Re: [asterisk-users] using analog phones
On 20 Aug 2012, at 01:37, Noam Birnbaum wrote: A client wants to keep their old Inter-Tel KTS analog phones for budget reasons. Two questions: 1. How could they use these with FreePBX? You'd need an ATA or an analogue card. And how well they work, depends on if they are 'true' analogue, or have extra digital pairs etc. 2. Would they be losing any features that they currently have with their analog PBX? You'd need to find out what features they have on their PBX, and compare this to FreePBX. 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-users
Re: [asterisk-users] using analog phones
u need an analog telephone adaptor to connect your analog phone with switch hub. but the device is rather expensive. Sent from Samsung Mobile Noam Birnbaum n...@maccentricsolutions.com wrote: Hi folks, A client wants to keep their old Inter-Tel KTS analog phones for budget reasons. Two questions: 1. How could they use these with FreePBX? 2. Would they be losing any features that they currently have with their analog PBX? Thanks! Noam Birnbaum Mac Daddy http://www.maccentricsolutions.com 877.luv.macs x666 tweet @noamb Tech support — 877.luv.macs or supp...@maccentricsolutions.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] using analog phones
On Monday 20 August 2012, Noam Birnbaum wrote: Hi folks, A client wants to keep their old Inter-Tel KTS analog phones for budget reasons. Two questions: 1. How could they use these with FreePBX? Via so-called Analogue Terminal Adaptors (ATAs). These are basically a SIP phone with an FXS interface instead of a handset and keypad. Alternatively, up to four analogue phones could be connected through something like a TDM400P fitted with four FXS modules. Note that ATAs are not cheap! Replacing the analogue phones with SIP phones might actually work out less expensive. 2. Would they be losing any features that they currently have with their analog PBX? This is very unlikely. Check the feature set to be sure. Asterisk is basically a telephony construction kit; so the chances are, any feature they seem to be missing can be added, either in the dialplan or with an AGI script. -- AJS Answers come *after* questions. -- _ -- 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] using analog phones
On Sun, Aug 19, 2012 at 8:37 PM, Noam Birnbaum n...@maccentricsolutions.com wrote: Hi folks, A client wants to keep their old Inter-Tel KTS analog phones for budget reasons. Two questions: 1. How could they use these with FreePBX? 2. Would they be losing any features that they currently have with their analog PBX? Thanks! Noam Birnbaum Mac Daddy http://www.maccentricsolutions.com 877.luv.macs x666 tweet @noamb Tech support — 877.luv.macs or supp...@maccentricsolutions.com I have done a good deal installations where analog handsets, headsets, or many faxes just made sense. I have used T1 cards to connect to 24 port channel banks to provide true analog dialtone. I have also used SIP based channel banks with great success but never bothered messing with the fax issues. FreePBX should give you way more functionality than their current PBX, sometimes it just takes a little creativity and doing things differently. Always get the details of what they currently use. An admin assistant or secretary is usually the person that knows what bells and whistles everyone uses. That is the person that you really want to befriend for a smooth implementation. Finally, I have installed a handful of Inter Tel/Mitel PBXen along the way, and the phones I installed were infact digital and proprietary. Check the model number of the phones to see if they are truly analog, which I doubt. In that case, you can still use a FreePBX box, but you are going to have to integrate it with the Inter Tel, I usually put Asterisk in front and just timeout your ring to VM a little bit shorter on the Asterisk box than the Inter Tel. With phone prices what they are, and a very good Polycom (or whatever you like) SIP phone so cheap, I would seriously consider ditching the phones if you find that they are digital. Thanks, Steve Totaro -- _ -- 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 hangs while starting in OpenSuse 12.2
Its really weird working with OpenSuse. I am not sure how others are using with OpenSuse. Through Yast also I tried to install Asterisk package, it didn't find. Now I am clueless to work with OpenSuse. Regards. On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.com should resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- __**__**_ -- 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 -- _ -- 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 hangs while starting in OpenSuse 12.2
From the forum I understand OpenSuse 12.2 is pre-relase and better to use OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Its really weird working with OpenSuse. I am not sure how others are using with OpenSuse. Through Yast also I tried to install Asterisk package, it didn't find. Now I am clueless to work with OpenSuse. Regards. On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.comshould resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- __**__** _ -- 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 -- _ -- 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] DTMF Issue.
