Re: [asterisk-users] SIP to Analog Devices
Brian, From http://www.voiplink.com/Linksys_Analog_Telephone_Adapters_s/51.htm they have adaptors compatible with Asterisk, but explicitly say in the product titles that they're unlocked, which I think is the key. On Thu, Dec 17, 2009 at 4:16 AM, Brian Cline br...@nw.brian.fm wrote: Hello, I'm running Asterisk v1.6.1.11 internally with a few Linksys SIP phones and will be receiving a machine containing a Dialogic card for a development project (in a nutshell, the card receives analog calls while the accompanying software handles automated prompts, etc). The Dialogic card is not SIP-based but will work with an analog line, so I'm looking into adapters that act themselves as SIP devices but provide an analog port on the other end so that I can internally dial that device's extension, and thus interact with the Dialogic card as though I had a basic POTS line attached. Mostly looking for input and recommendations on these kinds of adapters. Here are a few I've found so far. * Cisco/Linksys PAP2T - appears to be locked into specific mainstream VoIP vendors. Seems to be confirmed with * Cisco/Linksys SPA2101 and SPA3102 - looks promising, but not clear from the product literature whether either is just a straight SIP device and not VoIP vendor specific. Can anyone provide input on these, or other recommendations for this kind of device -- or in the event I am totally wrong about whether this would actually work, suggest alternatives? Many thanks. Brian ___ -- 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 -- David Cunningham Voisonics Limited IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- 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] Best way ro run 2 or more asterisk servers?
Since you're not registering either server with the other, where is the call
going to pick up the required username and password to connect to the other
server?
On Wed, Dec 16, 2009 at 12:09 AM, Landy Landy landysacco...@yahoo.comwrote:
I'm trying to get two server communicate with each other and call from one
to the other but, I'm having a lot of problems.
I tried to create a iax trunk between the two:
At the server:
[client]
type=friend
username=asterisk2
authuser=asterisk2
fromuser=asterisk2
secret=sss
auth=md5
context=from_client
;peercontext=from_asterisk
host=172.16.0.11
trunk=yes
qualify=yes
iax2 show peers
Name/UsernameHost Mask Port Status
client/asterisk 172.16.0.11 (S) 255.255.255.255 4569 (T) (E) OK (3
ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
extensions.conf
[to_client]
exten = _3XX,1,Verbose(. To Asterisk2 Server .)
exten = _3XX,n,Dial(IAX2/${ext...@client)
exten = _3XX,n,Hangup()
[from_client]
include = local-dial
At the client:
[server]
type=friend
host=172.16.0.3
username=asterisk
authuser=asterisk
fromuser=asterisk
secret=xxx
context=from_server
trunk=yes
auth=md5
qualify=yes
iax2 show peers
Name/UsernameHost Mask Port Status
server/asterisk 172.16.0.3 (S) 255.255.255.255 4569 (T) (E) OK (3
ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
extensions.conf
[from_server]
include = local-dial
[to_server]
exten = _5XXX,1,Verbose(. Trying to contact ${EXTEN:1} @ asterisk
.)
exten = _5XXX,n,Dial(IAX2/${ext...@server)
exten = _5XXX,n,Hangup
According to some reading, I do NOT need to register neither one.
When I try to call from one end to the other I get:
[Dec 15 03:06:04] NOTICE[4265]: chan_iax2.c:10338 socket_process: Host
172.16.0.3 failed to authenticate as 300
Please help.
Thanks.
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Re: [asterisk-users] New 1.4 system: registered, but not responding to invite?
Doug, It doesn't respond to the INVITE - the trace says No response to the INVITE?. If the phone doesn't even ring it's probably not getting anything, which points to a problem with the router it's behind. How is the router set up to deliver SIP and RTP to the phone? On Tue, Dec 22, 2009 at 5:33 AM, Doug d...@natel.net wrote: At 00:46 12/21/2009, Alex Balashov wrote: A packet capture would be needed to illuminate the source of the problem. Thanks, Alex for your suggestion. Here is a link for the packet capture: http://www.A7H.com/~stuph/TCPdump-2009-Dec-21-2304.txthttp://www.A7H.com/%7Estuph/TCPdump-2009-Dec-21-2304.txt I just don't see where the extension responds to the INVITE. What would prevent that? By the way, I have a bunch of phones behind this same router that work just fine on our old v1.2 system. On 12/21/2009 01:39 AM, Doug wrote: I've turned on NAT everywhere I can think, but even though I hear ringing on the calling phone (different system) the called phone does not ring. Has anyone bumped into this lately? ___ -- 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 -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- 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] Inquiry:Connect my Asterisk to external sip?
Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using sip set debug on might help you with the Host '192.168.0.139' does not implement 'REGISTER' problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi motamed...@gmail.com wrote: On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @192.168.0.139. To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.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 Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you 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 -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- 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] Problems with chan_sip
Jonas, Some possible causes: - File permission problem - Firewall blocking - Other network problem like no route On Wed, Dec 23, 2009 at 10:20 AM, jonas kellens jonas.kell...@telenet.bewrote: Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to 'sip:092xx9...@85.xx.xx.xx;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted[Dec 23 11:11:07] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:07] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:07] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted[Dec 23 11:11:08] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:08] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:08] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted[Dec 23 11:11:09] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:09] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:09] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted[Dec 23 11:11:10] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:10] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:10] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted[Dec 23 11:11:20] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:20] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:20] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted What is going on here ?? Jonas. ___ -- 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 -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- 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] Can't load cdr_radius.so module?
Shukun, It tells you No such file or directory. Is the file in your modules directory? On Wed, Dec 23, 2009 at 10:09 AM, Zhang Shukun bit...@gmail.com wrote: hi , all when i do the command module load cdr_radius.so ,error happens. i have installed radiusclient-ng , what's wrong with it? thanks! error message as follow: ZHANGSHUKUN*CLI module load cdr_radius.so Unable to load module cdr_radius.so Command 'module load cdr_radius.so' failed. [Dec 23 17:55:41] WARNING[31072]: loader.c:380 load_dynamic_module: Error loading module 'cdr_radius.so': libradiusclient-ng.so.2: cannot open shared object file: No such file or directory [Dec 23 17:55:41] WARNING[31072]: loader.c:730 load_resource: Module 'cdr_radius.so' could not be loaded. -- Regards, Sucan ___ -- 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 -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- 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] Inquiry:Connect my Asterisk to external sip?
AsteriskWin32 does have SIP server functionality, same as the linux version. I can't think of any reason why having your CentOS Asterisk be both client and server and register with itself wouldn't work. Although I am wondering how much help all this will be in debugging a connection problem to another SIP provider... On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi motamed...@gmail.comwrote: On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham dcunning...@voisonics.com wrote: Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using sip set debug on might help you with the Host '192.168.0.139' does not implement 'REGISTER' problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi motamed...@gmail.comwrote: On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.comwrote: On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @ 192.168.0.139 . To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.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 Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you 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 -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- 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 I downloaded installed the AsteriskWin32 PBX but it doesn't have sip server functionality . Can you please propose for an alternative to be used on the MS Windows client as external sip server for my Asterisk on CentOS ? Thank you ___ -- 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 -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- 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] Can't load cdr_radius.so module?
I'm not familiar with cdr_radius, but is there a debugging option? Anything in /var/log/messages? On Thu, Dec 24, 2009 at 1:35 AM, Zhang Shukun bit...@gmail.com wrote: Thank you ! i have load cdr_radius.so successfully! but another error occur. -- Executing [4...@tutorial:1] Dial(SIP/ivan-0a07dc80, SIP/test) in new stack -- Called test -- SIP/test-0a08b0f0 is ringing -- SIP/test-0a08b0f0 answered SIP/ivan-0a07dc80 -- Packet2Packet bridging SIP/ivan-0a07dc80 and SIP/test-0a08b0f0 [Dec 24 09:30:32] ERROR[10747]: cdr_radius.c:227 radius_log: Failed to record Radius CDR record! == Spawn extension (tutorial, 4321, 1) exited non-zero on 'SIP/ivan-0a07dc80' it says Failed to record Radius CDR record. Could you tell me , what's wrong with it? 2009/12/23 Olle E. Johansson o...@edvina.net: 23 dec 2009 kl. 11.25 skrev David Cunningham: Shukun, It tells you No such file or directory. Is the file in your modules directory? Actually, to be more specific. The module cdr_radius.so exists, but can't bind to the radius library libradiusclient-ng.so.2. Check LD_LIBRARY_PATH /O On Wed, Dec 23, 2009 at 10:09 AM, Zhang Shukun bit...@gmail.com wrote: hi , all when i do the command module load cdr_radius.so ,error happens. i have installed radiusclient-ng , what's wrong with it? thanks! error message as follow: ZHANGSHUKUN*CLI module load cdr_radius.so Unable to load module cdr_radius.so Command 'module load cdr_radius.so' failed. [Dec 23 17:55:41] WARNING[31072]: loader.c:380 load_dynamic_module: Error loading module 'cdr_radius.so': libradiusclient-ng.so.2: cannot open shared object file: No such file or directory [Dec 23 17:55:41] WARNING[31072]: loader.c:730 load_resource: Module 'cdr_radius.so' could not be loaded. -- Regards, Sucan ___ -- 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 -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- 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 --- o...@edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today! ___ -- 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 -- Regards, Sucan ___ -- 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 -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- 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] Inquiry:Connect my Asterisk to external sip?
It looks to me like calls from your Dial will route back to the sip-outgoing context and Dial again... it's loop. You'd really need to provide more logging information to advise further. On Thu, Dec 24, 2009 at 5:16 AM, hadi motamedi motamed...@gmail.com wrote: On Wed, Dec 23, 2009 at 1:55 PM, David Cunningham dcunning...@voisonics.com wrote: AsteriskWin32 does have SIP server functionality, same as the linux version. I can't think of any reason why having your CentOS Asterisk be both client and server and register with itself wouldn't work. Although I am wondering how much help all this will be in debugging a connection problem to another SIP provider... On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi motamed...@gmail.comwrote: On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham dcunning...@voisonics.com wrote: Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using sip set debug on might help you with the Host '192.168.0.139' does not implement 'REGISTER' problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi motamed...@gmail.comwrote: On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.comwrote: On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @ 192.168.0.139 . To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.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 Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you 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 -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- 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 I downloaded installed the AsteriskWin32 PBX but it doesn't have sip server functionality . Can you please propose for an alternative to be used on the MS Windows client as external sip server for my Asterisk on CentOS ? Thank you ___ -- 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 -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll
Re: [asterisk-users] 1.6 Troubleshooting help
It looks like whatever is being transmitted, or the response, isn't getting through. Possibly due to NAT or a firewall? It would help if you described the scenario where this is occurring. On Thu, Dec 24, 2009 at 7:18 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:13] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:14] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:15] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:16] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:19] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:20] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:22] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:24] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:26] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:28] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:30] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:32] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:35] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. ___ -- 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 -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- 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] Really Silly Question From Total Newbie
___ -- 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 -- David Cunningham Voisonics Limited IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 -- _ -- 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] How to use AGI php script function $agi - exec_dial
You might find this helpful: http://www.voip-info.org/wiki/view/Asterisk+AGI+php Regards, On Mon, Jan 11, 2010 at 2:19 AM, Zhang Shukun bit...@gmail.com wrote: hi, i want to use $agi - exec_dial() to dial . this is in extention.conf [tutorial] exten = 1234,1,Dial(SIP/ivan) is that i use $agi - exec_dial(SIP,tutorial|1234|1) can dial ? BTW, i want to know some turorial on how to use PHPAGI funtions? can you tell me some? Thanks! -- Best regards, Sucan -- _ -- 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 -- David Cunningham Voisonics Limited IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 -- _ -- 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] One way audio problem, a=sendonly and a re-invite
Hello all, I have a problem where problem with one way audio, and I think it's related to a=sendonly and a re-invite. Can anyone please assist? The scenario is as follows - We send an INVITE to a peer, and it replies with a 100 Trying, and then a 183 Session Progress message containing a=sendonly. - Asterisk plays the caller music on hold, which I believe is correct if we have an a=sendonly. - Then the peer sends a 200 OK which also has a=sendonly, and then sends a re-invite which I've copied and pasted below. - We have canreinvite=no set in sip.conf, but I'm not sure if we should be rejecting this re-invite or not because it does contain a=sendrecv. If it should be rejected what error should Asterisk return, and how can we establish two way audio? - After this re-invite Asterisk replies with a 100 Trying and then a 200 OK which contains a=recvonly. - Call is established but called party cannot hear caller. Here's the re-invite message - note that Asterisk is on port 5070: U 2010/05/05 12:47:38.139701 (peer):5060 - (asterisk):5070 INVITE sip:(called number)@(asterisk):5070 SIP/2.0. Via: SIP/2.0/UDP (peer):5060;branch=z9hG4bK2sansay7330954rdb6594. To: User sip:(called number)@(asterisk):5070;tag=as3ddcc528. From: sip:(called number)@(peer):5060;tag=sansay7330954rdb6594. Call-ID: 58eb52aa414c5e465c3c1a15603093fb@(asterisk). CSeq: 2 INVITE. Contact: sip:(called number)@(peer):5060. Max-Forwards: 69. Content-Type: application/sdp. Content-Length: 297. . v=0. o=Sansay-VSXi 188 1 IN IP4 (peer). s=Session Controller. c=IN IP4 (other unknown IP, maybe of called number?). t=0 0. m=audio 6932 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. Any help would be much appreciated! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] What does Asterisk give to reject a re-invite?
