefer ? (hmmm, maybe I need to refresh my graph theory...) ;-) #1 - Peer each PBX with both hubs #2 - Duplicate both hubs and peer each PBX with its hub and its hub dup For better understanding, take a look at: #1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png #2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png Thanks in advance for review and feedback. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi call impossible in one direction
Are you using IAX2 as your transport between the 2 servers or SIP? If you are using IAX2, are you using Asterisk 1.4.18.1 or 1.4.19.1 on either machine? If so, you may be encountering the IAX2 bug that some have been discussing on the list recently you can read it here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211000.html -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Russell Bryant [EMAIL PROTECTED] Sent: Wednesday, May 07, 2008 6:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi call impossible in one direction Andrea Spadaccini wrote: I've set up DUNDi between two asterisk boxes, and sometimes happens that calls from machine A can't reach peers in machine B, but calls from B to A work correctly. The strange thing is that the CLI command 'dundi show peers' shows correctly the registered peer in both servers, and in this situation if I make a call from B to A, suddenly peers in server A are able to call peers in machine B. Try using the DUNDi query CLI command to see what results your server is getting when you try to make calls. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi call impossible in one direction
Hello everybody, I've set up DUNDi between two asterisk boxes, and sometimes happens that calls from machine A can't reach peers in machine B, but calls from B to A work correctly. The strange thing is that the CLI command 'dundi show peers' shows correctly the registered peer in both servers, and in this situation if I make a call from B to A, suddenly peers in server A are able to call peers in machine B. Can anyone give me directions? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi and SIP
I think I'm going to go about this a different way, if it works I'll post my
solution.
Essentially I'm going to limit the calls by grouping(didn't know you could use
categories until I did the research) and math. Limiting our corporate office
to 10 IAX calls, both incoming and outgoing together, and denying the call if
it's above that(sending chanunavail or something similar).
I'll then run all dials through a macro, looking up dundi routes. If it fails
I'll fall back to zap.
Thanks for the help though.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 23, 2008 5:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Take a look at this setup, it does not use passwords on the sip peers
or the mappings in Dundi. As long as you inside your network this
maybe the way to go.
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
You could also look at the incominglimit and outgoinglimit on IAX peers
On Wed, Apr 23, 2008 at 4:51 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm fairly sure SIP will never work unless I hard-code peers everywhere,
which isn't going to happen. The only reason I want to use it is for the
call-limit option.
Looking at sip channels there is no option to pass the extension after the
IP, it's always [EMAIL PROTECTED], or [EMAIL PROTECTED], not [EMAIL
PROTECTED]/extension or [EMAIL PROTECTED]/extension
Looks like IAX and ZAP are the only two channel types that do a /extension
type setup.
Extensions.conf:
[macro-dundi-lookup]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-local
include = dundi-priv-lookup
[dundi-priv-local]
include = internal
[dundi-priv-lookup]
switch = DUNDi/priv
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 23, 2008 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
It is not the dip peer that is failing but the dial plan:
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
== Everyone is busy/congested at this time (1:0/0/1)
What is in the context macro-dundi-lookup?
On Wed, Apr 23, 2008 at 12:47 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Nope..
asterisk*CLI dundi lookup [EMAIL PROTECTED]
1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
from 00:1e:0b:dd:e9:99, expires in 5 s
DUNDi lookup completed in 104 ms
-- Executing [EMAIL PROTECTED]:1] Set(SIP/156-08274b60,
CDR(accountcode)=wth) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/156-08274b60,
CALLERID(all)=Corporate 100) in new stack
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/156-08274b60,
dundi-lookup|400) in new stack
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/156-08274b60, 400|1) in
new stack
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:4] Hangup(SIP/156-08274b60, ) in
new stack
== Spawn extension (from-sip, 400, 4) exited non-zero on
'SIP/156-08274b60'
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I
can't specify
Re: [asterisk-users] DUNDi and SIP
Nope..