Hi I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of ATA on the network who autenticate the phones: Linksys PAP2, Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at the same network all with g729 codecs and rfc2833 for the DTMF. Making calls via the Overtek ATA the DTMF works fine but at the others ATA it doesn't. My config: - asterisk 1.6.2.13 - dahdi 2.3.0.1 - The phones connected are all physical phones -- Att.* *** -- _ -- 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] BLF and Call Queues
On 19-08-12 21:58, Alec Davis wrote: So I'm just looking on how to make a BLF-button blink or turn red, to show to my customer that there are still calls inside the queue waiting. Can I only apply on Asterisk 1.8.5 ? Or can I apply to my Asterisk 1.8.11 also ? It's 4 lines, plus 2 debug statements. I haven't had time to see if it applies clean against 1.8.5. We are running it with 1.8 branch. Alec So I can just add these 4 lines to app_queue.c and this will give me the ability to use : exten = 566,hint,Queue:voipq1 ?? Kind regards, Jonas. -- _ -- 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] DTMF Issue.
On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote: Hi I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of ATA on the network who autenticate the phones: Linksys PAP2, Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at the same network all with g729 codecs and rfc2833 for the DTMF. Making calls via the Overtek ATA the DTMF works fine but at the others ATA it doesn't. My config: - asterisk 1.6.2.13 - dahdi 2.3.0.1 - The phones connected are all physical phones There is additional data you can provide to make it easier for others to help out: If you can pastebin an Asterisk log including all message types plus VERBOSE,DEBUG,DTMF [1] during a working call and a failed call that would be very helpful. A step beyond that is to also provide a SIP and RTP packet trace so that whoever wants to help can look through it in Wireshark. If you can get the packet trace for the same calls you gather log data for, that would be best. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Rusty Newton Digium, Inc | Open Source Community Support Manager 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
[asterisk-users] Asterisk as TLS server as well as TLS client
Hi, I have to connect 3 asterisk servers,each of them being TLS server for his clients and connected in both way in TLS with both others asterisk, each having hi own Common Name. Is this possible? I set up 2 asterik's , one server and the other client, this is OK. But I can't deal with certificats generated on both servers. I tried to put tlscertfile ans tlscafile in the peer definition, each pointing to the certificate generated by the server, but thatś not working. Thanks for any hint. -- 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
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br wrote: On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro jcas...@instant.com.br wrote: I still get unauthorized from sipml5 with these modifications. I used port 80 instead of 8088 (no other webserver listening on 80), was that wrong? Correction. It's actually Failed to connect to the server. I set the proxy address and port correctly in sipml5's call.htm (it registers on Kamailio). ...which is in fact a 404 response from Asterisk. Here's the response I received: http://users.vialink.com.br/jcastro/refused.cap I suspect I am configuring something wrong, but what is it? Juan -- _ -- 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] How to test Websocket support in SIP in Asterisk trunk?