Hello, If you have canreinvite=no and a peer sends you a re-invite, what will Asterisk reply with? Thanks, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] What does Asterisk give to reject a re-invite?
Kevin, Thank you for that reply! We're having an issue where a peer's response to an INVITE includes a=sendonly. Later it sends a re-invite with a=sendrecv, however Asterisk responds to that with an OK that includes a=recvonly. The end result is the called party can't hear the caller. Do you have any idea why this is, or where I could go for more information? Thanks for the help. On Thu, May 13, 2010 at 11:06 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/13/2010 01:41 PM, David Cunningham wrote: If you have canreinvite=no and a peer sends you a re-invite, what will Asterisk reply with? It will accept it. 'canreinvite' is mis-named, and that's why in more modern versions of Asterisk it has been renamed to 'directmedia'. Asterisk will *always* accept properly formed re-INVITEs that don't require capabilities that are not available, and it will also generate them for non-directmedia purposes (like switching to and from T.38) when necessary, regardless of whether 'canreinvite' is set to yes or no. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] What does Asterisk give to reject a re-invite?
Hi Kevin, We don't have mohinterpret set at all, so I think it uses default. Is there anything else you can suggest? Any other places to go for help? Thanks for your assistance! On Thu, May 13, 2010 at 11:32 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/13/2010 05:16 PM, David Cunningham wrote: We're having an issue where a peer's response to an INVITE includes a=sendonly. Later it sends a re-invite with a=sendrecv, however Asterisk responds to that with an OK that includes a=recvonly. The end result is the called party can't hear the caller. Do you have any idea why this is, or where I could go for more information? That would seem to indicate that the peer is placing Asterisk 'on hold', and then taking it back 'off hold' later. I do not know why Asterisk would respond with 'recvonly', it should only do that when it thinks the channel is still on hold. Are you using 'mohinterpret=passthrough', where Asterisk would send the hold indication to the bridged channel instead of reacting to it locally? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Cause and cure for Exceptionally long voice queue length queuing to Local?
Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come across is that the peer is congested and sending RTCP messages asking us to slow the RTP down. Is there any way we can verify this? [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412...@asterisk-phone-7e3d;1 Thanks in advance! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Cause and cure for Exceptionally long voice queuelength queuing to Local?
What should I expect see if it is the peer asking us to slow down RTP? Thanks again. On Wed, May 19, 2010 at 9:05 PM, Danny Nicholas da...@debsinc.com wrote: Sip debug peer? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Cunningham Sent: Wednesday, May 19, 2010 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cause and cure for Exceptionally long voice queuelength queuing to Local? Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come across is that the peer is congested and sending RTCP messages asking us to slow the RTP down. Is there any way we can verify this? [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412...@asterisk-phone-7e3d;1 Thanks in advance! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Cause and cure for Exceptionally long voice queuelength queuing to Local?
I don't see anything in the SIP trace related to the warning messages. Would anyone have any further tips? Thanks for any help! On Wed, May 19, 2010 at 9:12 PM, David Cunningham dcunning...@voisonics.com wrote: What should I expect see if it is the peer asking us to slow down RTP? Thanks again. On Wed, May 19, 2010 at 9:05 PM, Danny Nicholas da...@debsinc.com wrote: Sip debug peer? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Cunningham Sent: Wednesday, May 19, 2010 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cause and cure for Exceptionally long voice queuelength queuing to Local? Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come across is that the peer is congested and sending RTCP messages asking us to slow the RTP down. Is there any way we can verify this? [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412...@asterisk-phone-7e3d;1 Thanks in advance! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Cause and cure for Exceptionally long voice queue length queuing to Local?
Leif - thank you! Will try that. On Fri, May 21, 2010 at 12:19 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: David Cunningham wrote: Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come across is that the peer is congested and sending RTCP messages asking us to slow the RTP down. Is there any way we can verify this? [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412...@asterisk-phone-7e3d;1 Thanks in advance! Try upgrading to 1.6.1.13. You're using a version of Asterisk from early December 2009. Doing a search for closed issues on the Asterisk issue tracker at https://issues.asterisk.org caused me to find bug 15609 (https://issues.asterisk.org/view.php?id=15609) which was committed on December 30, 2009. On January 11, 2010, Asterisk version 1.6.1.13-rc1 was created which contains the commit from December 30, 2009, and was subsequently released as 1.6.1.13 on January 15, 2010. The ChangeLog showing the commit is here: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.13 The release is available here: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Recording maximum time and stop on silence
All, Two questions: 1. Is there a limit on how long a call can be recorded for? For example is 4 hours a problem? 2. Can recording be stopped after a configured period of silence? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Recording maximum time and stop on silence
Danny, thank you! On Wed, Sep 22, 2010 at 10:31 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham *Sent:* Wednesday, September 22, 2010 4:28 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Recording maximum time and stop on silence All, Two questions: 1. Is there a limit on how long a call can be recorded for? For example is 4 hours a problem? 2. Can recording be stopped after a configured period of silence? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 AFAIK, #1 is limited only by available disk space, #2 is yes, but you may have to tweak some settings to “get it right” -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] 'Bookmarking' a place in a sound file
Hi all, Is it possible to somehow 'bookmark' a place in a sound file? That is, the user presses a key while a sound file is playing and that point is saved, and some time in the future we can play the same sound file and tell it to start playing from that point. This would be done within a perl AGI program. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] 'Bookmarking' a place in a sound file
Steve, that looks just the job, thank you very much. On Wed, Dec 8, 2010 at 2:32 AM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 7 Dec 2010, David Cunningham wrote: Is it possible to somehow 'bookmark' a place in a sound file? That is, the user presses a key while a sound file is playing and that point is saved, and some time in the future we can play the same sound file and tell it to start playing from that point. This would be done within a perl AGI program. The AGI command 'stream file' will return 'endpos' when interrupted with a keypress. You could then save that in a channel variable or a database. A subsequent call to 'stream file' would include 'endpos' as the 'sample offset.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] progressinband, how much extra CPU load?
Hi everyone, We have an Asterisk 1.4.17 user who has problems with sometimes not getting a ring tone on the calling phone. We're considering setting progressinband = yes, but would like to know how much extra CPU load this will require? If anyone can give something even roughly specific (eg 30% increase) that would be great, rather than just lots. Also, are there any ATAs which are known to not work with progressinband = yes? We have Polycom, Linksys and Audiocode. Thanks for any advice, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Introducing easySysAdmin - automated security and telecom fraud protection
Hello all, Voisonics is pleased to introduce easySysAdmin, an automated support/security platform, designed to save your engineer's time and prevent hacking attempts and telecom fraud. It comprises of an online service run by us, and a lightweight and easy-to-install client on your side. Specifically of interest to Asterisk users is the monitoring of SIP registrations, and automatic blocking of repeated failed attempts. This is done though a local application on your server so your call data is totally secure. In addition, blocked IP addresses are shared via the service so you can pre-emptively block addresses that others experienced attacks from. We offer a FREE one week no obligation trial, and during the month of January we also offer a DISCOUNT for Asterisk users. Just enter voucher asterisk01 at the checkout to receive 30% off. Protection is available from just US$84 per server with the discount. You can read more and give the free trial a go at: http://easysysadmin.com/ Our standard security package includes: - Monitor VoIP traffic (SIP registrations) and block attackers. - Watch remote server access (SSH logins) and block attackers. - Detect spam relay attempts (SMTP) and block attackers. - Scan of network ports to find vulnerabilities. - Check of software for vendor (distribution) security updates. - Custom monitoring of any TCP port or log file you want. - Flexible configuration to set warn and blocking levels. The background to this service is that prior to founding Voisonics I worked with IBM for 10 years, and became responsible for the security planning and audit compliance of many of IBM's voicemail and IVR platforms across the world. Using this experience and knowledge of their standards we have created easySysAdmin. If you have any questions please don't hesitate to contact me directly. Regards, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Error compiling Dahdi: invalid use of undefined type struct module
/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:5945: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:5951: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:5966: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:5976: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6011: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6018: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_unregister’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6049: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6058: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6062: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6092: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘process_echocan_events’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:7092: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:7102: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘__putbuf_chunk’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:7594: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:7668: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:7810: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_poll’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8082: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘coretimer_func’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8448: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_receive’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8559: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_init’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8712: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8722: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8723: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_cleanup’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8752: error: invalid use of undefined type ‘struct module’ make[2]: *** [/usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.o] Erreur 1 make[1]: *** [_module_/usr/src/dahdi-linux-2.4.0/drivers/dahdi] Erreur 2 make[1]: quittant le répertoire « /usr/src/linux-headers-2.6.34.6 » make: *** [modules] Erreur 2 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Error compiling Dahdi: invalid use of undefined type struct module
Shaun, CONFIG_MODULES wasn't enabled - thanks for the advice! On Mon, Jan 31, 2011 at 4:02 PM, Shaun Ruffell sruff...@digium.com wrote: On 1/30/11 8:45 PM, David Cunningham wrote: I'm installing Asterisk with Dahdi on a server with a custom kernel compile. I've got the kernel source in /lib/modules/2.6.34.6--grs-ipv6-64/build which points to /usr/src/linux-headers-2.6.34.6 and I think that's fine, but am getting all these struct module errors. Can anyone advise? Thanks! # make make -C drivers/dahdi/firmware firmware-loaders make[1]: entrant dans le répertoire « /usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware » make[1]: quittant le répertoire « /usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware » make -C /lib/modules/2.6.34.6--grs-ipv6-64/build SUBDIRS=/usr/src/dahdi-linux-2.4.0/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.4.0/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: entrant dans le répertoire « /usr/src/linux-headers-2.6.34.6 » CC [M] /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.o /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_register_tone_zone’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:1440: error: invalid use of undefined type ‘struct module’ Normally this is the result of not having CONFIG_MODULES set in your kernel config. This is set when you check Enable loadable module support on the top level menu in menuconfig. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Higher CPU usage on 1.6.1 than 1.4?
Hello, We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now experiencing higher CPU utilization on their server. I can't see anything wrong, so is this just expected with 1.6? Can anyone help explain it? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Higher CPU usage on 1.6.1 than 1.4?
Jared, Thank you for that information! Has anyone else had an experience like this? On 12 May 2011 20:25, Jared Geiger compuw...@gmail.com wrote: Hi David, When I was testing 1.6.1 for high volume channels, I couldn't get over 1000 channels / 40 CPS without the load average spiking up due to io wait. I switched back to 1.4 and I can go to 3000 channels / 75 CPS with no io wait and a load average in the 1s. It seemed like it was caused by the new timing system in 1.6.1 even though I wasn't proxying media using only SIP. I haven't tried 1.8 yet to see if it handles large call volumes any better. ~Jared On Wed, May 11, 2011 at 8:29 PM, David Cunningham dcunning...@voisonics.com wrote: Hello, We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now experiencing higher CPU utilization on their server. I can't see anything wrong, so is this just expected with 1.6? Can anyone help explain it? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Change to pickups in Asterisk 1.8 - not working on local channels?
Hello all,
We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups
which previously worked fine have stopped working.
Can anyone advise if there has been a change in how pickups work?
Here is an example where 1000101 is trying to pick up a call to 1000103:
SIP/product-local-0005AGI Rx EXEC Dial
Local/1000103@product-pickup
/n,60,M(product-answered^0^1306286740.11)orL(360:6)
-- AGI Script Executing Application: (Dial) Options:
(Local/1000103@product-pickup
/n,60,M(product-answered^0^1306286740.11)orL(360:6))
Limit Data for this call:
timelimit = 360 ms (3600.000 s)
play_warning = 6 ms (60.000 s)
play_to_caller = yes
play_to_callee = no
warning_freq = 0 ms (0.000 s)
start_sound=
warning_sound = timeleft
end_sound =
-- Called 1000103@product-pickup/n
-- Executing [1000103@product-pickup:1]
Pickup(Local/1000103@product-pickup-db70;2, 1000103@product-phone) in
new stack
[May 25 11:25:40] NOTICE[1020]: app_directed_pickup.c:313 pickup_exec: No
target channel found for 1000103.
-- Auto fallthrough, channel 'Local/1000103@product-pickup-db70;2'
status is 'UNKNOWN'
The context doing the pickup looks like:
[product-pickup]
exten = _[0-9*#]!, 1, Pickup(${EXTEN}@product-phone)
Thanks for any advice,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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[asterisk-users] Voice recognition recommendations?
