[asterisk-users] Asterisk removes SDP from 180 with SDP
Asterisk receives a 180 Ringing with SDP from the called side, then it sends 180 without SDP to the calling side. We would like asterisk to sends to the calling side the same response that was received from the called side. This is Asterisk cert 13.1, is that a new behavior, is there a setting to change this ? -- _ -- 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] PJSIP configuration for AWS/EC2 based Asterisk 13.1.0
Hello All, I have an Asterisk server v13.1.0 running on EC2 and I am able to connect and register SIP devices and see them on the asterisk CLI. I am also able to place calls, but I am not able to hear any audio on either end after the call is picked up. I was wondering if you can tell me what a minimal configuration for Asterisk on EC2 looks like. My current pjsip.conf configuration looks like this: type=transport protocol=udp bind=0.0.0.0 local_net=172.31.32.0/20 ; In the following two lines, replace publicIP with the output of ; curl -s http://169.254.169.254/latest/meta-data/public-ipv4 external_media_address=publicIP external_signaling_address=publicIP [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw direct_media=no [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!) type=aor max_contacts=1 remove_existing=yes ;Definitions for our phones, using the templates above ;; usernames and passwords etc. below My security group configuration allows TCP, UDP posrt 5060 inbound, outbound from/to 0.0.0.0/0 and TCP, UDP ports 1-2 from/to 0.0.0.0/0. Should I turn on STUN for my zoiper softphones? Any specific flavor? What am I doing wrong? Any help appreciated. -- _ -- 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] Music on hold
OK - so somebody just handed me the new music on hold file to use for the organization... Unfortunately, I was never asked about this to enough detail to be able to tell them how to set up the music, and as a result I have an eight minute file with several different messages all tied together into that one file. In general, we don't ever see a user being placed on hold for more than a minute, so using this file directly is of no use in general if I were to place it directly in to the server, as all users will only hear the first little bit of it. I suspect that when this was created, the producer assumed that the file would play in a loop, starting and stopping as callers were on hold. I realize that the streaming category will do just that, but since this is a local file, the setup works differently. (This is replacing a set of about 10 previous files that worked perfectly.) Is there any way, other than splitting up the file and trying to make decent segues between the files, to get this to work on a current version? I realize that getting it redone would be the best way, but I don't know if that is going to be an easy possibility. Any recommendations? Thanks! Kris -- _ -- 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] OT - How does the blind transfer function work on Snom-870?
Take a look at two variables you can set on your SIP peer.
TRANSFER_CONTEXT and FORWARD_CONTEXT
You should be able to use this syntax in your sip.conf
setvar=_TRANSFER_CONTEXT=kiniston-xfr
You can then create the logic you need in your dialplan to change the ring
using something like
exten = _XXX,1,SIPAddHeader(Alert-Info: Ring1)
exten = _XXX,1,Goto(INTERNAL-EXTENSIONS,${EXTEN},1)
Or you could modify your extension macro and have a test there to see if
the call has been blind transferred.with
ExecIf($[${LEN(${BLINDTRANSFER})} 0 ]?SIPAddHeader(Alert-Info: Ring1))
On Thu, Mar 5, 2015 at 8:43 AM, James B. Byrne byrn...@harte-lyne.ca
wrote:
On Thu, March 5, 2015 09:56, Ruben Rögels wrote:
Hi again,
I'm glad to hear that I provided a somehow useful answer.
Unfortunatelly, I don't know these details.
If you wasn't lucky consulting the snom docs, maybe the snom support
can be helpful with information about the exact implementation
details.
You also could use sip debug on asterisk to check what's going on
when pressing the transfer button vs. what's happening when using
## via DTMF.
Are you forced to get the transfer information from the SIP
signalling, or can you use AMI events for example? I think
this would be possible if asterisk is configured to stay in
the media path, so re-inviting is handled over asterisk itself
and therefore detectable with AMI events.
I am working with a FreePBX12/Asterisk11 setup. Asterisk stays on the
path (B2B) and there are no peer-to-peer re-invites.
What I am trying to do is to get our Snom870s to use a distinctive
ring tone when external calls are transferred internally. I have an
extension context override that detects the origin of calls and
assigns a distinctive ring to each based on ${CallerIDNum}.