- Original Message - Joshua Can you copy and past into a wiki page for everyone's benefit? Maybe https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or like page would be good. If this thread has taught me anything it's that there needs to be a complete wiki page, just copying/pasting what I'm saying here isn't enough. It's on my list. I won't call it a demo setup though... since it won't actually work yet. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
- Original Message - On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br wrote: On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro jcas...@instant.com.br wrote: I still get unauthorized from sipml5 with these modifications. I used port 80 instead of 8088 (no other webserver listening on 80), was that wrong? Correction. It's actually Failed to connect to the server. I set the proxy address and port correctly in sipml5's call.htm (it registers on Kamailio). ...which is in fact a 404 response from Asterisk. Here's the response I received: http://users.vialink.com.br/jcastro/refused.cap I suspect I am configuring something wrong, but what is it? The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
- Original Message - On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br wrote: On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro jcas...@instant.com.br wrote: I still get unauthorized from sipml5 with these modifications. I used port 80 instead of 8088 (no other webserver listening on 80), was that wrong? Correction. It's actually Failed to connect to the server. I set the proxy address and port correctly in sipml5's call.htm (it registers on Kamailio). ...which is in fact a 404 response from Asterisk. Here's the response I received: http://users.vialink.com.br/jcastro/refused.cap Well, of course unless you changed the port as you did in which case 80 in the URL instead of 8088. That is all! As I've said previously though, you won't get bidirectional audio or video flowing so trying that will fail, and it's known that it will fail. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Mon, Aug 20, 2012 at 10:52 AM, Joshua Colp jc...@digium.com wrote: - Original Message - Joshua Can you copy and past into a wiki page for everyone's benefit? Maybe https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or like page would be good. If this thread has taught me anything it's that there needs to be a complete wiki page, just copying/pasting what I'm saying here isn't enough. It's on my list. I won't call it a demo setup though... since it won't actually work yet. -- Joshua Colp Digium, Inc. | Software 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 Agreed, but we need something and a place for comments. The wiki is great because we can rename and move things when they are no longer relevant to our needs. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.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
Re: [asterisk-users] Asterisk as TLS server as well as TLS client
On 20/08/12 16:23, Administrator TOOTAI wrote: Hi, I have to connect 3 asterisk servers,each of them being TLS server for his clients and connected in both way in TLS with both others asterisk, each having hi own Common Name. Is this possible? I set up 2 asterik's , one server and the other client, this is OK. But I can't deal with certificats generated on both servers. I tried to put tlscertfile ans tlscafile in the peer definition, each pointing to the certificate generated by the server, but thatś not working. Thanks for any hint. Asterisk doesn't seem to implement mutual TLS authentication, see the comments in this thread: http://java.net/projects/jitsi/lists/users/archive/2012-08/message/37 People who want strong TLS typically use a SIP proxy as a front-end to Asterisk, either repro or Kamailio stand out as leaders in TLS support http://www.opentelecoms.org/use-a-sip-proxy-instead-of-asterisk At the bottom, there are links to some practical guides how to use either repro or Kamailio with 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
[asterisk-users] Digium Phones
I have been looking for the specs (format, bit rate, ect) on custom ringtones for digium phones. Using the DPMA how would I deliver the ringtone to a digium phone? -- _ -- 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] Digium Phones
On 8/20/2012 10:14 AM, Josh Hopkins wrote: I have been looking for the specs (format, bit rate, ect) on custom ringtones for digium phones. Using the DPMA how would I deliver the ringtone to a digium phone? https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration All the answers you seek are located there. Ctrl + f for ringtone throughout the document. Also, 16-bit, 16kHz mono .wav according to the wiki. -- Rusty Newton Digium, Inc | Open Source Community Support Manager Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Phones
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Rusty Newton Sent: Monday, August 20, 2012 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium Phones On 8/20/2012 10:14 AM, Josh Hopkins wrote: I have been looking for the specs (format, bit rate, ect) on custom ringtones for digium phones. Using the DPMA how would I deliver the ringtone to a digium phone? https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration All the answers you seek are located there. Ctrl + f for ringtone throughout the document. Also, 16-bit, 16kHz mono .wav according to the wiki. -- Rusty Newton Digium, Inc | Open Source Community Support Manager 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 [Josh Hopkins] Thanks for the reply. I do see it now. Not sure how I missed all that before. One thing I don't think I did see was where I should place my ringtone file? I am guessing that if I put it in the same location as our custom imagefile that it might work. Is there a url_prefix option like there is for firmware? -- _ -- 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] How to test Websocket support in SIP in Asterisk trunk?
On Mon, Aug 20, 2012 at 11:53 AM, Joshua Colp jc...@digium.com wrote:
- Original Message -
On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br
wrote:
On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro
jcas...@instant.com.br wrote:
I still get unauthorized from sipml5 with these modifications. I
used port 80 instead of 8088 (no other webserver listening on 80),
was
that wrong?
Correction. It's actually Failed to connect to the server. I set
the
proxy address and port correctly in sipml5's call.htm (it registers
on
Kamailio).
...which is in fact a 404 response from Asterisk. Here's the response
I received: http://users.vialink.com.br/jcastro/refused.cap
I suspect I am configuring something wrong, but what is it?
The complete URL to use is http://asterisk IP address or host:8088/ws
Note the /ws at the end. WebSocket support is only available there. Doing
otherwise would have required core HTTP server changes, which I wanted to
avoid. Depending on what you are testing with you may need to change it
slightly to add that in.