asterisk*CLI dundi lookup [EMAIL PROTECTED]
1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
from 00:1e:0b:dd:e9:99, expires in 5 s
DUNDi lookup completed in 104 ms
-- Executing [EMAIL PROTECTED]:1] Set(SIP/156-08274b60,
CDR(accountcode)=wth) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/156-08274b60,
CALLERID(all)=Corporate 100) in new stack
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/156-08274b60,
dundi-lookup|400) in new stack
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/156-08274b60, 400|1) in new
stack
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such host:
192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:4] Hangup(SIP/156-08274b60, ) in new
stack
== Spawn extension (from-sip, 400, 4) exited non-zero on 'SIP/156-08274b60'
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box
is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL
PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments
Re: [asterisk-users] DUNDi and SIP
Jeremy,
It is not the dip peer that is failing but the dial plan:
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
== Everyone is busy/congested at this time (1:0/0/1)
What is in the context macro-dundi-lookup?
On Wed, Apr 23, 2008 at 12:47 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Nope..
asterisk*CLI dundi lookup [EMAIL PROTECTED]
1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
from 00:1e:0b:dd:e9:99, expires in 5 s
DUNDi lookup completed in 104 ms
-- Executing [EMAIL PROTECTED]:1] Set(SIP/156-08274b60,
CDR(accountcode)=wth) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/156-08274b60,
CALLERID(all)=Corporate 100) in new stack
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/156-08274b60,
dundi-lookup|400) in new stack
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/156-08274b60, 400|1) in
new stack
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such host:
192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:4] Hangup(SIP/156-08274b60, ) in new
stack
== Spawn extension (from-sip, 400, 4) exited non-zero on 'SIP/156-08274b60'
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I
can't specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL
PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other
box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
Re: [asterisk-users] DUNDi and SIP
I'm fairly sure SIP will never work unless I hard-code peers everywhere, which
isn't going to happen. The only reason I want to use it is for the call-limit
option.
Looking at sip channels there is no option to pass the extension after the IP,
it's always [EMAIL PROTECTED], or [EMAIL PROTECTED], not [EMAIL
PROTECTED]/extension or [EMAIL PROTECTED]/extension
Looks like IAX and ZAP are the only two channel types that do a /extension type
setup.
Extensions.conf:
[macro-dundi-lookup]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-local
include = dundi-priv-lookup
[dundi-priv-local]
include = internal
[dundi-priv-lookup]
switch = DUNDi/priv
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 23, 2008 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
It is not the dip peer that is failing but the dial plan:
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
== Everyone is busy/congested at this time (1:0/0/1)
What is in the context macro-dundi-lookup?
On Wed, Apr 23, 2008 at 12:47 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Nope..
asterisk*CLI dundi lookup [EMAIL PROTECTED]
1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
from 00:1e:0b:dd:e9:99, expires in 5 s
DUNDi lookup completed in 104 ms
-- Executing [EMAIL PROTECTED]:1] Set(SIP/156-08274b60,
CDR(accountcode)=wth) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/156-08274b60,
CALLERID(all)=Corporate 100) in new stack
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/156-08274b60,
dundi-lookup|400) in new stack
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/156-08274b60, 400|1) in
new stack
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such host:
192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:4] Hangup(SIP/156-08274b60, ) in new
stack
== Spawn extension (from-sip, 400, 4) exited non-zero on 'SIP/156-08274b60'
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I
can't specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL
PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure
Re: [asterisk-users] DUNDi and SIP
Take a look at this setup, it does not use passwords on the sip peers
or the mappings in Dundi. As long as you inside your network this
maybe the way to go.
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
You could also look at the incominglimit and outgoinglimit on IAX peers
On Wed, Apr 23, 2008 at 4:51 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm fairly sure SIP will never work unless I hard-code peers everywhere,
which isn't going to happen. The only reason I want to use it is for the
call-limit option.
Looking at sip channels there is no option to pass the extension after the
IP, it's always [EMAIL PROTECTED], or [EMAIL PROTECTED], not [EMAIL
PROTECTED]/extension or [EMAIL PROTECTED]/extension
Looks like IAX and ZAP are the only two channel types that do a /extension
type setup.