But when a call is transferred then the tone does not change since the
CallerIDNum does not. An external original call always rings as if it
were coming from the outside (which it is but transferred calls have a
different handling procedure than unanswered calls). I need some way
to distinguish when the call has already been answered at least once
without changing the CallerID.
I am not worried about attended transfers since then the internal ring
tone is what should be used and that is what happens now. I just need
to deal with blind transfers.
What I have now is:
1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring1 || ring2 (depending on 1 or 2)
What I want is:
1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring3 (regardless of 1 or 2)
If everything went though ## then that would be simple enough. The
trick is that most (all) users employ the transfer button and the
touch screen to forward calls using blind transfer. But whatever
method they use to transfer I want the transfer ring tone to be the
same, albeit different from the one used for a new incoming call.
If the transfer is done using a sip message then that should be doable
as well. I just have to discover what the message is. If someone
already knows and would care to share the information then that would
be helpful. Otherwise wireshark and debug will eventually reveal it.
I may not know what I am doing. But, at least I know that I do not
know what I am doing.
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
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asterisk-users mailing list
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--
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
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Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?
Am 05.03.2015 um 15:09 schrieb James B. Byrne: On Thu, March 5, 2015 05:30, Ruben Rögels wrote: Am 05.03.2015 um 01:09 schrieb James B. Byrne: I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? Hi, I hope I understood your question correctly. AFAIK, the transfer button sends a SIP message. Entering ## on the handset is recognized via DTMF by asterisk. I hope that I understood what I was asking for. Sometimes I do not. Yes, that is what I wanted to know. Does the implementation of the transfer button feature on the Snomp-870 use exactly the same technique as the ## feature code entered through the dial pad and produce exactly the same SIP message that Asterisk produces when it gets the ## DTMF? The reason is that I wish to be able to detect a call transfer performed via either method (## or Transfer-Button) and process the result of both in the same fashion. If the button and DTMF transfers are not performed using the same switching techniques in Asterisk then I need to discover what those differences are. If both are totally equivalent from a SIP message signalling point of view then the issue is far easier to handle. I searched, in vain, in the Snom-870 docs for specifics on this and either could not find or did not recognize anything that applied. Do you know where I can locate these sorts of details. My knowledge of SIP/RTP/VOIP etc. is cursory but, given an adequate reference, I can usually sort things out. Hi again, I'm glad to hear that I provided a somehow useful answer. Unfortunatelly, I don't know these details. If you wasn't lucky consulting the snom docs, maybe the snom support can be helpful with information about the exact implementation details. You also could use sip debug on asterisk to check what's going on when pressing the transfer button vs. what's happening when using ## via DTMF. Are you forced to get the transfer information from the SIP signaling, or can you use AMI events for example? I think this would be possible if asterisk is configured to stay in the media path, so re-inviting is handled over asterisk itself and therefore detectable with AMI events. Regards, Ruben -- _ -- 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] hangup call gw FXO
2015-03-05 6:11 GMT-06:00 Steve Davies davies...@gmail.com: Looking at the pastebin, the Vega device sends a CANCEL with reason: Reason: Q.850 ;cause=16. Cause 16 is normal clearing and suggests that the original caller has disconnected. I would take a look at the Vega's logs I tried to contact support sangoma, I send a log to them and they have not contacted me! ,a disappointment asterisk shows active channels, zombie type ;) , for example the extension 160 call the 122, 122 is not connected and tells me this on the phone , I have the impression that rtptimeout not working as it should http://pastebin.com/vTZ0WGqq look cli asterisk: 200.62.89.140(None) koV6foZnHTr3gEf (nothing) No Rx: REGISTER guest 200.62.89.140(None) 690e01185aa2f36 (nothing)No Rx: REGISTER guest 200.62.89.140gatewayVEGA0010-0C09-6C8EF (ulaw) No Rx: ACKgatewayVEGA 200.62.89.140(None) 5db8c434570dfb9 (nothing)No Rx: REGISTER guest 190.184.84.10(None) 3654c4f8-1fd27d (nothing)No Rx: NOTIFY guest 5 active SIP dialogs regardss -- _ -- 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 2.10 on CentOS 5.11
I have just installed DAHDI 2.10.0.1 on a system running CentOS 5.11
(let's not get sidetracked into discussing the version of CentOS - there
are reasons for using it in this case).