Well, I did the following changes in sipml5 and now I get a Bad
Request on REGISTER, instead of 404. Clearly, I'm still missing
something. Here are the changes I made:
Index: call.htm
===
--- call.htm(revision 68)
+++ call.htm(working copy)
@@ -351,8 +351,9 @@
// we will connect to one of them and let the
balancer to choose the right one (less connected sockets)
// each port can accept up to 65K connections which
means that the cloud can manage 325K active connections
// the number of port will be increased or decreased
based on the current trafic
-i_port = 4062 + (((new Date().getTime()) % 5) * 1000);
-s_proxy = sipml5.org;
+// i_port = 4062 + (((new Date().getTime()) % 5) * 1000);
+i_port = 80;
+s_proxy = 192.168.0.111;
}
// create a new SIP stack. Not mandatory as it's possible
to reuse the same satck
Index: src/tinySIP/src/tsip_stack.js
===
--- src/tinySIP/src/tsip_stack.js (revision 68)
+++ src/tinySIP/src/tsip_stack.js (working copy)
@@ -351,7 +351,7 @@
return -2;
}
-tsk_utils_log_info(SIP stack start: proxy=' +
this.network.s_proxy_cscf_host + : + this.network.i_proxy_cscf_port
+ ', realm=' + this.network.o_uri_realm + ', impi=' +
this.identity.s_impi + ', impu=' + this.identity.o_uri_impu + ');
+tsk_utils_log_info(SIP stack start: proxy=' +
this.network.s_proxy_cscf_host + : + this.network.i_proxy_cscf_port
+ /ws', realm=' + this.network.o_uri_realm + ', impi=' +
this.identity.s_impi + ', impu=' + this.identity.o_uri_impu + ');
this.network.o_transport =
this.o_layer_transport.transport_new(this.network.e_proxy_cscf_type,
this.network.s_proxy_cscf_host, this.network.i_proxy_cscf_port, SIP
Transport, __tsip_stack_transport_callback);
if (!this.network.o_transport) {
@@ -716,4 +716,4 @@
}
return 0;
-}
\ No newline at end of file
+}
Index: src/tinySIP/src/transports/tsip_transport.js
===
--- src/tinySIP/src/transports/tsip_transport.js(revision 68)
+++ src/tinySIP/src/transports/tsip_transport.js(working copy)
@@ -368,7 +368,7 @@
return -1;
}
-var s_url = tsk_string_format({0}://{1}:{2},o_self.s_protocol,
o_self.s_host, o_self.i_port);
+var s_url =
tsk_string_format({0}://{1}:{2}/ws,o_self.s_protocol, o_self.s_host,
o_self.i_port);
tsk_utils_log_info(Connecting to '+s_url+');
o_self.o_ws = new WebSocket(s_url, 'sip');
o_self.o_ws.binaryType = arraybuffer;
@@ -458,7 +458,7 @@
}
var b_isInternetExplorer = (WebRtc4all_GetType() == WebRtcType_e.IE);
-var s_url = tsk_string_format({0}://{1}:{2},o_self.s_protocol,
o_self.s_host, o_self.i_port);
+var s_url =
tsk_string_format({0}://{1}:{2}/ws,o_self.s_protocol, o_self.s_host,
o_self.i_port);
tsk_utils_log_info(Connecting to '+s_url+');
if(b_isInternetExplorer){
o_self.o_transport = new ActiveXObject(webrtc4ie.NetTransport);
@@ -480,7 +480,7 @@
if(o_self.o_transport.defaultDestAddr
o_self.o_transport.defaultDestPort){
o_self.s_host = o_self.o_transport.defaultDestAddr;
o_self.i_port = o_self.o_transport.defaultDestPort;
-tsk_utils_log_info(Transport default destination= +
o_self.s_host + : + o_self.i_port);
+tsk_utils_log_info(Transport default destination= +
o_self.s_host + : + o_self.i_port + /ws);
}
o_self.b_started = true;
o_self.signal(tsip_transport_event_type_e.STARTED, Network
transport started, null);
Index: src/tinyMEDIA/src/tmedia_session_jsep.js
Re: [asterisk-users] Digium Phones
On 8/20/2012 10:31 AM, Josh Hopkins wrote: [Josh Hopkins] Thanks for the reply. I do see it now. Not sure how I missed all that before. One thing I don't think I did see was where I should place my ringtone file? I am guessing that if I put it in the same location as our custom imagefile that it might work. Is there a url_prefix option like there is for firmware? Looks like it. Copy paste from the wiki page below: Ringtone Configuration Options A Ringtone defines an actual ringing tone to by used by a phone. When the phone rings, it's what's played over the phone's speaker. Option Values Description alias string An identifier for the ringtone, e.g. FancyRinger. This identifier will be present in the phone's preferences menu and in its web menu filename string A named 16-bit 16kHz mono .wav file as reachable from the file_url_prefix. Example In this example: The ringtone is identified as FancyRinger Example Ringtone Configuration. [fancyringer] type=ringtone alias=FancyRinger filename=FancyRinger.wav -- Rusty Newton Digium, Inc | Open Source Community Support Manager Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
- Original Message - The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. Well, I did the following changes in sipml5 and now I get a Bad Request on REGISTER, instead of 404. Clearly, I'm still missing something. Here are the changes I made: You are probably getting hit by a bug in Asterisk 11 that has been fixed. It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support along with a work around via configuration. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using analog phones