Hi all, We have a project involving voice recognition, and will need a vocabulary of 10,000 words (actually names). Can anyone recommend a product that works with Asterisk? Thanks, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Permanent sip and agi debug on?
Hi all, I can't find the answer to this via google - is there some way to permanently enable sip set debug on and agi set debug on in Asterisk? I want this to be automatically enabled even after restarts. Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Permanent sip and agi debug on?
Kevin, Thank you very much! On 10 November 2011 00:15, Kevin P. Fleming kpflem...@digium.com wrote: On 11/09/2011 04:22 AM, David Cunningham wrote: Hi all, I can't find the answer to this via google - is there some way to permanently enable sip set debug on and agi set debug on in Asterisk? I want this to be automatically enabled even after restarts. In recent versions of Asterisk, you can put CLI commands into cli.conf and they will be run automatically when Asterisk starts. There are even examples of doing this for 'sip set debug' in cli.conf.sample :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Continue AGI after Dial() following caller hang up?
Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Continue AGI after Dial() following caller hang up?
Thorsten, We have SIGHUP set to 'IGNORE', but it still does not continue the AGI after the Dial(). Do you have any idea why that might happen? Thanks for your advice. On 21 November 2011 22:19, Thorsten Göllner t...@ovm-group.com wrote: If the caller hangs up Asterisk sends a SIGHUP. You can catch the signal and do whatever you want to do. Am 21.11.2011 07:38, schrieb David Cunningham: Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Continue AGI after Dial() following caller hang up?
Kingsley, Thanks for the reply, but I am looking to continue within the same AGI process and I believe that method would require starting a new AGI. On 21 November 2011 22:22, Kingsley Tart kings...@skymarket.co.uk wrote: We do that with the F option in Dial(). From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial : F(context^exten^pri): When the caller hangs up, transfer the called party to the specified context and extension and continue execution. Cheers, Kingsley. On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote: Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Continue AGI after Dial() following caller hang up?
The strange thing is that we are using fast AGI, and for some reason the
AGI always exits when the caller hangs up - even when I set HUP to IGNORE.
If I set HUP to a subroutine that just logs a message, that message is
never logged.
Thanks for all the help.
On 22 November 2011 05:23, Kingsley Tart kings...@skymarket.co.uk wrote:
Yeah fastAGI is great, I've been using it for a while for performance
reasons but yes I guess it would solve problems like this too.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 08:34 -0600, Danny Nicholas wrote:
Just offhand, I think you should utilize the FastAGI protocol, since it
doesn't seem to live or die based on when the call hangs up. Otherwise,
the
$SIG{'HUP'} = 'IGNORE';
Statement will separate the process so it doesn't die on a hangup.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley
Tart
Sent: Monday, November 21, 2011 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Continue AGI after Dial() following caller
hang up?
Yeah I think I slightly misread your original question, which I realised
when I saw Thorsten's reply. I initially thought you just wanted to avoid
going into the h extension.
I'm not doing any AGI stuff here that hangs around while the call does
stuff
- the AGI process just runs quickly then quits, returning control back to
the dialplan. I had incorrectly assumed you were doing the same.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote:
Kingsley,
Thanks for the reply, but I am looking to continue within the same AGI
process and I believe that method would require starting a new AGI.
On 21 November 2011 22:22, Kingsley Tart kings...@skymarket.co.uk
wrote:
We do that with the F option in Dial().
From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial :
F(context^exten^pri): When the caller hangs up, transfer the
called
party to the specified context and extension and continue
execution.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote:
Hello,
We would like to continue a Perl AGI after a Dial() it has
done
completes following caller hangup. We would like to do this
in the
same AGI, and not using a new AGI from the 'h' extension. It
works
fine when the called party hangs up and the 'g' option is
used, but
not for caller hangup.
Is this possible?
If not a confirmation that this is the case would be very
helpful.
Thanks for any advice!
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
--
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Thurs:
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US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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--
Cheers,
Kingsley.
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Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
Kingsley,
We have the same - the daemon forks child processes to handle individual
calls.
We need the fastAGI to continue so it can take some further action
recording details of the call. This could be done using the 'h' extension,
but it would be nice to avoid this method for simplicity sake. It does
appear that some people can continue after the Dial and we can't for some
reason.
On 22 November 2011 21:21, Kingsley Tart kings...@skymarket.co.uk wrote:
When something makes a socket connection to your fastAGI daemon, does
your daemon fork a child process to deal with that connection, or handle
it in the main process?
I've set ours up to fork a child process and detach itself from the
parent socket. When it ends, the child exits (which is what we want) and
the parent stays running (which is also what we want).
Is there any particular reason you want your fastAGI instance to persist
for the duration of the call?
Cheers,
Kingsley.
On Tue, 2011-11-22 at 14:27 +1100, David Cunningham wrote:
The strange thing is that we are using fast AGI, and for some reason
the AGI always exits when the caller hangs up - even when I set HUP to
IGNORE. If I set HUP to a subroutine that just logs a message, that
message is never logged.
Thanks for all the help.
On 22 November 2011 05:23, Kingsley Tart kings...@skymarket.co.uk
wrote:
Yeah fastAGI is great, I've been using it for a while for
performance
reasons but yes I guess it would solve problems like this too.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 08:34 -0600, Danny Nicholas wrote:
Just offhand, I think you should utilize the FastAGI
protocol, since it
doesn't seem to live or die based on when the call hangs up.
Otherwise,
the
$SIG{'HUP'} = 'IGNORE';
Statement will separate the process so it doesn't die on a
hangup.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of Kingsley Tart
Sent: Monday, November 21, 2011 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Continue AGI after Dial()
following caller
hang up?
Yeah I think I slightly misread your original question,
which I realised
when I saw Thorsten's reply. I initially thought you just
wanted to avoid
going into the h extension.
I'm not doing any AGI stuff here that hangs around while the
call does stuff
- the AGI process just runs quickly then quits, returning
control back to
the dialplan. I had incorrectly assumed you were doing the
same.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote:
Kingsley,
Thanks for the reply, but I am looking to continue within
the same AGI
process and I believe that method would require starting a
new AGI.
On 21 November 2011 22:22, Kingsley Tart
kings...@skymarket.co.uk
wrote:
We do that with the F option in Dial().
From http://www.voip-info.org/wiki/view/Asterisk
+cmd+Dial :
F(context^exten^pri): When the caller hangs up,
transfer the
called
party to the specified context and extension and
continue
execution.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 17:38 +1100, David
Cunningham wrote:
Hello,
We would like to continue a Perl AGI after a
Dial() it has
done
completes following caller hangup. We would like
to do this
in the
same AGI, and not using a new AGI from the 'h'
extension. It
works
fine when the called party hangs up and the 'g'
option is
used, but
not for caller hangup.
Is this possible?
If not a confirmation that this is the case
would be very
helpful.
Thanks for any advice!
--
David Cunningham, Voisonics
http://voisonics.com
[asterisk-users] SendFAX timestamp
Hello, Does SendFAX have the ability to put the caller ID and timestamp on the fax? If so, is there a way to adjust the timezone used for the timestamp? Thanks for any assistance. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] FastAGI script and DIAL execution
is ringing
-- SIP/193.17.66.71-004d is making progress passing it to
SIP/139255423-004c
-- SIP/193.17.66.71-004d answered SIP/139255423-004c
-- Executing [h@customer:1] Set(SIP/139255423-004c,
CDR(q931)=16) in new stack
-- Executing [h@customer:2] Set(SIP/139255423-004c,
CDR(userfield)={agi:,a-leg-id:2118d872-305e-4bb4-8c47-30e1514cb934,b-leg-id:36b232e73ac326bd0407b1594627c589@y.y.y.y:5060})
in new stack
SIP/139255423-004cAGI Tx 200 result=-1
SIP/139255423-004cAGI Tx HANGUP
SIP/139255423-004cAGI Rx GET VARIABLE HANGUPCAUSE
SIP/139255423-004cAGI Tx 200 result=1 (16)
SIP/139255423-004cAGI Rx GET VARIABLE Q16
SIP/139255423-004cAGI Tx 200 result=1 (0)
SIP/139255423-004cAGI Rx SET VARIABLE AJ_AGISTATUS SUCCESS
SIP/139255423-004cAGI Tx 200 result=1
-- SIP/139255423-004cAGI Script agi://localhost/auth completed,
returning 4
SIP/139255423-004cAGI Tx HANGUP
== Spawn extension (customer, 003462999, 8) exited non-zero on
'SIP/139255423-004c'
--
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Re: [asterisk-users] SendFAX timestamp
Steve, Thanks for the reply. Would anyone else know if Asterisk allows use of SpanDSP's time zone conversion? On 27 June 2012 00:24, Steve Underwood ste...@coppice.org wrote: On 06/26/2012 10:24 AM, David Cunningham wrote: Hello, Does SendFAX have the ability to put the caller ID and timestamp on the fax? If so, is there a way to adjust the timezone used for the timestamp? Thanks for any assistance. SpanDSP has that ability, including per instance time zones, but I don't know if the Asterisk module exposes that facility. Steve -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk as a translating proxy only?
Hello, We want to use Asterisk as a proxy to translate between Skinny/SCCP and SIP, with as little as possible work required in between. Does Asterisk have a way for custom programs to read and write raw packets? If we can get the input data in a readable format and output it in the required format, that may do the trick. Thank you for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] faxdetect on/off on the fly?
Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] faxdetect on/off on the fly?
Hi Steve, We have all calls going to an AGI, which decides where the number will get routed to, and if fax detection should be enabled for this call. The choice should only apply to the current call. Thanks very much. On 3 January 2013 17:46, Steve Edwards asterisk@sedwards.com wrote: On Thu, 3 Jan 2013, David Cunningham wrote: We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. What's the 'use case?' You're going to call in and execute an AGI that will enable faxdetect for future calls to this channel or other channels? Should the 'change' survive an Asterisk restart or an OS reboot? -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] faxdetect on/off on the fly?