The system has a TE220 card with 2xE1.
It has been working fine, but when booting, the udev startup gives out
warnings about ATTRS{hardware_id} and ATTRS{location} not being found.
After doing some research, I changed ATTRS to SYSFS in the file
/etc/udev/rules.d/dahdi.rules, and the warnings went away. Although I did
notice that after making the change and rebooting, /dev/dahdi/devices only
contained the symbolic link @Board, pointing to ../chan/001/031, and no
longer contained subdirectories called 1 and 2, each of which previously
contained 31 symbolic links, one for each channel (although it looks like
those links actually contained one too many ../).
So my questions are:
1. Was it correct on this system to change ATTRS back to SYSFS? I notice
that it was changed to ATTRS in DAHDI 2.9.
2. Should /dev/dahdi/devices contain just @Board, or also 1/ and 2/?
3. Are the /dev/dahdi/devices entries used by Asterisk or anything else?
Thanks for any advice!
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
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Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?
On Thu, March 5, 2015 09:56, Ruben Rögels wrote:
Hi again,
I'm glad to hear that I provided a somehow useful answer.
Unfortunatelly, I don't know these details.
If you wasn't lucky consulting the snom docs, maybe the snom support
can be helpful with information about the exact implementation
details.
You also could use sip debug on asterisk to check what's going on
when pressing the transfer button vs. what's happening when using
## via DTMF.
Are you forced to get the transfer information from the SIP
signalling, or can you use AMI events for example? I think
this would be possible if asterisk is configured to stay in
the media path, so re-inviting is handled over asterisk itself
and therefore detectable with AMI events.
I am working with a FreePBX12/Asterisk11 setup. Asterisk stays on the
path (B2B) and there are no peer-to-peer re-invites.
What I am trying to do is to get our Snom870s to use a distinctive
ring tone when external calls are transferred internally. I have an
extension context override that detects the origin of calls and
assigns a distinctive ring to each based on ${CallerIDNum}.
But when a call is transferred then the tone does not change since the
CallerIDNum does not. An external original call always rings as if it
were coming from the outside (which it is but transferred calls have a
different handling procedure than unanswered calls). I need some way
to distinguish when the call has already been answered at least once
without changing the CallerID.
I am not worried about attended transfers since then the internal ring
tone is what should be used and that is what happens now. I just need
to deal with blind transfers.
What I have now is:
1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring1 || ring2 (depending on 1 or 2)
What I want is:
1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring3 (regardless of 1 or 2)
If everything went though ## then that would be simple enough. The
trick is that most (all) users employ the transfer button and the
touch screen to forward calls using blind transfer. But whatever
method they use to transfer I want the transfer ring tone to be the
same, albeit different from the one used for a new incoming call.
If the transfer is done using a sip message then that should be doable
as well. I just have to discover what the message is. If someone
already knows and would care to share the information then that would
be helpful. Otherwise wireshark and debug will eventually reveal it.
I may not know what I am doing. But, at least I know that I do not
know what I am doing.
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
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asterisk-users mailing list
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Re: [asterisk-users] Understanding the right way to get started with multiple trunks/extensions
In the 'home-number' example that was provided the caller ID was being
replaced with the string 'Home'
It's easy to prepend the caller ID instead however.
Set(CALLERID(name)=Home-${CALLERID(name)})
You could even get fancy and set it based on what number was called, This
would prepend the CallerID with the last 4 digits of the incoming number
assuming that your calls come in to an extension that way:
Set(CALLERID(name)=${CDR(firstext):-4}-${CALLERID(name)})
There is no 'Best' or 'Better' way to handle extension and voicemail
routing, It's all down to your preference as a programmer and your users.
Try things, Find what works best for you, The only thing you have to loose
is your free time and if you are like me you will have fun during the
process.
On Thu, Mar 5, 2015 at 5:54 AM, Mark Rogers m...@more-solutions.co.uk
wrote:
For some reason I didn't see David's reply by email, and have
copy/pasted the following from the list archives to make my reply,
sorry if that messes up anyone's threading.
On 4 March 2015 at 12:15, David Duffett wrote:
If you would like to set things up via the GUI on your incredible PBX,
[...]