From: Noam Birnbaum n...@maccentricsolutions.com Sent: Sunday, August 19, 2012 8:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] using analog phones Hi folks, A client wants to keep their old Inter-Tel KTS analog phones for budget reasons. Two questions: 1. How could they use these with FreePBX? 2. Would they be losing any features that they currently have with their analog PBX? Thanks! Noam Birnbaum Mac Daddy http://www.maccentricsolutions.com 877.luv.macs x666 tweet @noamb Tech support - 877.luv.macs or supp...@maccentricsolutions.com Noam By the time you outfit the customer with analog gateways you are much better off with IP phones. You don't get all the nice features of FreePBX without dialing codes using analog phones. Also are the analog phones True FXS PSTN phones or are they specfic to their old system? How many phones do they need. I have six cisco IP 303 phones that I traded-in from a customer that wanted to step-up to our nicer 4 line Grandstream phones. We would be willing to make a good deal on the six as we do not use cisco. Contact me off list if you are interested in the phones. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] DTMF Issue.
Thanks for your answer. The logs where posted at pastebin, here the links: - Working Phone: http://pastebin.com/q3pHcwna - Not working phone: http://pastebin.com/iiCHPMmn 2012/8/20 Rusty Newton rnew...@digium.com On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote: Hi I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of ATA on the network who autenticate the phones: Linksys PAP2, Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at the same network all with g729 codecs and rfc2833 for the DTMF. Making calls via the Overtek ATA the DTMF works fine but at the others ATA it doesn't. My config: - asterisk 1.6.2.13 - dahdi 2.3.0.1 - The phones connected are all physical phones There is additional data you can provide to make it easier for others to help out: If you can pastebin an Asterisk log including all message types plus VERBOSE,DEBUG,DTMF [1] during a working call and a failed call that would be very helpful. A step beyond that is to also provide a SIP and RTP packet trace so that whoever wants to help can look through it in Wireshark. If you can get the packet trace for the same calls you gather log data for, that would be best. Thanks! [1] https://wiki.asterisk.org/**wiki/display/AST/Collecting+** Debug+Informationhttps://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Rusty Newton Digium, Inc | Open Source Community Support Manager 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 -- Att.* *** -- _ -- 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] using analog phones
Thanks to all who applied. I found out that their Inter-Tel phones are connected to an Inter-Tel Axxess box-thingamadoodle (that's a technical term). Inter-Tel was purchased by Mitel in 2009 so there's no support left. I think I'm gonna recommend they use IP phones or a different VoIP consultant…! Best, noam On Aug 20, 2012, at 9:04 AM, Bryant Zimmerman wrote: From: Noam Birnbaum n...@maccentricsolutions.com Sent: Sunday, August 19, 2012 8:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] using analog phones Hi folks, A client wants to keep their old Inter-Tel KTS analog phones for budget reasons. Two questions: 1. How could they use these with FreePBX? 2. Would they be losing any features that they currently have with their analog PBX? Thanks! Noam Birnbaum Mac Daddy http://www.maccentricsolutions.com 877.luv.macs x666 tweet @noamb Tech support — 877.luv.macs or supp...@maccentricsolutions.com Noam By the time you outfit the customer with analog gateways you are much better off with IP phones. You don't get all the nice features of FreePBX without dialing codes using analog phones. Also are the analog phones True FXS PSTN phones or are they specfic to their old system? How many phones do they need. I have six cisco IP 303 phones that I traded-in from a customer that wanted to step-up to our nicer 4 line Grandstream phones. We would be willing to make a good deal on the six as we do not use cisco. Contact me off list if you are interested in the phones. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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 Noam Birnbaum Mac Daddy http://www.maccentricsolutions.com 877.luv.macs x666 tweet @noamb Tech support — 877.luv.macs or supp...@maccentricsolutions.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
[asterisk-users] Asterisk 11 queue calls - emulate Dial(b) functionality