Hi Danny, Can you please elaborate on how in the dialplan we can set faxdetect on and off? We currently have it set on in sip.conf. Thanks. On 3 January 2013 17:21, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham *Sent:* Thursday, January 03, 2013 3:13 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] faxdetect on/off on the fly? ** ** Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. You should be able to call the AGI and set a dialplan variable and use Gotoif to do/not do faxdetect. Reading the .sample files for 11.0 it seems that normally these are “configured until restart/reload” but with a little testing, the default should be overrideable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk 1.8 drop calls after 15 minutes
If it's a reinvite problem, check the domain in the request URI. Might be something it shouldn't be, eg something to do with the VPN. On 11 September 2013 05:28, isr...@gmail.com wrote: Some providers send a reinvite after 15 min and if asterisk doesn't respond will disconnect the call Maybe playaround with canreinvite --Original Message-- From: Jeremy Kister Sender: asterisk-users-boun...@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes Sent: Sep 10, 2013 10:23 PM On 9/10/2013 7:05 AM, Administrator TOOTAI wrote: I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk Just for kicks, I would disable session-timers to see if the problem goes away. in the general section and/or each peer in sip.conf: session-timers=refuse -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk not receiving call from VPN address
Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers. Thanks in advance for any help. The ngrep on the Asterisk server: U 2014/01/17 13:15:15.599557 172.x.x.x:5060 - 103.y.y.y:5060 INVITE sip:9067268@103.y.y.y:5060;transport=udp SIP/2.0. Record-Route: sip:172.x.x.x;lr=on. Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.f49ceb73.0. Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;branch=z9hG4bK806710997. From: 9067271 sip:9067271@172.x.x.x;tag=198791249. To: sip:9067268@172.x.x.x. Call-ID: 1905625787@192.z.z.z. ... 172.x.x.x is the Kamailio server's VPN address 103.y.y.y is the Asterisk server's real address 192.z.z.z is the calling phone's LAN address -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk not receiving call from VPN address
Hi Paul, The ngrep on the Asterisk server does show it being received. Have you any idea what would prevent it getting from the network stack to Asterisk on that machine? On 21 January 2014 05:30, Paul Belanger paul.belan...@polybeacon.comwrote: On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham dcunning...@voisonics.com wrote: Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers. Thanks in advance for any help. The ngrep on the Asterisk server: U 2014/01/17 13:15:15.599557 172.x.x.x:5060 - 103.y.y.y:5060 INVITE sip:9067268@103.y.y.y:5060;transport=udp SIP/2.0. Record-Route: sip:172.x.x.x;lr=on. Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.f49ceb73.0. Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;branch=z9hG4bK806710997. From: 9067271 sip:9067271@172.x.x.x;tag=198791249. To: sip:9067268@172.x.x.x. Call-ID: 1905625787@192.z.z.z. ... 172.x.x.x is the Kamailio server's VPN address 103.y.y.y is the Asterisk server's real address 192.z.z.z is the calling phone's LAN address Sounds like a routing problem opposed to an application issue. You'll have to fire up tcpdump on Kamailio and see what happens to the packet. The look at the local routing tables to see where it is getting routed. If Asterisk is not receiving the patch, then Kamailio is not routing it properly. You'll be able to see everything once you have a pcap of the call. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk not receiving call from VPN address
Hi Duncan, The Asterisk machine also has a VPN IP address, so it has a route for 172.x addresses to go to tun0 VPN interface. On 21 January 2014 08:30, Duncan Turnbull dun...@e-simple.co.nz wrote: On 21/01/2014, at 10:24 am, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, The ngrep on the Asterisk server does show it being received. Have you any idea what would prevent it getting from the network stack to Asterisk on that machine? Have you got a static route on asterisk or your default gateway showing how to get back to the 172. addresses i.e. pointing to the vpn box for 172 addresses? Cheers Duncan On 21 January 2014 05:30, Paul Belanger paul.belan...@polybeacon.comwrote: On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham dcunning...@voisonics.com wrote: Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers. Thanks in advance for any help. The ngrep on the Asterisk server: U 2014/01/17 13:15:15.599557 172.x.x.x:5060 - 103.y.y.y:5060 INVITE sip:9067268@103.y.y.y:5060;transport=udp SIP/2.0. Record-Route: sip:172.x.x.x;lr=on. Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.f49ceb73.0. Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;branch=z9hG4bK806710997. From: 9067271 sip:9067271@172.x.x.x;tag=198791249. To: sip:9067268@172.x.x.x. Call-ID: 1905625787@192.z.z.z. ... 172.x.x.x is the Kamailio server's VPN address 103.y.y.y is the Asterisk server's real address 192.z.z.z is the calling phone's LAN address Sounds like a routing problem opposed to an application issue. You'll have to fire up tcpdump on Kamailio and see what happens to the packet. The look at the local routing tables to see where it is getting routed. If Asterisk is not receiving the patch, then Kamailio is not routing it properly. You'll be able to see everything once you have a pcap of the call. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk not receiving call from VPN address
Hi Eric, Thanks for the suggestion. It was on bindaddr of 0.0.0.0, but we tried removing that too, and Asterisk still doesn't see anything. On 21 January 2014 09:18, Eric Wieling ewiel...@nyigc.com wrote: Make sure you do NOT have any *bindaddr options set in your sip.conf. If you do, you are telling Asterisk to not allow the OS to pick the source IP and hence the routing. The *bindaddr options are seldom useful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of David Cunningham Sent: Monday, January 20, 2014 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk not receiving call from VPN address Hi Duncan, The Asterisk machine also has a VPN IP address, so it has a route for 172.x addresses to go to tun0 VPN interface. On 21 January 2014 08:30, Duncan Turnbull dun...@e-simple.co.nz wrote: On 21/01/2014, at 10:24 am, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, The ngrep on the Asterisk server does show it being received. Have you any idea what would prevent it getting from the network stack to Asterisk on that machine? Have you got a static route on asterisk or your default gateway showing how to get back to the 172. addresses i.e. pointing to the vpn box for 172 addresses? Cheers Duncan On 21 January 2014 05:30, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham dcunning...@voisonics.com wrote: Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers. Thanks in advance for any help. The ngrep on the Asterisk server: U 2014/01/17 13:15:15.599557 172tel:15.599557%20172 .x.x.x:5060 - 103.y.y.y:5060 INVITE sip:9067268@103.y.y.y:5060;transport=udp SIP/2.0. Record-Route: sip:172.x.x.x;lr=on. Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.f49ceb73.0. Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;branch=z9hG4bK806710997. From: 9067271 sip:9067271@172.x.x.x ;tag=198791249. To: sip:9067268@172.x.x.x. Call-ID: 1905625787@192.z.z.z. ... 172.x.x.x is the Kamailio server's VPN address 103.y.y.y is the Asterisk server's real address 192.z.z.z is the calling phone's LAN address Sounds like a routing problem opposed to an application issue. You'll have to fire up tcpdump on Kamailio and see what happens to the packet. The look at the local routing tables to see where it is getting routed. If Asterisk is not receiving the patch, then Kamailio is not routing it properly. You'll be able to see everything once you have a pcap of the call
Re: [asterisk-users] Asterisk not receiving call from VPN address
Hi Paul, Using ngrep/tcpdump shows the packet clearly going from the Kamailio server and arriving at the Asterisk server. This is why it's a mystery that Asterisk doesn't see the call coming in. We tried removing the firewall (so iptables -L shows no rules at all) but that didn't help unfortunately. On 21 January 2014 15:29, Paul Belanger paul.belan...@polybeacon.comwrote: On Mon, Jan 20, 2014 at 4:24 PM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, The ngrep on the Asterisk server does show it being received. Have you any idea what would prevent it getting from the network stack to Asterisk on that machine? Well, you need to use tcpdump on each hop across your network. If are Asterisk is not getting anything, either it is not receiving anything (check transmit side) or the firewall is dropping it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk not receiving call from VPN address
Hi Duncan, Thank you for your reply. Here's the netstat: [root]# netstat -udpln | grep asterisk udp0 0 0.0.0.0:50000.0.0.0:* 6672/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 6672/asterisk udp0 0 0.0.0.0:50600.0.0.0:* 6672/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 6672/asterisk Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Kamailio server: 17:13:17.103771 IP 103.x.x.x.5060 172.y.y.y.5060: SIP, length: 1228 E...@./g.v.INVITE sip:*1@172.y.y.y:5060;transport=udp SIP/2.0 Record-Route: sip:103.x.x.x;lr=on Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0 Via: SIP/2.0/UDP 192.168.1.40:5060 ;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850 From: sip:9067273@103.x.x.x;tag=1880695235 To: sip:*1@103.x.x.x Call-ID: 1898224288 Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Asterisk server: 17:13:17.093676 IP 103.x.x.x.5060 172.y.y.y.5060: SIP, length: 1228 E...?.?/g.v.INVITE sip:*1@172.y.y.y:5060;transport=udp SIP/2.0 Record-Route: sip:103.x.x.x;lr=on Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0 Via: SIP/2.0/UDP 192.168.1.40:5060 ;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850 From: sip:9067273@103.x.x.x;tag=1880695235 To: sip:*1@103.x.x.x Call-ID: 1898224288 On 21 January 2014 16:56, Duncan Turnbull dun...@e-simple.co.nz wrote: On 21/01/2014, at 6:40 pm, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, Using ngrep/tcpdump shows the packet clearly going from the Kamailio server and arriving at the Asterisk server. This is why it's a mystery that Asterisk doesn't see the call coming in. We tried removing the firewall (so iptables -L shows no rules at all) but that didn't help unfortunately. Can you show a packet dump of the SIP invites arriving at the asterisk PBX , mostly just confirming the ip address that the server is receiving packets on *root@zespri*:*~*# tcpdump udp port 5060 -A -n tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes 18:52:23.063862 IP 192.168.51.7.5060 27.111.14.65.5060: SIP, length: 534 E`.2.L..@.3..o.A..u.OPTIONS sip:sip.2talk.co.nz SIP/2.0 Via: SIP/2.0/UDP 192.168.51.7:5060;branch=z9hG4bK45a08b58;rport Max-Forwards: 70 From: Unknown sip:049343953@192.168.51.7;tag=as32fe455a To: sip:sip.2talk.co.nz Contact: sip:0412345678@192.168.51.7:5060 Call-ID: 10c0242d16529fff78572ef91ef47237@192.168.51.7:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.6.1) Date: Tue, 21 Jan 2014 05:52:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 18:52:23.084330 IP 27.111.14.65.5060 192.168.51.7.5060: SIP, length: 472 E...9o.A..3...r.SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.51.7:5060 ;branch=z9hG4bK45a08b58;received=192.168.51.7;rport=5060 From: Unknown sip:049343953@192.168.51.7:5060;tag=as32fe455a To: sip:sip.2talk.co.nz;tag=as7b633145 Call-ID: 10c0242d16529fff78572ef91ef47237@192.168.51.7:5060 CSeq: 102 OPTIONS Server: 2talk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Accept: application/sdp Content-Length: 0 Also the udp ports asterisk is listening on e.g netstat -udpl Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name udp0 0 0.0.0.0:45200.0.0.0:* 1413/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 1413/asterisk udp0 0 0.0.0.0:50000.0.0.0:* 1413/asterisk udp0 0 0.0.0.0:50600.0.0.0:* 1413/asterisk On 21 January 2014 15:29, Paul Belanger paul.belan...@polybeacon.comwrote: On Mon, Jan 20, 2014 at 4:24 PM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, The ngrep on the Asterisk server does show it being received. Have you any idea what would prevent it getting from the network stack to Asterisk on that machine? Well, you need to use tcpdump on each hop across your network. If are Asterisk is not getting anything, either it is not receiving anything (check transmit side) or the firewall is dropping it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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
Re: [asterisk-users] Asterisk not receiving call from VPN address
Hi Larry, Thanks for the reply. We have all of those settings left out of our sip.conf, so this should allow everything, right? On 21 January 2014 17:38, Larry Moore lmo...@omninet.net.au wrote: Have you checked your localnet=, deny=, permit=, contactdeny= contactpermit= settings? My 2c worth. On 20/01/2014 10:51 AM, David Cunningham wrote: Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers. Thanks in advance for any help. The ngrep on the Asterisk server: U 2014/01/17 13:15:15.599557 172.x.x.x:5060 - 103.y.y.y:5060 INVITE sip:9067268@103.y.y.y:5060;transport=udp SIP/2.0. Record-Route: sip:172.x.x.x;lr=on. Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.f49ceb73.0. Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;branch=z9hG4bK806710997. From: 9067271 sip:9067271@172.x.x.x;tag=198791249. To: sip:9067268@172.x.x.x. Call-ID: 1905625787@192.z.z.z. ... 172.x.x.x is the Kamailio server's VPN address 103.y.y.y is the Asterisk server's real address 192.z.z.z is the calling phone's LAN address -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk not receiving call from VPN address
Hi Duncan, We have sip set debug on and nothing is shown, even though tcpdump/ngrep on the same server does. It's very strange. The output of ip address list is: [root]# ip address list 1: lo: LOOPBACK,UP,LOWER_UP mtu 16436 qdisc noqueue state UNKNOWN link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00 inet 127.0.0.1/8 scope host lo inet6 ::1/128 scope host valid_lft forever preferred_lft forever 2: eth0: BROADCAST,MULTICAST,UP,LOWER_UP mtu 1500 qdisc mq state UP qlen 1000 link/ether 44:1e:a1:4e:2f:b8 brd ff:ff:ff:ff:ff:ff inet 103.y.y.19/24 brd 103.y.y.255 scope global eth0 inet6 fe80::461e:a1ff:fe4e:2fb8/64 scope link valid_lft forever preferred_lft forever 3: eth1: BROADCAST,MULTICAST mtu 1500 qdisc noop state DOWN qlen 1000 link/ether 44:1e:a1:4e:2f:ba brd ff:ff:ff:ff:ff:ff 4: eth2: BROADCAST,MULTICAST mtu 1500 qdisc noop state DOWN qlen 1000 link/ether 44:1e:a1:4f:30:a4 brd ff:ff:ff:ff:ff:ff 5: eth3: BROADCAST,MULTICAST mtu 1500 qdisc noop state DOWN qlen 1000 link/ether 44:1e:a1:4f:30:a6 brd ff:ff:ff:ff:ff:ff 6: tun0: POINTOPOINT,MULTICAST,NOARP,UP,LOWER_UP mtu 1500 qdisc pfifo_fast state UNKNOWN qlen 100 link/[65534] inet 172.x.x.14 peer 172.x.x.13/32 scope global tun0 The output of netstat -rn is: [root]# netstat -rn Kernel IP routing table Destination Gateway Genmask Flags MSS Window irtt Iface 172.x.x.10 172.x.x.13 255.255.255.255 UGH 0 0 0 tun0 172.x.x.13 0.0.0.0 255.255.255.255 UH0 0 0 tun0 172.x.x.1 172.x.x.13 255.255.255.255 UGH 0 0 0 tun0 172.x.x.18 172.x.x.13 255.255.255.255 UGH 0 0 0 tun0 192.168.234.0 172.x.x.13 255.255.255.0 UG0 0 0 tun0 192.168.235.0 172.x.x.13 255.255.255.0 UG0 0 0 tun0 103.y.y.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0 169.z.z.0 0.0.0.0 255.255.0.0 U 0 0 0 eth0 172.21.0.0 172.x.x.13 255.255.0.0 UG0 0 0 tun0 10.0.0.0172.x.x.13 255.0.0.0 UG0 0 0 tun0 0.0.0.0 103.y.y.1 0.0.0.0 UG0 0 0 eth0 On 21 January 2014 17:44, Duncan Turnbull dun...@e-simple.co.nz wrote: Cool That looks like it is arriving at Asterisk - are you sure asterisk is not getting it? If you turn on sip debug in asterisk can you see the SIP packets? It maybe asterisk is ignoring them or replying to them but its going out an interface you hadn’t thought of, I have had that a few times. I should have mentioned to print out your route table and ifconfig. Asterisk can reply on a different address to the original destination especially if it came through a tunnel. Often it will be the tunnel interface address. Usually then we set the secondary address as the outbound proxy on the phone so the phone will also respond to it. Cheers Duncan On 21/01/2014, at 7:18 pm, David Cunningham dcunning...@voisonics.com wrote: Hi Duncan, Thank you for your reply. Here's the netstat: [root]# netstat -udpln | grep asterisk udp0 0 0.0.0.0:50000.0.0.0:* 6672/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 6672/asterisk udp0 0 0.0.0.0:50600.0.0.0:* 6672/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 6672/asterisk Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Kamailio server: 17:13:17.103771 IP 103.x.x.x.5060 172.y.y.y.5060: SIP, length: 1228 E...@./g.v.INVITE sip:*1@172.y.y.y:5060;transport=udpSIP/2.0 Record-Route: sip:103.x.x.x;lr=on Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0 Via: SIP/2.0/UDP 192.168.1.40:5060 ;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850 From: sip:9067273@103.x.x.x;tag=1880695235 To: sip:*1@103.x.x.x Call-ID: 1898224288 Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Asterisk server: 17:13:17.093676 IP 103.x.x.x.5060 172.y.y.y.5060: SIP, length: 1228 E...?.?/g.v.INVITE sip:*1@172.y.y.y:5060;transport=udpSIP/2.0 Record-Route: sip:103.x.x.x;lr=on Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0 Via: SIP/2.0/UDP 192.168.1.40:5060 ;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850 From: sip:9067273@103.x.x.x;tag=1880695235 To: sip:*1@103.x.x.x Call-ID: 1898224288 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642