I'm trying to avoid a GUI for now so that I learn something, but
knowing how to do it that way is appreciated, thanks.
If, on the other hand, you want to achieve your aim through native
configuration files, you could add a line like:
exten = *home-number*,1,Set(CALLERID(name)=Home)
exten = *home-number*,n,*continue handling call as you were before*
I haven't got my head round the syntax yet; will this retain the real
caller ID but add something to it, or will I lose the real ID?
From your answers I take it that it is better if all users have
their own extension and I route calls to the relevant extensions as
required, rather than having users monitor multiple extensions. It's
what I expected but can I ask why? Is it a scalability or performance
issue, do I lose something by not doing it this way, or is it just
about doing things right?
With the above, what's the best way to handle voicemail? I would
expect anyone who could have taken the call to be able to access the
voicemail, and once one person has dealt with a message it's no
longer a new message to anyone else.
Mark
--
Mark Rogers // More Solutions Ltd (Peterborough Office) // 0844 251 1450
Registered in England (0456 0902) 21 Drakes Mews, Milton Keynes, MK8 0ER
--
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--
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
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Re: [asterisk-users] hangup call gw FXO
On Wednesday, March 4, 2015, ricky gutierrez xserverli...@gmail.com wrote: I'm having some problems with a vega sangoma, if a call comes into my ivr and hangs up, the call continues to ring and leaves hanging the channel, I have to restart Asterisk and everything works Ok my sangoma is a vega 50 , 4 FXO . I tried different tone of countries and does not work, this is the trace of which is for hanging up the channel: http://pastebin.com/y410Rhzt I was thinking that might help rpt timeout , I have put in 30s, but does not work any advice? regardss something strange, I have some extensions not connected to Asterisk and if I call, I get the message busy, the version I'm using is asterisk 11.15 -- rickygm http://gnuforever.homelinux.com -- _ -- 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] Understanding the right way to get started with multiple trunks/extensions
For some reason I didn't see David's reply by email, and have copy/pasted the following from the list archives to make my reply, sorry if that messes up anyone's threading. On 4 March 2015 at 12:15, David Duffett wrote: If you would like to set things up via the GUI on your incredible PBX, [...] I'm trying to avoid a GUI for now so that I learn something, but knowing how to do it that way is appreciated, thanks. If, on the other hand, you want to achieve your aim through native configuration files, you could add a line like: exten = *home-number*,1,Set(CALLERID(name)=Home) exten = *home-number*,n,*continue handling call as you were before* I haven't got my head round the syntax yet; will this retain the real caller ID but add something to it, or will I lose the real ID? From your answers I take it that it is better if all users have their own extension and I route calls to the relevant extensions as required, rather than having users monitor multiple extensions. It's what I expected but can I ask why? Is it a scalability or performance issue, do I lose something by not doing it this way, or is it just about doing things right? With the above, what's the best way to handle voicemail? I would expect anyone who could have taken the call to be able to access the voicemail, and once one person has dealt with a message it's no longer a new message to anyone else. Mark -- Mark Rogers // More Solutions Ltd (Peterborough Office) // 0844 251 1450 Registered in England (0456 0902) 21 Drakes Mews, Milton Keynes, MK8 0ER -- _ -- 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] OT - How does the blind transfer function work on Snom-870?
Am 05.03.2015 um 01:09 schrieb James B. Byrne: I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? Hi, I hope I understood your question correctly. AFAIK, the transfer button sends a SIP message. Entering ## on the handset is recognized via DTMF by asterisk. -- _ -- 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] hangup call gw FXO
Looking at the pastebin, the Vega device sends a CANCEL with reason: Reason: Q.850 ;cause=16. Cause 16 is normal clearing and suggests that the original caller has disconnected. I would take a look at the Vega's logs Regards, Steve On Thu, 5 Mar 2015 at 11:41 ricky gutierrez xserverli...@gmail.com wrote: On Wednesday, March 4, 2015, ricky gutierrez xserverli...@gmail.com wrote: I'm having some problems with a vega sangoma, if a call comes into my ivr and hangs up, the call continues to ring and leaves hanging the channel, I have to restart Asterisk and everything works Ok my sangoma is a vega 50 , 4 FXO . I tried different tone of countries and does not work, this is the trace of which is for hanging up the channel: http://pastebin.com/y410Rhzt I was thinking that might help rpt timeout , I have put in 30s, but does not work any advice? regardss something strange, I have some extensions not connected to Asterisk and if I call, I get the message busy, the version I'm using is asterisk 11.15 -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?