I currently run an Asterisk 10 system with hotdesking functionality set up. Several of the users have worked with a system in the past that supported BLF on their IP phones, and would like their current phones to behave in a similar fashion. Right now I have a really kludgy system that mostly works, but doesn't consistently trigger the cleanup macro to clear the device state on the end of a call. Rather than continue to beat my head against the wall playing which context isn't firing an h extension to dump calls into the cleanup macro, I decided to investigate Asterisk 11 for the new Dial() b function and the new hangup handler CHANNEL variable. I have the hints working more or less correctly on direct calls to/from the phones, making use of the b and U functions in Dial() and some judicious use of GROUP channel variables and CHANNEL(hangup_handler_wipe). But, on my live system, sometimes the users receive calls from a queue, and I don't see any way with the queue calls to emulate the b functionality in Dial() to be able to set the agent extension's device state to RINGING when the queue call gets created. Obviously, I can use membergosub to set the agent to INUSE after they pick up the call (like Dial() U), but is there anything that I can use to manipulate the channel that is calling the agent while/before it is ringing? Thank you, Noah Engelberth MetaLINK Technologies -- _ -- 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] How to test Websocket support in SIP in Asterisk trunk?
Hoo-hah. It registers. Progress! Now... media. Or not. On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote: - Original Message - The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. Well, I did the following changes in sipml5 and now I get a Bad Request on REGISTER, instead of 404. Clearly, I'm still missing something. Here are the changes I made: You are probably getting hit by a bug in Asterisk 11 that has been fixed. It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support along with a work around via configuration. -- Joshua Colp Digium, Inc. | Software 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 -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- _ -- 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] How to test Websocket support in SIP in Asterisk trunk?
On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro jcas...@instant.com.br wrote: Hoo-hah. It registers. Progress! Now... media. Or not. On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote: - Original Message - The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. Well, I did the following changes in sipml5 and now I get a Bad Request on REGISTER, instead of 404. Clearly, I'm still missing something. Here are the changes I made: You are probably getting hit by a bug in Asterisk 11 that has been fixed. It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support along with a work around via configuration. -- Joshua Colp Digium, Inc. | Software 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 -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- Juan Matt just opened https://issues.asterisk.org/jira/browse/ASTERISK-20267 to document some of this. Feel free to pipe in. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.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
Re: [asterisk-users] Asterisk as TLS server as well as TLS client
This is all nice and good but the documentation all assumes that you are on a Debian box and use MYSQL. What about us SUSE/Postgresql folks? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Pocock Sent: Monday, August 20, 2012 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk as TLS server as well as TLS client On 20/08/12 16:23, Administrator TOOTAI wrote: Hi, I have to connect 3 asterisk servers,each of them being TLS server for his clients and connected in both way in TLS with both others asterisk, each having hi own Common Name. Is this possible? I set up 2 asterik's , one server and the other client, this is OK. But I can't deal with certificats generated on both servers. I tried to put tlscertfile ans tlscafile in the peer definition, each pointing to the certificate generated by the server, but thatś not working. Thanks for any hint. Asterisk doesn't seem to implement mutual TLS authentication, see the comments in this thread: http://java.net/projects/jitsi/lists/users/archive/2012-08/message/37 People who want strong TLS typically use a SIP proxy as a front-end to Asterisk, either repro or Kamailio stand out as leaders in TLS support http://www.opentelecoms.org/use-a-sip-proxy-instead-of-asterisk At the bottom, there are links to some practical guides how to use either repro or Kamailio with 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 -- _ -- 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 11 - BLF on Custom devices
In testing Asterisk 11, I've found that Asterisk doesn't seem to be sending BLF
updates to SIP peers that have subscribed to a hint looking at a Custom device
if that Custom device state is RINGING or RING_INUSE. All other states seem to
be working correctly.