Re: [asterisk-users] Asterisk not receiving call from VPN address
Hi Larry, No, they are on separate machines. On 21 January 2014 17:54, Larry Moore lmo...@omninet.net.au wrote: Is Kamalio running on the same system as Asterisk? On 21/01/2014 2:41 PM, David Cunningham wrote: Hi Larry, Thanks for the reply. We have all of those settings left out of our sip.conf, so this should allow everything, right? On 21 January 2014 17:38, Larry Moore lmo...@omninet.net.au mailto:lmo...@omninet.net.au wrote: Have you checked your localnet=, deny=, permit=, contactdeny= contactpermit= settings? My 2c worth. On 20/01/2014 10:51 AM, David Cunningham wrote: Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers. Thanks in advance for any help. The ngrep on the Asterisk server: U 2014/01/17 13:15:15.599557 172 tel:15.599557%20172.x.x.x:5060 - 103.y.y.y:5060 INVITE sip:9067268@103.y.y.y:5060;__transport=udp SIP/2.0. Record-Route: sip:172.x.x.x;lr=on. Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.__f49ceb73.0. Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;__branch=z9hG4bK806710997. From: 9067271 sip:9067271@172.x.x.x;tag=__198791249. To: sip:9067268@172.x.x.x. Call-ID: 1905625787@192.z.z.z. ... 172.x.x.x is the Kamailio server's VPN address 103.y.y.y is the Asterisk server's real address 192.z.z.z is the calling phone's LAN address -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 tel:%2B1%20213%20221%201092 UK: +44 (0) 20 3298 1642 tel:%2B44%20%280%29%2020%203298%201642 Australia: +61 (0) 2 8063 9019 tel:%2B61%20%280%29%202%208063%209019 -- _ -- 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 http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk not receiving call from VPN address
Hi Andres, Thanks for the idea. We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the sip set debug on didn't show the packets. On 22 January 2014 01:40, Andres and...@telesip.net wrote: David, It seems to me that Asterisk is not seeing/binding to your VPN interface. You need to debug that first. I would set en explicit bind statement in sip.conf to the VPN interface address and nothing else. Then start your asterisk and watch the log messages. It should confirm that it cannot bind to that address. If it does bind, then try your test again and asterisk should see the SIP packets coming in. -- Technical Support http://www.cellroute.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk not receiving call from VPN address
Hi Paul, Thanks, we did try restarting Asterisk after the VPN was up but that didn't solve the issue either. On 22 January 2014 02:55, Paul Belanger paul.belan...@polybeacon.comwrote: On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, Using ngrep/tcpdump shows the packet clearly going from the Kamailio server and arriving at the Asterisk server. This is why it's a mystery that Asterisk doesn't see the call coming in. We tried removing the firewall (so iptables -L shows no rules at all) but that didn't help unfortunately. You might think Kamailio is transmitting it to Asterisk, however without looking at the actually routing tables on Kamailio you'll never know if it actually made it to Asterisk. Again, we need a pcap trace on both Kamailio and Asterisk, plus what your routes look like (route -n), for a call. It will show us clearly what is happening. This all sounds like a routing issue, so your network admins should be able to help troubleshoot. I finally re-read the complete thread. When are you starting the VPN on your Asterisk server, before or after Asterisk has started? If after, and you are binding to 0.0.0.0, it is likely Asterisk is not actually bound to your tun0 interface. So, for a test, explicitly have asterisk listen only on the tun0 interface, retry your call. Or setup your tunnel, then stop Asterisk and start it again, that should cause it to bind properly. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk not receiving call from VPN address
On 22 January 2014 09:11, Steve Edwards asterisk@sedwards.com wrote: (Please don't top-post.) On Wed, 22 Jan 2014, David Cunningham wrote: We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the sip set debug on didn't show the packets. Have you confirmed via 'netstat' (or some other system level toop) that Asterisk is actually listening to UDP port 5060 on the VPN IP address? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 Hi Steve, When we have bindport = 172.x.x.14 then netstat -udpln shows the following. When bindport is 0.0.0.0 then netstat shows it listening on 0.0.0.0 as you'd expect. udp0 0 172.x.x.14:50600.0.0.0:* 18114/asterisk -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk not receiving call from VPN address
At this point in time, you'll need to show us a .pcap on the Asterisk box, when you make a call to it via Kamailio. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Hi Paul, Thanks for the reply. What are you looking for in the PCAP, that isn't in the tcpdump earlier in the thread? I just want to make sure we gather the information required. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk not receiving call from VPN address
Hello, We tried Asterisk 1.8 and 1.6, but not yet Asterisk 11. We'll keep it in mind. In the meantime we've decided to try a different network configuration instead, so the VPN network is separated from what Asterisk sees. Thanks for all the advice given. On 23 January 2014 00:42, Administrator TOOTAI ad...@tootai.net wrote: Le 20/01/2014 03:51, David Cunningham a écrit : Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers. Thanks in advance for any help. Hi David, if you have a 11.7.0 version try to downgrade to 11.6.0.1 See my post about similar issue with no registration of our intranet phones -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] maxsecs not working
Hello, We have servers running Asterisk 1.8.20.1 and 11.7.0, and on both setting maxsecs in voicemail.conf doesn't seem to have any effect. A voicemail keeps recording after the specified time, and when the caller hangs up the voicemail is saved in the mailbox. Are we doing something really silly? Here's the voicemail.conf. We have tried 'voicemail reload' and restarting Asterisk to make it take effect. [general] format = wav mailcmd = /bin/true review = no ; Maximum length of a voicemail message in seconds maxsecs=180 [zonemessages] #include /etc/asterisk/voicemail_zonemessages.conf [default] Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] maxsecs not working
Hi Rusty, We found the problem - a configuration error. Thank you for the response. On 29 May 2014 23:35, Rusty Newton rnew...@digium.com wrote: On Thu, May 22, 2014 at 6:22 PM, David Cunningham dcunning...@voisonics.com wrote: Hello, We have servers running Asterisk 1.8.20.1 and 11.7.0, and on both setting maxsecs in voicemail.conf doesn't seem to have any effect. A voicemail keeps recording after the specified time, and when the caller hangs up the voicemail is saved in the mailbox. Are we doing something really silly? snip Nope that configuration looks fine and it works on my systems as expected in the latest of those branches. Using your configuration I tried changing the maxsecs value and it appears to be respected. If you can reproduce the issue and provide debug to demonstrate, then you might file a bug report. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] dahdi-linux 2.6.2 failing to compile with linux 3.13
Hello, I'm getting the following errors when compiling dahdi-linux 2.6.2 under Ubuntu 14.04 with kernel 3.13.0-24-generic. I did google and found one thread suggesting the errors should be fixed in 2.6.2, and another suggesting to try 2.4 which didn't make sense but I tried anyway, and it gave similar warnings. Would anyone know how to make it compile? Thanks in advance. make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.2/drivers/dahdi/firmware' make -C /lib/modules/3.13.0-24-generic/build SUBDIRS=/usr/src/dahdi-linux-2.6.2/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.6.2/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/linux-headers-3.13.0-24-generic' CC [M] /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.o /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:91:2: warning: #warning No CONFIG_BKL is an experimental configuration. [-Wcpp] #warning No CONFIG_BKL is an experimental configuration. ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c: In function ‘dahdi_proc_open’: /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:902:2: error: implicit declaration of function ‘PDE’ [-Werror=implicit-function-declaration] return single_open(file, dahdi_seq_show, PDE(inode)-data); ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:902:53: error: invalid type argument of ‘-’ (have ‘int’) return single_open(file, dahdi_seq_show, PDE(inode)-data); ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c: In function ‘_dahdi_assign_span’: /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:6945:3: error: implicit declaration of function ‘create_proc_entry’ [-Werror=implicit-function-declaration] span-proc_entry = create_proc_entry(tempfile, 0444, ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:6945:20: warning: assignment makes pointer from integer without a cast [enabled by default] span-proc_entry = create_proc_entry(tempfile, 0444, ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:6952:19: error: dereferencing pointer to incomplete type span-proc_entry-data = (void *)(long)span-spanno; ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:6953:19: error: dereferencing pointer to incomplete type span-proc_entry-proc_fops = dahdi_proc_ops; ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c: In function ‘_dahdi_unassign_span’: /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:7137:37: error: dereferencing pointer to incomplete type remove_proc_entry(span-proc_entry-name, root_proc_entry); ^ cc1: some warnings being treated as errors make[2]: *** [/usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.o] Error 1 make[1]: *** [_module_/usr/src/dahdi-linux-2.6.2/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-headers-3.13.0-24-generic' make: *** [modules] Error 2 make: Leaving directory `/usr/src/dahdi-linux-2.6.2' 'make -C dahdi-linux-2.6.2 install' failed with 512. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi-linux 2.6.2 failing to compile with linux 3.13
Thank you very much. On 14 June 2014 00:33, Shaun Ruffell sruff...@digium.com wrote: On Fri, Jun 13, 2014 at 12:54:14PM +1000, David Cunningham wrote: Hello, I'm getting the following errors when compiling dahdi-linux 2.6.2 under Ubuntu 14.04 with kernel 3.13.0-24-generic. I did google and found one thread suggesting the errors should be fixed in 2.6.2, and another suggesting to try 2.4 which didn't make sense but I tried anyway, and it gave similar warnings. Would anyone know how to make it compile? Thanks in advance. make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.2/drivers/dahdi/firmware' make -C /lib/modules/3.13.0-24-generic/build SUBDIRS=/usr/src/dahdi-linux-2.6.2/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.6.2/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/linux-headers-3.13.0-24-generic' CC [M] /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.o /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:91:2: warning: #warning No CONFIG_BKL is an experimental configuration. [-Wcpp] #warning No CONFIG_BKL is an experimental configuration. ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c: In function ‘dahdi_proc_open’: /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:902:2: error: implicit declaration of function ‘PDE’ [-Werror=implicit-function-declaration] return single_open(file, dahdi_seq_show, PDE(inode)-data); ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:902:53: error: invalid type argument of ‘-’ (have ‘int’) return single_open(file, dahdi_seq_show, PDE(inode)-data); ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c: In function ‘_dahdi_assign_span’: /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:6945:3: error: implicit declaration of function ‘create_proc_entry’ [-Werror=implicit-function-declaration] span-proc_entry = create_proc_entry(tempfile, 0444, ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:6945:20: warning: assignment makes pointer from integer without a cast [enabled by default] span-proc_entry = create_proc_entry(tempfile, 0444, ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:6952:19: error: dereferencing pointer to incomplete type span-proc_entry-data = (void *)(long)span-spanno; ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:6953:19: error: dereferencing pointer to incomplete type span-proc_entry-proc_fops = dahdi_proc_ops; ^ /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c: In function ‘_dahdi_unassign_span’: /usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.c:7137:37: error: dereferencing pointer to incomplete type remove_proc_entry(span-proc_entry-name, root_proc_entry); ^ cc1: some warnings being treated as errors make[2]: *** [/usr/src/dahdi-linux-2.6.2/drivers/dahdi/dahdi-base.o] Error 1 make[1]: *** [_module_/usr/src/dahdi-linux-2.6.2/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-headers-3.13.0-24-generic' make: *** [modules] Error 2 make: Leaving directory `/usr/src/dahdi-linux-2.6.2' 'make -C dahdi-linux-2.6.2 install' failed with 512. This was fixed in the DAHDI sources in dahdi: Replace create_proc_entry() with proc_create_data() [1] You will need to update to at least dahdi-linux 2.8.0 if you would like to run against kernels newer than 3.10. [1] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=84ccc65 -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Detect hangup due to RTP timeout
Hello, Is there a way to tell in the dialplan, Asterisk AGI, or Asterisk AMI, when a call has been hung up because the SIP rtptimeout has been reached? Thank you, -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] WebRTC demo phones
Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the enable video checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with Rejecting secure video stream without encryption details. - sipML5, but it won't register, perhaps something to do with not using the Asterisk Websocket server (which I don't see an option to choose) - Janus, but the INVITE SDP contains RTP/AVP not RTP/SAVP, and Asterisk rejects it with We are requesting SRTP for audio, but they responded without it! Thanks for any suggestions. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk 13 chan_sip trunk appending @string to dialled number
Hi Brendan, Can you attach an Asterisk log with sip set debug on, core set verbose 9 and core set debug 9? On 18 August 2015 at 10:33, Brendan Ord b...@staff.onthenet.com.au wrote: Hello, I’m having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends ‘@CUBE’ onto the end of the dialled number, as per the following examples; Asterisk log; app_dial.c: Called SIP/test/0429123456@CUBE chan_sip.c: Got SIP response 500 Internal Server Error back from 172.22.4.12:5060 In the SIP SDP; INVITE sip:0429920437%40CUBE@172.22.4.12 SIP/2.0. To: sip:0429920437%40CUBE@172.22.4.12. As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I’m really not sure where this is coming from, and why. Here is my trunk configuration; PEER type=friend qualify=yes nat=no insecure=port,invite host=172.22.4.12 dtmfmode=rfc2833 context=from-trunk allow=ulaw disallow=all USER type=friend qualify=yes nat=no host=172.22.4.12 dtmfmode=rfc2833 allow=ulaw disallow=all canreinvite=no Thanks for any help J Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map https://goo.gl/maps/p25WF) www.OntheNet.com.au http://www.onthenet.com.au/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Is peer order in sip.conf important?