On Thu, March 5, 2015 05:30, Ruben Rögels wrote: Am 05.03.2015 um 01:09 schrieb James B. Byrne: I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? Hi, I hope I understood your question correctly. AFAIK, the transfer button sends a SIP message. Entering ## on the handset is recognized via DTMF by asterisk. I hope that I understood what I was asking for. Sometimes I do not. Yes, that is what I wanted to know. Does the implementation of the transfer button feature on the Snomp-870 use exactly the same technique as the ## feature code entered through the dial pad and produce exactly the same SIP message that Asterisk produces when it gets the ## DTMF? The reason is that I wish to be able to detect a call transfer performed via either method (## or Transfer-Button) and process the result of both in the same fashion. If the button and DTMF transfers are not performed using the same switching techniques in Asterisk then I need to discover what those differences are. If both are totally equivalent from a SIP message signalling point of view then the issue is far easier to handle. I searched, in vain, in the Snom-870 docs for specifics on this and either could not find or did not recognize anything that applied. Do you know where I can locate these sorts of details. My knowledge of SIP/RTP/VOIP etc. is cursory but, given an adequate reference, I can usually sort things out. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] PJSIP configuration for AWS/EC2 based Asterisk 13.1.0
OK. I think I found the issue. The key is to add rtp_symmetric=yes Here's what my final configuration looks like: [transport-udp] type=transport protocol=udp bind=0.0.0.0 ;; for within EC2 local_net=172.31.32.0/20 ;; For softphones within EC2 local_net=192.168.1.0/24 external_media_address=publicIPOfEC2Instance external_signaling_address=publicIPOfEC2Instance ;Templates for the necessary config sections [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=!all,ulaw direct_media=no rtp_symmetric=yes On Thu, Mar 5, 2015 at 5:52 PM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: Hello All, I have an Asterisk server v13.1.0 running on EC2 and I am able to connect and register SIP devices and see them on the asterisk CLI. I am also able to place calls, but I am not able to hear any audio on either end after the call is picked up. I was wondering if you can tell me what a minimal configuration for Asterisk on EC2 looks like. My current pjsip.conf configuration looks like this: type=transport protocol=udp bind=0.0.0.0 local_net=172.31.32.0/20 ; In the following two lines, replace publicIP with the output of ; curl -s http://169.254.169.254/latest/meta-data/public-ipv4 external_media_address=publicIP external_signaling_address=publicIP [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw direct_media=no [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!) type=aor max_contacts=1 remove_existing=yes ;Definitions for our phones, using the templates above ;; usernames and passwords etc. below My security group configuration allows TCP, UDP posrt 5060 inbound, outbound from/to 0.0.0.0/0 and TCP, UDP ports 1-2 from/to 0.0.0.0/0. Should I turn on STUN for my zoiper softphones? Any specific flavor? What am I doing wrong? Any help appreciated. -- _ -- 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] Music on hold
Just split the file into multiple files n have it all uploaded to the same music on hold class. Now every time a caller is put on hold they will hear the files randomly. On 06-Mar-2015 8:32 AM, Kris Stark kris.st...@godataflow.com wrote: OK - so somebody just handed me the new music on hold file to use for the organization... Unfortunately, I was never asked about this to enough detail to be able to tell them how to set up the music, and as a result I have an eight minute file with several different messages all tied together into that one file. In general, we don't ever see a user being placed on hold for more than a minute, so using this file directly is of no use in general if I were to place it directly in to the server, as all users will only hear the first little bit of it. I suspect that when this was created, the producer assumed that the file would play in a loop, starting and stopping as callers were on hold. I realize that the streaming category will do just that, but since this is a local file, the setup works differently. (This is replacing a set of about 10 previous files that worked perfectly.) Is there any way, other than splitting up the file and trying to make decent segues between the files, to get this to work on a current version? I realize that getting it redone would be the best way, but I don't know if that is going to be an easy possibility. Any recommendations? Thanks! Kris -- _ -- 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