Extensions.conf:
[macro-dundi-lookup]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-local
include = dundi-priv-lookup
[dundi-priv-local]
include = internal
[dundi-priv-lookup]
switch = DUNDi/priv
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 23, 2008 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
It is not the dip peer that is failing but the dial plan:
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
== Everyone is busy/congested at this time (1:0/0/1)
What is in the context macro-dundi-lookup?
On Wed, Apr 23, 2008 at 12:47 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Nope..
asterisk*CLI dundi lookup [EMAIL PROTECTED]
1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
from 00:1e:0b:dd:e9:99, expires in 5 s
DUNDi lookup completed in 104 ms
-- Executing [EMAIL PROTECTED]:1] Set(SIP/156-08274b60,
CDR(accountcode)=wth) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/156-08274b60,
CALLERID(all)=Corporate 100) in new stack
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/156-08274b60,
dundi-lookup|400) in new stack
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/156-08274b60, 400|1) in
new stack
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:4] Hangup(SIP/156-08274b60, ) in
new stack
== Spawn extension (from-sip, 400, 4) exited non-zero on
'SIP/156-08274b60'
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I
can't specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
Bruce Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but
the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box
Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.
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EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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to whom it is addressed. If you are the intended recipient, further
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intended recipient, any disclosure, copying, printing, or use of this
information is strictly prohibited and possibly a violation of federal or
state law and regulations. If you have received this information in error,
please notify Texas Health
Re: [asterisk-users] DUNDi and SIP
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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To UNSUBSCRIBE or update options visit:
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This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and entity(ies)
to whom it is addressed. If you are the intended recipient, further
disclosures are prohibited without proper authorization. If you
Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box
is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL
PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments
transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you
are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of
federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
___
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Re: [asterisk-users] DUNDi and SIP
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
to this information.
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[asterisk-users] DUNDi and SIP
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]mailto:SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and pass
the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer a.b.c.d/${NUMBER}
cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip peer
per box(I have 25 connected in my dundi cloud), nor would I like to enable
anonymous SIP calls, as I have the ports open to the world for inbound sip from
bandwidth.com
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
to this information.
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[asterisk-users] DUNDi or ENUM
Hi List, What is the best way to implement this solution with several branches? SIP PHONE 1 --- *SERVER 1 ---WAN--- *SERVER 2 --- SIP PHONE 2 WAN SIP PHONE 3 --- *SERVER 3 ---WAN--- *SERVER 4 --- SIP PHONE 4 The Asterisk boxes will have one leg on the LAN side and another on the WAN side. The SIP phones will only be able to see the local server and not the remote server or the remote SIP phone since the WAN service may be provided by different service providers I guess I would have to do this with SIP or IAX trunks Would DUNDi or ENUM work in this situation? Thanks Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi
Is there a way to have a dundi host advertise extensions for another server? A---B---C I'd like A to reach C through B. A and C would handle the call, B would just be the DUNDi intermediary. Assuming A has 101-199 B has 201-299 And C has 301-399 A sample dundi/extensions/iax config for B is all I need. I can get single DUNDi queries running fine(A-B, B-C, A-C(directly setup in dundi.conf)) I just have 25 offices that all connect to a central location, I'd rather the central location be the hub of all dundi queries for all other locations. Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi
Nevermind, figured it out. I had restrictions on the unsolicited calls in dundi.conf. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Wednesday, March 12, 2008 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DUNDi Is there a way to have a dundi host advertise extensions for another server? A---B---C I'd like A to reach C through B. A and C would handle the call, B would just be the DUNDi intermediary. Assuming A has 101-199 B has 201-299 And C has 301-399 A sample dundi/extensions/iax config for B is all I need. I can get single DUNDi queries running fine(A-B, B-C, A-C(directly setup in dundi.conf)) I just have 25 offices that all connect to a central location, I'd rather the central location be the hub of all dundi queries for all other locations. Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi
A---B---C I’d like A to reach C through B. A and C would handle the call, B would just be the DUNDi intermediary. I don't know if you can cut C out of the loop completely (so B responds to all requests on C's behalf), but you can enable precaching which'll avoid most of the A - C queries. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi with two servers
Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch = DUNDi/context. dundi lookup number@dundified works great as well as test calls. What is the proper method of handling DUNDi between only two servers? Should I be using a dummy context on one server to handle this? I'm listing the relevant files below for only one server for brevities sake. --- dundi.conf [general] department=Test Lab organization=My Test lab locality=Anywhere stateprov=CA country=US [EMAIL PROTECTED] phone=+55 entityid=00:11:22:33:44:55 cachetime=5 ttl=1 autokill=yes [mappings] dundified = internal,0,SIP,[EMAIL PROTECTED][EMAIL PROTECTED] ,nopartial [55:44:33:22:11:00] model=symmetric host=server2.domain.com inkey=dundikey outkey=dundikey include=dundified permit=dundified qualify=yes order=primary --- extensions.conf [general] static = yes writeprotect = no clearglobalvars = no [globals] [default] include = internal include = parkedcalls [internal] include = external include = parkedcalls switch = DUNDi/dundified exten = 300,1,Dial(SIP/300) exten = 300,n,Hangup() exten = 5551234567,1,Goto(300,1) exten = 301,1,Dial(SIP/301) exten = 301,n,Hangup() exten = 8885551212,1,Goto(301,1) exten = _NXXNXX,1,Dial([EMAIL PROTECTED]) exten = _NXXNXX,n,Hangup() [external] exten = 5551234567,1,Goto(internal,300,1) --- sip.conf [dundified] type=friend dbsecret=dundi/secret context=internal [voipprovider] type=friend host=voipprovider.web dtmfmode=rfc2833 insecure=port,invite disallow=all allow=g729 context=external [300] type=peer callerid=300 username=300 secret=secret host=dynamic context=internal [EMAIL PROTECTED] notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 [301] type=peer callerid=301 username=301 secret=secret host=dynamic context=internal [EMAIL PROTECTED] notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 Thanks in advance! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi lookup
--- Jared Smith [EMAIL PROTECTED] wrote: However, when used in conjunction with the regexten (and corresponding regcontext) settings available in both iax.conf and sip.conf, it effectively allows you to route calls to the host on which a device has registered. Now I see. Very useful. I'm reading this guide: http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf which is very informative but I hope I can find a way to do the same thing without Realtime. In fact, I don't see why it shouldn't work with plain .conf files (but then again I may be missing something). Thanks again. Vieri Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi with two servers
On Sunday 24 February 2008 02:01:10 am arkda wrote: Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch = DUNDi/context. dundi lookup number@dundified works great as well as test calls. What is the proper method of handling DUNDi between only two servers? Should I be using a dummy context on one server to handle this? I'm listing the relevant files below for only one server for brevities sake. --- dundi.conf [general] department=Test Lab organization=My Test lab locality=Anywhere stateprov=CA country=US [EMAIL PROTECTED] phone=+55 entityid=00:11:22:33:44:55 cachetime=5 ttl=1 autokill=yes [mappings] dundified = internal,0,SIP,[EMAIL PROTECTED][EMAIL PROTECTED] ,nopartial [55:44:33:22:11:00] model=symmetric host=server2.domain.com inkey=dundikey outkey=dundikey include=dundified permit=dundified qualify=yes order=primary --- extensions.conf [general] static = yes writeprotect = no clearglobalvars = no [globals] [default] include = internal include = parkedcalls [internal] include = external include = parkedcalls switch = DUNDi/dundified exten = 300,1,Dial(SIP/300) exten = 300,n,Hangup() exten = 5551234567,1,Goto(300,1) exten = 301,1,Dial(SIP/301) exten = 301,n,Hangup() exten = 8885551212,1,Goto(301,1) exten = _NXXNXX,1,Dial([EMAIL PROTECTED]) exten = _NXXNXX,n,Hangup() [external] exten = 5551234567,1,Goto(internal,300,1) --- sip.conf [dundified] type=friend dbsecret=dundi/secret