Hi Murthy, You probably want [69.59.234.67] first so that it matches on inbound calls. For outbound the question is what exactly do you specify in your dial? It should be something like number@vonage-out. On 14 August 2015 at 02:25, Murthy Gandikota murth...@hotmail.com wrote: Hi All Noticed in sip.conf that the asterisk (v11) is sensitive to the order of peers. Here is my sip.conf [general] context = demo ; 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 context=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=72.220.28.226 localnet=192.168.0.0 nat=yes maxexpiry=15 minexpiry=14 register =16194077214:password@69.59.234.67:5060 [vonage-out] username=16194077214 type=friend secret=password port=5061 nat=yes host=69.59.234.67 fromuser=1619xxx fromdomain=69.59.234.67 dtmfmode=rfc2833 auth=md5 [69.59.234.67] username=1619xxx ;type=friend type=peer ;type=user secret=password port=5061 nat=yes insecure=port,invite host=69.59.234.67 fromuser=1619xxx fromdomain=69.59.234.67 ;dtmfmode=inband context=from-pstn canreinvite=no ;auth=md5 disallow=all allow=ulaw ;allow=alaw ;allow=g729 ;allow=g723 When I make the INBOUD call, vonage-out peer is selected based on the debug. In other words if my sip.conf is as follows [general] [vonage-out] [69.59.234.67] ... Then the peer Asterisk selects is vonage-out. I want vonage-out to be used for OUTBOUND as the name implies. However if I switch them, as follows: [general] ... [69.59.234.67] ... [vonage-out] . Then the peer 69.59.234.67 is selected which is what I want for an INBOUND. Any idea why? Your kind help is appreciated. Best regards murthy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] re-invite update dialog
Hi Kelvin, Can you copy and paste in the dialog so we can see the call being set up and the re-invite? What version of Asterisk is it? On 28 July 2015 at 20:05, Kelvin Chua kel...@gmail.com wrote: I don't know if this is something asterisk can do at the moment but on my setup, it does not. What I intend to do is, while a client is in a call, it will send an in-dialog re-invite to asterisk (after changes on the client i.e. IP address). Asterisk should handle this and update internal dialog. when the other party hangs up, BYE will be sent to the new IP. in my setup, asterisk still sends BYE to the old IP. Is this something we can already do? or possible to add? Kelvin Chua -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk 13 chan_sip trunk appending @string to dialled number
Yes indeed. Do you have the dialplan, eg from /etc/asterisk/extensions.conf? Something is getting this OUT_3_SUFFIX variable and including it in a Dial to 172.22.4.12. On 18 August 2015 at 16:21, Brendan Ord b...@staff.onthenet.com.au wrote: Starting to make sense when I saw this line: [2015-08-18 15:01:33] DEBUG[19366][C-1cfc]: pbx.c:3785 ast_str_retrieve_variable: Result of 'OUT_3_SUFFIX' is '@CUBE' But I can’t find where this is in configuration .. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map https://goo.gl/maps/p25WF) www.OntheNet.com.au http://www.onthenet.com.au/ *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Brendan Ord *Sent:* Tuesday, 18 August 2015 3:44 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number David, I should also note; 246 is my extension, it has IP 172.22.3.238. 172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway. The trunk is called ‘testing’ at the moment. The route that selects this trunk uses a 9 prefix. This system is in semi-production, so there might be fluff in the log from other active calls. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map https://goo.gl/maps/p25WF) www.OntheNet.com.au http://www.onthenet.com.au/ *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham *Sent:* Tuesday, 18 August 2015 2:39 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number Hi Brendan, Can you attach an Asterisk log with sip set debug on, core set verbose 9 and core set debug 9? On 18 August 2015 at 10:33, Brendan Ord b...@staff.onthenet.com.au wrote: Hello, I’m having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends ‘@CUBE’ onto the end of the dialled number, as per the following examples; Asterisk log; app_dial.c: Called SIP/test/0429123456@CUBE chan_sip.c: Got SIP response 500 Internal Server Error back from 172.22.4.12:5060 In the SIP SDP; INVITE sip:0429920437%40CUBE@172.22.4.12 SIP/2.0. To: sip:0429920437%40CUBE@172.22.4.12. As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I’m really not sure where this is coming from, and why. Here is my trunk configuration; PEER type=friend qualify=yes nat=no insecure=port,invite host=172.22.4.12 dtmfmode=rfc2833 context=from-trunk allow=ulaw disallow=all USER type=friend qualify=yes nat=no host=172.22.4.12 dtmfmode=rfc2833 allow=ulaw disallow=all canreinvite=no Thanks for any help J Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map https://goo.gl/maps/p25WF) www.OntheNet.com.au http://www.onthenet.com.au/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Asterisk 13 chan_sip trunk appending @string to dialled number
Glad to hear it's sorted.
On 18 August 2015 at 17:08, Brendan Ord b...@staff.onthenet.com.au wrote:
Halt the wild goose chase
It was obviously something left over in the dial plan. Restarted Asterisk
seeing as though we're now after-hours and I can do interruptive work, and
it seems to have solved my @CUBE problem.
Interestingly, it persisted through a dialplan reload and the equivalent
of a core reload too ..
[2015-08-18 17:04:30] VERBOSE[25543][C-] app_dial.c: Called
SIP/testing/0429920437
[2015-08-18 17:04:30] VERBOSE[25543][C-] app_dial.c: Everyone is
busy/congested at this time (1:0/0/1)
This is expected, I need to review the dial-peer configurations on the
Cisco GW. At least it isn't throwing the suffix on the end anymore it
seems...
Thanks for the help and apologies for the goose chase ..
Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au
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-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Brendan Ord
Sent: Tuesday, 18 August 2015 4:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string
to dialled number
Hello,
So, I found this line under macro-dialout-trunk, in
extensions_additional.conf (FreePBX, so it controls the conf files mostly);
exten =
s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS})
If I grep for OUT_3_SUFFIX in all files in /etc/asterisk I get nothing..
Here's a paste of a few things out of the two files that I thought were
relevant to how FreePBX configured this trunk ...
http://pastebin.com/5fRy2Ai9
Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au
NOTICE:
This e-mail and any attachments are private and confidential and may
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-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell
Sent: Tuesday, 18 August 2015 4:38 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string
to dialled number
just got back to my mail.
What I'd do is go into /etc/asterisk and grep for OUT_3_SUFFIX in all the
files
once the file with that variable is located, we can figure out why it's
adding it
On 08/17/2015 11:26 PM, David Cunningham wrote:
Yes indeed.
Do you have the dialplan, eg from /etc/asterisk/extensions.conf?
Something is getting this OUT_3_SUFFIX variable and including it in a
Dial to 172.22.4.12.
On 18 August 2015 at 16:21, Brendan Ord b...@staff.onthenet.com.au
mailto:b...@staff.onthenet.com.au wrote:
Starting to make sense when I saw this line:
[2015-08-18 15:01:33] DEBUG[19366][C-1cfc]: pbx.c:3785
ast_str_retrieve_variable: Result of 'OUT_3_SUFFIX' is '@CUBE'
But I can’t find where this is in configuration ..
Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map
https://goo.gl/maps/p25WF)
www.OntheNet.com.au http://www.onthenet.com.au/
*From:*asterisk-users-boun...@lists.digium.com mailto:
asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Brendan Ord
*Sent:* Tuesday, 18 August 2015 3:44 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk 13 chan_sip trunk
appending @string to dialled number
David,
I should also note;
246 is my extension, it has IP 172.22.3.238.
172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway.
The trunk is called ‘testing’ at the moment. The route
Re: [asterisk-users] Shared RealTime Database
You need to: 1. Have a systemname = whatever in asterisk.conf under [options] 2. Set rtsavesysname=yes in sip.conf under [general] 3. Have a column called “regserver” in the SIP peer database Source: https://asterblog.wordpress.com/2007/07/24/using-multiple-servers-and-one-database-with-regserver/ On 18 August 2015 at 16:38, Bruce Ferrell bferr...@baywinds.org wrote: yes, the sip_buddies (or equal) has a field that says which server handled the registration On 08/17/2015 07:58 AM, Mehdi Shirazi wrote: Hi If we have a shared RealTime database for sip registration of multiple Asterisk servers, is there a way to realize which Asterisk server registered sip phones ? Regards M.Shirazi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] SRV lookups in Asterisk 11
Hello, Can anyone advise on the status of SRV lookups in Asterisk 11? (specifically 11.17.1) Is there any difference given how the Dial is done, and how supported are weights and priorities? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Mixing PJSIP realtime and flat files
Hello, Is it possible to mix PJSIP realtime and flat file configuration in pjsip,conf? What we want is to set up endpoints in the ps_endpoints table with some columns set but most being NULL, and then allow end-customers to optionally add configuration by adding a pjsip.conf section. For example, in ps_endpoinds might be an endpoint with id "asterisk-1" with the transport, aors, auth, and context columns set but all other fields NULL. Then the end-customer could add a [asterisk-1] section in pjsip.conf which sets the codecs they want to enable. We tried this but it seemed that the [asterisk-1] section in pjsip.conf had no effect. Our sorcery.conf is attached. Is this possible, and how do we do it? Thanks very much for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 sorcery.conf Description: Binary data -- _ -- 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] Mixing PJSIP realtime and flat files
Shame, but thank you very much for the reply Joshua. On 22 January 2016 at 10:26, Joshua Colp <jc...@digium.com> wrote: > David Cunningham wrote: > >> Hello, >> >> Is it possible to mix PJSIP realtime and flat file configuration in >> pjsip,conf? >> >> What we want is to set up endpoints in the ps_endpoints table with some >> columns set but most being NULL, and then allow end-customers to >> optionally add configuration by adding a pjsip.conf section. >> >> For example, in ps_endpoinds might be an endpoint with id "asterisk-1" >> with the transport, aors, auth, and context columns set but all other >> fields NULL. Then the end-customer could add a [asterisk-1] section in >> pjsip.conf which sets the codecs they want to enable. >> >> We tried this but it seemed that the [asterisk-1] section in pjsip.conf >> had no effect. Our sorcery.conf is attached. >> >> Is this possible, and how do we do it? Thanks very much for any advice. >> > > It's not possible to do this. Each source (realtime, config file) provides > the complete definition. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] "Expected to acknowledge ticks" problem
Hello, We are seeing an issue where a call will be "stuck" for roughly a minute or so and in the Asterisk log give many "Expected to acknowledge ticks" warnings as below. Can anyone suggest a cause? The system is running 11.21.2. Thanks in advance. [Mar 10 08:00:25] DEBUG[3158][C-273f] channel.c: Set channel Local/4171411@product-phone-217b;2 to write format slin [Mar 10 08:00:25] DEBUG[3158][C-273f] res_musiconhold.c: Local/4171411@product-phone-217b;2 Opened file 0 '/var/lib/product/music/2/2/1' [Mar 10 08:00:25] DEBUG[3146][C-273f] res_rtp_asterisk.c: Difference is 888, ms is 131 [Mar 10 08:00:25] DEBUG[3158][C-273f] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Mar 10 08:00:30] DEBUG[3146][C-273f] res_rtp_asterisk.c: Got RTCP report of 76 bytes [Mar 10 08:00:30] DEBUG[3158][C-273f] res_musiconhold.c: Local/4171411@product-phone-217b;2 Opened file 0 '/var/lib/product/music/2/2/1' [Mar 10 08:00:30] DEBUG[3158][C-273f] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Mar 10 08:00:30] DEBUG[3146][C-273f] res_rtp_asterisk.c: Difference is 752, ms is 114 [Mar 10 08:00:32] DEBUG[3146][C-273f] res_rtp_asterisk.c: Got RTCP report of 76 bytes [Mar 10 08:00:35] DEBUG[3158][C-273f] res_musiconhold.c: Local/4171411@product-phone-217b;2 Opened file 0 '/var/lib/product/music/2/2/1' [Mar 10 08:00:35] DEBUG[3158][C-273f] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Mar 10 08:00:35] DEBUG[3146][C-273f] res_rtp_asterisk.c: Difference is 824, ms is 123 [Mar 10 08:00:36] DEBUG[3146][C-273f] res_rtp_asterisk.c: Got RTCP report of 76 bytes [Mar 10 08:00:40] DEBUG[3158][C-273f] res_musiconhold.c: Local/4171411@product-phone-217b;2 Opened file 0 '/var/lib/product/music/2/2/1' [Mar 10 08:00:40] DEBUG[3158][C-273f] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Advice of Charge for non-Snom SIP phones
Hello, We are generating AOC messages via the AMI and trying to deliver them to various brands of SIP phone. When "snom_aoc_enabled = yes" in sip.conf then a message is set in the Snom format correctly. We're not having much luck sending any other type of AOC message though. I can't find any documentation to say what if anything is available. The "aoc_enable" setting doesn't seem to have any effect in sip.conf. Can anyone advise if there is any other support for AOC over SIP besides Snom, and how to configure it? Thank you, -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -- _ -- 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] AMI version of CONNECTEDLINE
Hi Jacek, Thank you very much for the suggestion. Using SetVar and CONNECTEDLINE(number) works. On 12 December 2016 at 18:31, Jacek Konieczny <jaj...@jajcus.net> wrote: > On 2016-12-12 02:21, David Cunningham wrote: > >> Is there any equivalent of the CONNECTEDLINE function which can be >> called from an application using the AMI? >> > > You can use dialplan functions from AMI using GetVar, so this should work: > > Action: GetVar > Variable: CONNECTEDLINE(num) > > Jacek > > -- > _ > -- 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 > -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -- _ -- 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] AMI version of CONNECTEDLINE
Hello, Is there any equivalent of the CONNECTEDLINE function which can be called from an application using the AMI? Thanks for any ideas. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Custom INFO for Advice Of Charge
Hello, Does anyone know how to send a SIP INFO with custom data on a specified channel from Asterisk? The intention is to provide an Advice Of Charge. Asterisk has it's own AOC function but it only seems to support Snom's format, not the more general XML format. Searching came up with SIPSendCustomInfo but apparently it sends on all active SIP channels, and is only available with TEST_FRAMEWORK. Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -- _ -- 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] How to steal an answered call?
Hello, I'm familiar with Pickup/PickupChan for taking a ringing call, but does anyone know how a phone can "steal" an already answered call from another phone? Our users have decided that call parking is too long-winded and don't want to use that. For example: phone A calls phone B, phone B answers the call, phone C dials something to "steal" the call from B, and finally A and C are talking. Searching on voip-info.org shows a "BristuffSteal" command but it's very out of date (Asterisk 1.2). Thanks in advance for any suggestions. Kind regards, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] How to steal an answered call?
Hello Patrick and others, Thanks, I wasn't familiar with the Bridge application and it may allow us to do what's needed. A transfer would of course be simpler but the user wants what the user wants... Thank you. On 9 July 2018 at 19:52, John Kiniston wrote: > David, > > You should be able to use the Bridge dialplan application to do what you > want. > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Bridge > > I use the CHANNELS function and the IMPORT function to find the channel to > bridge to my caller. > > > On Sun, Jul 8, 2018 at 8:17 PM David Cunningham > wrote: > >> Hello, >> >> I'm familiar with Pickup/PickupChan for taking a ringing call, but does >> anyone know how a phone can "steal" an already answered call from another >> phone? Our users have decided that call parking is too long-winded and >> don't want to use that. >> >> For example: phone A calls phone B, phone B answers the call, phone C >> dials something to "steal" the call from B, and finally A and C are talking. >> >> Searching on voip-info.org shows a "BristuffSteal" command but it's very >> out of date (Asterisk 1.2). >> >> Thanks in advance for any suggestions. >> >> Kind regards, >> >> -- >> David Cunningham, Voisonics Limited >> http://voisonics.com/ >> USA: +1 213 221 1092 >> New Zealand: +64 (0)28 2558 3782 >> -- >> _ >> -- 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 > > > > -- > A human being should be able to change a diaper, plan an invasion, butcher > a hog, conn a ship, design a building, write a sonnet, balance accounts, > build a wall, set a bone, comfort the dying, take orders, give orders, > cooperate, act alone, solve equations, analyze a new problem, pitch manure, > program a computer, cook a tasty meal, fight efficiently, die gallantly. > Specialization is for insects. > ---Heinlein > > -- > _ > -- 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 > -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] Getting DTMF from Asterisk Record?
Hello, Does anyone know if it's possible to get the non-terminating DTMF keys that may have been pressed while using the Record command in Asterisk? The intended purpose is that a caller can say a name or enter it using DTMF keys. If we can get the keys in a variable that would be very helpful. Thanks for any advice. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] ChanSpy "Audiohook has stale audio in its factories" problem
Hello, We have an issue where using ChanSpy with the 'B' option to talk to both parties doesn't work in some call scenarios (calling out to a T-mobile device). The problem is that the spying party can talk to the called person but not the caller. The Asterisk log in this case records warnings as below, which then continue for the life of the spying call. Would anyone know the cause of this? We are using Asterisk 11.25.3. Thanks in advance for any advice. [Jan 18 15:59:39] DEBUG[39503][C-0bbd] autochan.c: Created autochan 0x152564004ea0 to hold channel SIP/15-xx.yy.122.136-0c85 (0x15251806fb28) [Jan 18 15:59:39] NOTICE[39503][C-0bbd] app_chanspy.c: Attaching SIP/enswitch-local-0c86 to SIP/15-xx.yy.122.136-0c85 [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Audiohook 0x1525f5d59c58 has stale audio in its factories. Flushing them both [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Audiohook 0x1525f5d5b6b0 has stale audio in its factories. Flushing them both [Jan 18 15:59:39] DEBUG[38605][C-0bbc] audiohook.c: Audiohook 0x1525f5d58200 has stale audio in its factories. Flushing them both [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160 samples from write factory 0x1525f5d58ec8 [Jan 18 15:59:39] DEBUG[39503][C-0bbd] res_rtp_asterisk.c: Difference is 888, ms is 131 [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160 samples from write factory 0x1525f5d58ec8 [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Read factory 0x1525f5d58288 and write factory 0x1525f5d58ec8 both fail to provide 160 samples [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160 samples from write factory 0x1525f5d58ec8 [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160 samples from write factory 0x1525f5d58ec8 [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Audiohook 0x1525f5d59c58 has stale audio in its factories. Flushing them both [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160 samples from write factory 0x1525f5d58ec8 [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160 samples from write factory 0x1525f5d58ec8 [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160 samples from write factory 0x1525f5d58ec8 [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160 samples from write factory 0x1525f5d58ec8 [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160 samples from write factory 0x1525f5d58ec8 -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] Reliable information on which SIP party is transferring call
Hello, Can anyone advise whether there's some method to reliably determine which SIP device is the party performing a transfer (REFER or using features.conf), and to whom they're transferring? We've been analysing the AMI AttendedTransfer events and of course an extension isn't necessarily a SIP device, and while we can usually figure one from the other for one transfer, for two or more transfers it appears to be a mess of information. Is there some secret to figuring out a clean set of rules? Ideally for our purpose the events would closely resemble those in the SIP packets themselves. Thanks in advance for any help. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] Find out which key ended recording?
Hi Steve,
What language is that please? We're using Perl and so far I haven't found
an equivalent there.
Thanks for your help.
On Fri, 7 Jun 2019 at 12:10, Steve Edwards
wrote:
> On Fri, 7 Jun 2019, David Cunningham wrote:
>
> > We have a need to record audio and allow the user to press any DTMF key
> > to end the recording. Currently we're using the AGI command "record
> > file" which does allow us to specify which DTMF keys can end the
> > recording.
> >
> > However we also need to know which key actually ended the recording.
> > Note that only allowing # or * to end the recording won't work for us.
> >
> > Does anyone know how we can tell which key ended the recording? Thanks
> > in advance for any help.
>
> Here's a snippet from one of my AGIs:
>
> // record the voice
> exec_agi("RECORD FILE"
>" %s" // filename
>" wav"// format
>" #*1234567890"
> // escape digits
>" %d000" // timeout in ms
>" BEEP" // BEEP
> , recorded_path
> , recording_limit
> );
>
> // should we abort?
> if ('*' == agi_environment.result)
> {
> agi_set_variable("STATUS", "*");
> exit(EXIT_SUCCESS);
> }
>
> // are we finished?
> if ('#' == agi_environment.result)
> {
> break;
> }
>
> Looks like agi_environment.result is your Huckleberry.
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
>
> --
> _
> -- 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
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
--
_
-- 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:
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[asterisk-users] Find out which key ended recording?
Hello, We have a need to record audio and allow the user to press any DTMF key to end the recording. Currently we're using the AGI command "record file" which does allow us to specify which DTMF keys can end the recording. However we also need to know *which* key actually ended the recording. Note that only allowing # or * to end the recording won't work for us. Does anyone know how we can tell which key ended the recording? Thanks in advance for any help. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] Find out which key ended recording?
Hi Steve,
Thank you very much for that information. The result is the key in ascii
perfectly!
On Fri, 7 Jun 2019 at 18:05, Steve Edwards
wrote:
> On Fri, 7 Jun 2019, David Cunningham wrote:
>
> > We're using Perl and so far I haven't found an equivalent there.
>
> On Thu, 6 Jun 2019, Steve Edwards wrote:
>
> > I'm not much of a Perl programmer...
>
> But you should never turn down an opportunity to develop your skills :)
>
> Try something like:
>
> my $result = $AGI->record_file(
>'/tmp/foo'# filename
> , 'wav' # format
> , '#*0123456789'# escape digits
> , '5000'# timeout
> );
> $AGI->verbose('result = ' . $result, 0);
>
> Which results in:
>
> AGI Rx << RECORD FILE /tmp/foo wav #*0123456789
> 5000
> AGI Tx >> 200 result=50 (dtmf) endpos=0
> AGI Rx << VERBOSE "result = 50"
>
> when '2' is pressed.
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
>
> --
> _
> -- 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
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
--
_
-- 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:
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Re: [asterisk-users] pjsip and tls client: How to decrypt Wireshark trace?