context=internal [voipprovider] type=friend host=voipprovider.web dtmfmode=rfc2833 insecure=port,invite disallow=all allow=g729 context=external [300] type=peer callerid=300 username=300 secret=secret host=dynamic context=internal [EMAIL PROTECTED] notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 [301] type=peer callerid=301 username=301 secret=secret host=dynamic context=internal [EMAIL PROTECTED] notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 Thanks in advance! i believe that you don't want to have your mappings point to a context that includes the switch = DUNDi/... statement. The switch is what a server uses to interface to the rest of the DUNDi world. your mappings should point to what a particular host is serving up, not including the rest of the DUNDi world. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi with two servers
JR Richardson Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch = DUNDi/context. dundi lookup number@dundified works great as well as test calls. What is the proper method of handling DUNDi between only two servers? Should I be using a dummy context on one server to handle this? I'm listing the relevant files below for only one server for brevities sake. --- dundi.conf [general] department=Test Lab organization=My Test lab locality=Anywhere stateprov=CA country=US [EMAIL PROTECTED] phone=+55 entityid=00:11:22:33:44:55 cachetime=5 ttl=1 autokill=yes [mappings] dundified = internal,0,SIP,[EMAIL PROTECTED][EMAIL PROTECTED] ,nopartial [55:44:33:22:11:00] model=symmetric host=server2.domain.com inkey=dundikey outkey=dundikey include=dundified permit=dundified qualify=yes order=primary --- extensions.conf [general] static = yes writeprotect = no clearglobalvars = no [globals] [default] include = internal include = parkedcalls [internal] include = external include = parkedcalls switch = DUNDi/dundified exten = 300,1,Dial(SIP/300) exten = 300,n,Hangup() exten = 5551234567,1,Goto(300,1) exten = 301,1,Dial(SIP/301) exten = 301,n,Hangup() exten = 8885551212,1,Goto(301,1) exten = _NXXNXX,1,Dial([EMAIL PROTECTED]) exten = _NXXNXX,n,Hangup() [external] exten = 5551234567,1,Goto(internal,300,1) --- sip.conf [dundified] type=friend dbsecret=dundi/secret context=internal [voipprovider] type=friend host=voipprovider.web dtmfmode=rfc2833 insecure=port,invite disallow=all allow=g729 context=external [300] type=peer callerid=300 username=300 secret=secret host=dynamic context=internal [EMAIL PROTECTED] notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 [301] type=peer callerid=301 username=301 secret=secret host=dynamic context=internal [EMAIL PROTECTED] notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 Thanks in advance! You can't map the [internal] context in dundi.conf because you have the switch = DUNDi/dandified statement in there. That is causing your loop. Map dundi to a dial plan [context] that doesn't have access to the [internal] context. Put your dundi extensions in the new context as a NoOp and things should work fine. JR --- Engineering for the Masses. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi with two servers
Aha, thanks guys. I like the idea of just using NoOp extensions that don't actually do anything so I'm going to give that a shot. On Sun, Feb 24, 2008 at 2:14 PM, JR Richardson [EMAIL PROTECTED] wrote: JR Richardson You can't map the [internal] context in dundi.conf because you have the switch = DUNDi/dandified statement in there. That is causing your loop. Map dundi to a dial plan [context] that doesn't have access to the [internal] context. Put your dundi extensions in the new context as a NoOp and things should work fine. JR --- Engineering for the Masses. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dundi lookup
Pardon my ignorance but I understand that DUNDi lookups (*CLI dundi lookup [EMAIL PROTECTED]) reveal if a given extension is served by some host, ie. if it's present in its dialplan. It does not say if it's registered or not. Is this correct? Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi lookup
On Sat, 2008-02-23 at 12:20 -0800, Vieri wrote: Pardon my ignorance but I understand that DUNDi lookups (*CLI dundi lookup [EMAIL PROTECTED]) reveal if a given extension is served by some host, ie. if it's present in its dialplan. It does not say if it's registered or not. Is this correct? That is correct. a DUNDi lookup will only tell you if that extension (or a patern match that happens to match that extension) exists in the advertising context on the DUNDi server. However, when used in conjunction with the regexten (and corresponding regcontext) settings available in both iax.conf and sip.conf, it effectively allows you to route calls to the host on which a device has registered. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi ${NUMBER} variable not defined
If one does a dundi lookup, shouldn't the ${NUMBER}
variable be replaced with the current value?
ie. if I run dundi lookup [EMAIL PROTECTED] shouldn't I get
an answer string like
IAX2/priv:[EMAIL PROTECTED]/4065 (EXISTS)?
The *CLI does not show me the dst extension:
*CLI dundi lookup [EMAIL PROTECTED]
1. 0
IAX2/priv:[EMAIL PROTECTED]/${NUMBER}
(EXISTS)
from 00:1d:60:b0:25:10, expires in 5 s
DUNDi lookup completed in 53 ms
*CLI dundi lookup [EMAIL PROTECTED]
1. 0
IAX2/priv:[EMAIL PROTECTED]/${NUMBER}
(EXISTS)
from 00:1d:60:b0:25:10, expires in 5 s
DUNDi lookup completed in 37 ms
What could be wrong?
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[asterisk-users] DUNDI setup help
HELLO ALL!
I followed a tutorial called DUNDi so easy to set up DUNDi peers.
Unsurprising it was not that easy hehe.
I have the following files up and running, peers are visible but when I
do a query e.g dundi lookup [EMAIL PROTECTED] I get the following error.
CAUSE: NOAUTH: Unsupported DUNDi context.
Could anybody help?
Thank you kindly.
James
=== IAX.CONF on both servers ===
[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi
=== DUNDI.CONF FILE ON SERVER 1 ===
[mappings]
priv=dundiextens,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},noparti
al
[00:0B:CD:08:23:00] ;We can see server .151
model=symmetric
host=XXX.XXX.XX.151
inkey=dundi
outkey=dundi
include=priv
permit=priv
qualify=yes
order=primary
=== DUNDI.CONF FILE ON SERVER 2 ===
[mappings]
priv=dundiextens,0,IAX2,priv:${SECRET}XXX.XXX.XX.151/${NUMBER},nopartia
l
[00:0B:CD:08:22:F6] ;We Can see the Server .150
model=symmetric
host=XXX.XXX.XX.150
inkey=dundi
outkey=dundi
include=priv
permit=priv
qualify=yes
order=primary
=== DUNDI.CONF FILE ON SERVER 1 ===
[General]
[lookupdundi]
;this is where DUNDi querys the peers and requests and extension
switch = DUNDI/priv
[dundiextens]
;this is where we list the actual extensions that this pbx responds to
exten = AS,1,NoOp
[incomingdundi]
;this is the entry point where dundi calls come into this server - we
specified this context in iax.conf
;simply forward the actual extenstion into [internal] using goto
exten = AS,1,Goto(internal|AS|1)
[internal]
;change the context and executethe switch statement which enables dundi
to query the peers
include = lookupdundi
; phone line AS
exten = AS,1,MixMonitor(ASDUNDI.wav|av(0)V(0))
exten = AS,2,Dial(SIP/ASCHCP)
exten = AS,3,Answer()
exten = AS,4,Busy(10)
exten = AS,5,Hangup()
=== DUNDI.CONF FILE ON SERVER 2 ===
[General]
[lookupdundi]
;this is where DUNDi querys the peers and requests and extension
switch = DUNDI/priv
[dundiextens]
;this is where we list the actual extensions that this pbx responds to
exten = DO,1,NoOp
[incomingdundi]
;this is the entry point where dundi calls come into this server - we
specified this context in iax.conf
;simply forward the actual extenstion into [internal] using goto
exten = DO,1,Goto(internal|DO|1)
[internal]
;change the context and executethe switch statement which enables dundi
to query the peers
include = lookupdundi
; phone line DO
exten = DO,1,MixMonitor(DODUNDI.wav|av(0)V(0))
exten = DO,2,Dial(SIP/DO)
exten = DO,3,Answer()
exten = DO,4,Busy(10)
exten = DO,5,Hangup()
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Re: [asterisk-users] DUNDI setup help
Could anybody help? Can you show a CLI session? The error you get is not familiar. Otherwise your configs look ok, did you make the keys priv or dundi? There was an error in the howto, the example was to make the keys named priv but in dundi.conf the keys were named dundi, double check that as well. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDI
Hi ALL; Any one knows a websites that has really a members that use DUDNI wouldwide and ready to do route exchanges? I tried www.dundi.com but it look like still not working, as most of its pages are not accessible except the home page :) - Regards Bilal Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi, regcontext, softphones.. fail.