Hi Michael, If you can get a copy of the private key you can import that to Wireshark and see the encrypted information: https://support.citrix.com/article/CTX116557 On Sun, 12 May 2019 at 04:55, Michael Maier wrote: > Hello! > > I'm just wondering if it's possible to decrypt sips packages in Wireshark > while asterisk runs as sips client (connecting to the provider w/ > tls 1.2)? I don't use an own certificate. > > > Thanks > Michael > > -- > _ > -- 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 -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] Change of H264 profile level problem
Hello, Can anyone help with an issue regarding the H264 profile level being passed through Asterisk? We have a video call like this: Caller A -> Asterisk -> Called B Caller A's INVITE SDP offers "profile-level-id=42801f", and Called B replies a 200 OK containing "profile-level-id=42801e" in its SDP. Note that this ends with an 'e' rather than an 'f'. The problem is that Asterisk forwards the 200 OK to Caller A with "profile-level-id=42801f" in the SDP, so not what Called B sent. Caller A then starts transmitting video in a resolution that Called B can't handle, and the video is displayed as blank. My question is, since Asterisk doesn't do video transcoding, why doesn't it pass though Called B's "profile-level-id=42801e" unchanged? If it did then Caller A might use a resolution that Called B can handle. We are using Asterisk 11.25.3. Thanks in advance for any assistance. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] One Touch Record and a matching entry in sip.conf
Hello, This has a bit of a long explanation... ultimately the question is why adding a section to sip.conf made a difference to One Touch Record. We're implementing a recording toggle using the "Record" button on a SIP telephone and Asterisk's "One Touch Record" feature in features.conf. It worked without problem when the calling party pressed the Record button, but it didn't work when the called party pressed the Record button. When the called party pressed the button Asterisk logged "Recording requested, but no One Touch Monitor registered. (See features.conf)", even though it was certainly enabled in features.conf The destination was called with a dial string like SIP/@12.34.56.78:5060. Eventually we worked out that adding a section to sip.conf like this allowed it to start working: [12.34.56.78:5060] type = friend ... etc.. We already (before it started working) had a section almost identical but without the port in the section name: [12.34.56.78] type = friend ... etc.. The question is, why did adding the section with the port in the name make a difference? Is it because only Asterisk sip users (of which a friend is one) are allowed to use One Touch Record? If so I don't see this documented anywhere. Note that adding or removing the 'w' or 'W' options to the Dial seems to make no difference at all, and following the addition of the sip.conf section it works without either of these options in the Dial. Thank you in advance for any insight into this. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] Music on hold depending on who put call on hold
Hello, Does anyone know of a way to play different music on hold depending on which party puts the call on hold? We can specify the music on hold per channel, but that doesn't do what is needed. We want to play one music if the caller puts the call on hold, and a different music if the called party puts the call on hold. Thanks in advance for any assistance. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] VoIP support engineer opportunity
Hello, Voisonics is hiring a VoIP support engineer to assist our customers running Asterisk based hosted PBX platforms. This is a part-time contract work-from-home position. For communication reasons we're looking for someone in a timezone encompassing Far East Asia, Australia, New Zealand, Canada, the USA, and Mexico. If you are not physically located in that area please do not apply - being "flexible" from another part of the world is not what we're looking for. The role involves providing technical support of Asterisk based PBX platforms to our customer's technical staff, Linux system administration, and small dev-ops type development projects. It does not involve providing technical support to end users or the general public. Customers are located around the world. You will generally be responding during your business hours, though sometimes out of hours work will be necessary. Once training is completed, the position will involve providing 24x7 on-call emergency cover in rotation with other staff. Must-have: 1. Fluent command-line Linux ability on Ubuntu, Debian, CentOS, and/or RedHat. 2. Asterisk administration and configuration experience. 3. SIP debugging experience. For example, you should know what packets are typically involved in setting up a call. 4. Experience with administration and configuration of Apache or MySQL. 5. Ability to program at least one language, such as Perl, Shell script, C, Go, PHP, etc. 6. Good written and verbal English language ability. 7. Independant and critical thinking ability. 8. Have experience providing professional IT support to business. 9. Be an individual self-employed contractor. Nice-to-have: 1. Experience with Kamailio, NFS, GlusterFS, Puppet, or Zabbix. 2. Javascript or AngularJS programming ability. 3. HTML and CSS programming ability. 4. Advanced network knowledge (beyond basic Linux networking which is a must-have). To apply for this role email me off-list. In your application: 1. List your experience/compatibility for each of the must-have requirements individually, plus any of the nice-to-have items you fit as well. 2. Provide your physical location, hours of availability, and indication of hourly rate. 3. Let us know what other work you have during business hours. Thank you, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] rotatestrategy = none not working
Hello, We have an Asterisk 11.3 server where we want log rotation handled purely by Linux's logrotate, and not by Asterisk. To this end we've configured the [general] action of /etc/asterisk/logger.conf with: rotatestrategy = none However, an "asterisk -rx 'logger reload'" still rotates the log files. Does anyone know why? Thank you in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] rotatestrategy = none not working
Hi Steve, Thanks for the answer. Since that's what we already have configured, any idea why it wouldn't work? As I said, when "asterisk -rx 'logger reload'" is run it still rotates the log file. On Wed, 20 May 2020 at 18:37, Steve Edwards wrote: > On Wed, 20 May 2020, David Cunningham wrote: > > > We have an Asterisk 11.3 server where we want log rotation handled > > purely by Linux's logrotate, and not by Asterisk. To this end we've > > configured the [general] action of /etc/asterisk/logger.conf with: > > > > rotatestrategy = none > > > > However, an "asterisk -rx 'logger reload'" still rotates the log files. > > Does anyone know why? > > I had to hunt, but I found an 11.17.1 system :) > > 'none' does not rotate a log file on this host. Here's my logger.conf: > > ; Created by makefile on 2020-05-19 at 23:05:08 > ; from /source/src/obl-server/logger.conf.pre > > [general] > rotatestrategy = none > > [logfiles] > /tmp/ast-log-test = > debug,dtmf,error,event,notice,verbose,warning > > ; (end of /etc/asterisk/obl/logger.conf) > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > https://www.linkedin.com/in/steve-edwards-4244281 > > -- > _ > -- 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 -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] Multiple IP addresses and using same IP for outbound calls as inbound
Hello, We have an Asterisk server with two public IP addresses, let's say 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call dialled from Asterisk to an external destination. The external destination sees the SIP packet as coming from 1.1.1.1 and the media address in the SDP is 1.1.1.1, which is great. However if we receive a call in to 2.2.2.2 then the call dialled from Asterisk to an external destination still comes from 1.1.1.1, whereas we want it to come from 2.2.2.2. The source of any dialled call (the IP packet and the SDP media address) should be the same as the address the related inbound call was received to. For example: INVITE received to 1.1.1.1:5060 -> Asterisk dials destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to termination.com INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@pstn.com -> INVITE sent from 2.2.2.2:5060 to pstn.com Does anyone know how this can be achieved? Thanks in advance for your help, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] Multiple IP addresses and using same IP for outbound calls as inbound
OK, thank you George. On Sat, 24 Oct 2020 at 03:16, George Joseph wrote: > > > On Thu, Oct 22, 2020 at 4:13 PM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hi George, >> >> Thank you for the response. I'm a little unclear on what you mean by a >> transport. We're using chan_sip, not pjsip. >> >> Do you mean a device in sip.conf, using bindaddr to set the address to >> bind for that device? We've only used bindaddr in the [general] section >> before, but if it will work in a device that could be the answer. >> > > Sorry. I just assume chan_pjsip these days. Not sure how you'd do it for > chan_sip. > > > >> >> >> On Fri, 23 Oct 2020 at 00:13, George Joseph wrote: >> >>> >>> >>> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham < >>> dcunning...@voisonics.com> wrote: >>> >>>> Hello, >>>> >>>> We have an Asterisk server with two public IP addresses, let's say >>>> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with >>>> a call dialled from Asterisk to an external destination. The external >>>> destination sees the SIP packet as coming from 1.1.1.1 and the media >>>> address in the SDP is 1.1.1.1, which is great. >>>> >>>> However if we receive a call in to 2.2.2.2 then the call dialled from >>>> Asterisk to an external destination still comes from 1.1.1.1, whereas we >>>> want it to come from 2.2.2.2. The source of any dialled call (the IP packet >>>> and the SDP media address) should be the same as the address the related >>>> inbound call was received to. >>>> >>>> For example: >>>> INVITE received to 1.1.1.1:5060 -> Asterisk dials >>>> destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to >>>> termination.com >>>> INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@pstn.com >>>> -> INVITE sent from 2.2.2.2:5060 to pstn.com >>>> >>>> Does anyone know how this can be achieved? >>>> >>> >>> If termination.com is only on 1.1.1.1 and pstn.com is only on 2.2.2.2, >>> create 2 transports, one specifically bound to 1.1.1.1, transport-1.1.1.1 >>> for instance, and another to 2.2.2.2: transport-2.2.2.2. The names >>> aren't important as long as you can tell the difference. Then explicitly >>> configure endpoint termination.com's "transport" parameter to >>> "transport-1.1.1.1" and pstn.com's "transport" parameter to >>> "transport-2.2.2.2". In your dialplan, you can see which endpoint the >>> call came in on, and route it out the same endpoint. >>> >>> If both providers are available from both interfaces, you can create 2 >>> endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1, >>> termination.com-2.2.2.2 and pstn.com-2.2.2.2; Then configure each with the >>> same transports as above. >>> >>> >>> >>> >>> >>>> >>>> Thanks in advance for your help, >>>> >>>> -- >>>> David Cunningham, Voisonics Limited >>>> http://voisonics.com/ >>>> USA: +1 213 221 1092 >>>> New Zealand: +64 (0)28 2558 3782 >>>> -- >>>> _ >>>> -- 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 >>> >>> >>> >>> -- >>> George Joseph >>> Asterisk Software Developer >>> direct/fax +1 256 428 6012 >>> Check us out at www.sangoma.com and www.asterisk.org >>> [image: image.png] >>> -- >>> _ >>> -- 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://wi
Re: [asterisk-users] NAT problem with recvonly calls
Hi Dovid, Thanks for that. Can you explain how the Progress() and/or Playback() actually help the NAT problem? I'm trying to figure out how it tells Asterisk the correct address to send the RTP to. On Thu, 3 Dec 2020 at 16:10, Dovid Bender wrote: > David, > > You should be able to do that via the agi as well. > > On Wed, Dec 2, 2020 at 20:32 David Cunningham > wrote: > >> Hi Dovid, >> >> We're using Enswitch so it uses AGI rather than a regular Asterisk >> dialplan, but perhaps sending it to a custom-made Asterisk context with the >> lines you suggest could be the best way forward. >> >> Thank you for that. >> >> >> On Thu, 3 Dec 2020 at 13:01, Dovid Bender wrote: >> >>> David, >>> >>> Does Asterisk send a 180 or a 183 with SDP? We have found that using >>> these two lines help (where xc is a 500ms blank sound file) >>> Exten => _X.,n, Progress() >>> Exten => _X.,n, Playback(xc,noanswer) >>> >>> >>> On Wed, Dec 2, 2020 at 4:30 PM David Cunningham < >>> dcunning...@voisonics.com> wrote: >>> >>>> Hello, >>>> >>>> We have a problem with a SIP doorbell device which sends media one way >>>> only, and NAT at the receiving device. >>>> >>>> When the doorbell button is pressed it makes a call to a configured >>>> destination. Since the doorbell only sends and doesn't receive it sends the >>>> INVITE with sendonly in the SDP, and the destination then replies with a >>>> 200 OK with recvonly in the SDP. >>>> >>>> The problem is that the destination is behind NAT, and its reply >>>> contains a private network IP in the SDP. Normally Asterisk when nat=yes >>>> works around that by adjusting the destination for RTP to be the address it >>>> actually receives audio from, however because this device is recvonly >>>> Asterisk never receives audio from it. This means Asterisk keeps trying to >>>> send the doorbell's RTP to the private network IP which of course fails, >>>> and the destination never gets the RTP from the doorbell. >>>> >>>> Does anyone know how to work around this issue? >>>> >>>> Thank you in advance, >>>> >>>> -- >>>> David Cunningham, Voisonics Limited >>>> http://voisonics.com/ >>>> USA: +1 213 221 1092 >>>> New Zealand: +64 (0)28 2558 3782 >>>> -- >>>> _ >>>> -- 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 >> >> >> >> -- >> David Cunningham, Voisonics Limited >> http://voisonics.com/ >> USA: +1 213 221 1092 >> New Zealand: +64 (0)28 2558 3782 >> -- >> _ >> -- 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 -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] NAT problem with recvonly calls
Hi Dovid, We're using Enswitch so it uses AGI rather than a regular Asterisk dialplan, but perhaps sending it to a custom-made Asterisk context with the lines you suggest could be the best way forward. Thank you for that. On Thu, 3 Dec 2020 at 13:01, Dovid Bender wrote: > David, > > Does Asterisk send a 180 or a 183 with SDP? We have found that using these > two lines help (where xc is a 500ms blank sound file) > Exten => _X.,n, Progress() > Exten => _X.,n, Playback(xc,noanswer) > > > On Wed, Dec 2, 2020 at 4:30 PM David Cunningham > wrote: > >> Hello, >> >> We have a problem with a SIP doorbell device which sends media one way >> only, and NAT at the receiving device. >> >> When the doorbell button is pressed it makes a call to a configured >> destination. Since the doorbell only sends and doesn't receive it sends the >> INVITE with sendonly in the SDP, and the destination then replies with a >> 200 OK with recvonly in the SDP. >> >> The problem is that the destination is behind NAT, and its reply contains >> a private network IP in the SDP. Normally Asterisk when nat=yes works >> around that by adjusting the destination for RTP to be the address it >> actually receives audio from, however because this device is recvonly >> Asterisk never receives audio from it. This means Asterisk keeps trying to >> send the doorbell's RTP to the private network IP which of course fails, >> and the destination never gets the RTP from the doorbell. >> >> Does anyone know how to work around this issue? >> >> Thank you in advance, >> >> -- >> David Cunningham, Voisonics Limited >> http://voisonics.com/ >> USA: +1 213 221 1092 >> New Zealand: +64 (0)28 2558 3782 >> -- >> _ >> -- 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 -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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