JR Richardson [EMAIL PROTECTED] writes:
I'm having an issue deploying softphones into my DUNDi/regcontext
setup. My current design is that all SIP users get registered into a
sipregistration context in the sip.conf. I then have a dialplan
function that includes that and does the dial:
include = sipregistration
exten = _,2,Answer()
exten = _,3,Wait(1)
exten = _,4,NoOp(sipregistration call - Name: ${CALLERID(name)} Num:
${CALLERID(num)})
exten = _,5,Macro(stdexten,${EXTEN},SIP/${EXTEN}) ; 913-563-77XX
This works really well for hard phones. They register, exist in the
sipregistration context and are dialed on whichever server they register
by DUNDi. I only started to run into problems when I had to deploy
softphones.
The softphones register when they are up and running, and the system
works as designed. But when they close their softphone, there's no way
for the system to know where the extension is, so the call dies. It
doesn't go to voicemail like I would like it to because that extension
never proceeds through my dialplan.
Looking for suggestions on getting around this so I can keep deploying
soft phones to agents in the field.
Just use an 'invalid' extension to send the call to voicemail or something.
exten = i,1,Voicemail(u${INVALID_EXTEN})
I would *love* for it to be that simple, but I'm not doing this from an
IVR. Voip-Info says about the 'i' extension:
The 'i' extension only gets fired when there's a prompt or input been
made with 'background'. You can set up a 'exten = i,1...' to prompt for
wrong keypresses - insult the user and so on. So this wont work if
someone just dials somthing wrong.
--
Kyle Sexton
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Re: [asterisk-users] DUNDi, regcontext, softphones.. fail.
On 18:04, Tue 09 Oct 07, Kyle Sexton wrote:
JR Richardson [EMAIL PROTECTED] writes:
I'm having an issue deploying softphones into my DUNDi/regcontext
setup. My current design is that all SIP users get registered into a
sipregistration context in the sip.conf. I then have a dialplan
function that includes that and does the dial:
include = sipregistration
exten = _,2,Answer()
exten = _,3,Wait(1)
exten = _,4,NoOp(sipregistration call - Name: ${CALLERID(name)} Num:
${CALLERID(num)})
exten = _,5,Macro(stdexten,${EXTEN},SIP/${EXTEN}) ; 913-563-77XX
This works really well for hard phones. They register, exist in the
sipregistration context and are dialed on whichever server they register
by DUNDi. I only started to run into problems when I had to deploy
softphones.
The softphones register when they are up and running, and the system
works as designed. But when they close their softphone, there's no way
for the system to know where the extension is, so the call dies. It
doesn't go to voicemail like I would like it to because that extension
never proceeds through my dialplan.
Looking for suggestions on getting around this so I can keep deploying
soft phones to agents in the field.
Just use an 'invalid' extension to send the call to voicemail or something.
exten = i,1,Voicemail(u${INVALID_EXTEN})
I would *love* for it to be that simple, but I'm not doing this from an
IVR. Voip-Info says about the 'i' extension:
The 'i' extension only gets fired when there's a prompt or input been
made with 'background'. You can set up a 'exten = i,1...' to prompt for
wrong keypresses - insult the user and so on. So this wont work if
someone just dials somthing wrong.
I did not follow this thread at all so please excuse me if
I'm redundant.
If you dont trust the i exten to handle things make sure
everything you want to handle is configured and put
something like this in you dialplan:
exten = _.,1,Goto(i,1)
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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Re: [asterisk-users] DUNDi, regcontext, softphones.. fail.
I'm having an issue deploying softphones into my DUNDi/regcontext
setup. My current design is that all SIP users get registered into a
sipregistration context in the sip.conf. I then have a dialplan
function that includes that and does the dial:
include = sipregistration
exten = _,2,Answer()
exten = _,3,Wait(1)
exten = _,4,NoOp(sipregistration call - Name: ${CALLERID(name)} Num:
${CALLERID(num)})
exten = _,5,Macro(stdexten,${EXTEN},SIP/${EXTEN}) ; 913-563-77XX
This works really well for hard phones. They register, exist in the
sipregistration context and are dialed on whichever server they register
by DUNDi. I only started to run into problems when I had to deploy
softphones.
The softphones register when they are up and running, and the system
works as designed. But when they close their softphone, there's no way
for the system to know where the extension is, so the call dies. It
doesn't go to voicemail like I would like it to because that extension
never proceeds through my dialplan.
Looking for suggestions on getting around this so I can keep deploying
soft phones to agents in the field.
Just use an 'invalid' extension to send the call to voicemail or something.
exten = i,1,Voicemail(u${INVALID_EXTEN})
JR
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[asterisk-users] DUNDi, regcontext, softphones.. fail. :(
All,
I'm having an issue deploying softphones into my DUNDi/regcontext
setup. My current design is that all SIP users get registered into a
sipregistration context in the sip.conf. I then have a dialplan
function that includes that and does the dial:
include = sipregistration
exten = _,2,Answer()
exten = _,3,Wait(1)
exten = _,4,NoOp(sipregistration call - Name: ${CALLERID(name)} Num:
${CALLERID(num)})
exten = _,5,Macro(stdexten,${EXTEN},SIP/${EXTEN}) ; 913-563-77XX
This works really well for hard phones. They register, exist in the
sipregistration context and are dialed on whichever server they register
by DUNDi. I only started to run into problems when I had to deploy softphones.
The softphones register when they are up and running, and the system
works as designed. But when they close their softphone, there's no way
for the system to know where the extension is, so the call dies. It
doesn't go to voicemail like I would like it to because that extension
never proceeds through my dialplan.
Looking for suggestions on getting around this so I can keep deploying
soft phones to agents in the field.
Thanks in advance!
--
Kyle Sexton
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Re: [asterisk-users] DUNDi, So Easy A Caveman Could Do It!
Well done! It's top-news on AstPligg right now. http://oinko.net/astpligg/story.php?title=DUNDi_So_Easy_A_Caveman_Could_Do_It Thanks l. On Wed, 22 Aug 2007 03:51:51 +0200, JR Richardson [EMAIL PROTECTED] wrote: Here you go folks: ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf If someone would be so kind as to upload to the wiki, it will be much appriciated. Thank you all who replied to my poll questions. As always, I hope this help. JR -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi, So Easy A Caveman Could Do It!
Here you go folks: ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf If someone would be so kind as to upload to the wiki, it will be much appriciated. Thank you all who replied to my poll questions. As always, I hope this help. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi limitation?
Greetings list,
I've been using DUNDi for some time now to prevent calls between users going
out via PSTN if there's no need, set up as follows:
[macro-dundi-e164]
exten = s,1,Goto(${ARG1},1)
include = dundi-e164
[dundi-e164]
include = in-e164
switch = DUNDi/e164
My outbound call macro tries [macro-dundi-e164] first, if that fails then goes
onto PSTN connectivity.
However, I noticed a strange error last night when programming up a simple IVR
for a new customer. I added the following to [in-e164]:
exten = number,n,Background(ivr/hello)
exten = number,n,WaitExten(5)
; service
exten = 1,1,Macro(queue, service,300)
; accounts
exten = 2,1,Macro(queue,accounts,300)
exten = t,1,Goto(number,1)
(e164 number removed to protect client's privacy)
This works fine when calling externally, but when calling internally via DUNDi,
it's impossible to hit either IVR button because the call appears to still be
in the originating context rather than [in-e164].
I tried getting round this by changing the first line in [macro-dundi-e164]
from:
exten = s,1,Goto(${ARG1},1)
to:
exten = s,1,Goto(in-e164,${ARG1},1)
However, this breaks all external calling with an invalid extension error.
Can anyone suggest a way round this?
Thanks in advance.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
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