The hint section of the dialplan is:
[hints]
exten = _3XX,hint,Custom:${EXTEN}
Console shows the following for core show hints with no calls:
-= Registered Asterisk Dial Plan Hints =-
_3XX@hints : Custom:${EXTEN} State:Idle
Watchers 0
302@hints : Custom:302State:Idle
Watchers 2
303@hints : Custom:303State:Idle
Watchers 2
301@hints : Custom:301State:Idle
Watchers 2
And with a ringing call (301 calling 302):
-= Registered Asterisk Dial Plan Hints =-
_3XX@hints : Custom:${EXTEN} State:Idle
Watchers 0
302@hints : Custom:302
State:Ringing Watchers 2
303@hints : Custom:303State:Idle
Watchers 2
301@hints : Custom:301State:InUse
Watchers 2
And after 302 picks up (301 and 302 on a call):
-= Registered Asterisk Dial Plan Hints =-
_3XX@hints : Custom:${EXTEN} State:Idle
Watchers 0
302@hints : Custom:302State:InUse
Watchers 2
303@hints : Custom:303State:Idle
Watchers 2
301@hints : Custom:301State:InUse
Watchers 2
And after 303 tries to call 302 while 301 302 are still on a call (301 302
on a call, plus 303 calling 302):
-= Registered Asterisk Dial Plan Hints =-
_3XX@hints : Custom:${EXTEN} State:Idle
Watchers 0
302@hints : Custom:302
State:InUseRinging Watchers 2
303@hints : Custom:303State:InUse
Watchers 2
301@hints : Custom:301State:InUse
Watchers 2
But despite the above, the BLF fields on my phones (Cisco SPA 509G for all 3
extensions) only update for Idle or InUse - they do not show the Ringing or
InUseRinging statuses. I have verified the SPA phones BLFs do still show the
correct Ringing and InUseRinging statuses if they subscribe directly to a SIP
device's state with the hint - the issue only seems to be effecting Custom
devices. Can anyone think of anything else I should check?
Thank you,
Noah Engelberth
MetaLINK Technologies
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Re: [asterisk-users] Asterisk 11 queue calls - emulate Dial(b) functionality
I currently run an Asterisk 10 system with hotdesking functionality set up. Several of the users have worked with a system in the past that supported BLF on their IP phones, and would like their current phones to behave in a similar fashion. Right now I have a really kludgy system that mostly works, but doesn’t consistently trigger the cleanup macro to “clear” the device state on the end of a call. Rather than continue to beat my head against the wall playing “which context isn’t firing an h extension to dump calls into the cleanup macro”, I decided to investigate Asterisk 11 for the new Dial() b function and the new hangup handler CHANNEL variable. I have the hints working more or less correctly on direct calls to/from the phones, making use of the b and U functions in Dial() and some judicious use of GROUP channel variables and CHANNEL(hangup_handler_wipe). But, on my live system, sometimes the users receive calls from a queue, and I don’t see any way with the queue calls to emulate the b functionality in Dial() to be able to set the agent extension’s device state to RINGING when the queue call gets created. Obviously, I can use membergosub to set the agent to “INUSE” after they pick up the call (like Dial() U), but is there anything that I can use to manipulate the channel that is calling the agent while/before it is ringing? You could use local channels as queue members. Then you can use Dial(b) when the call goes out to the actual extension. Richard -- _ -- 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] How to test Websocket support in SIP in Asterisk trunk?
Put my sipml5 changes there. By the way, this is what happens when I try to call a X-Lite extension from a sipml5 extension: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 [Aug 20 17:24:02] ERROR[22737][C-0009]: chan_sip.c:32140 setup_srtp: No SRTP module loaded, can't setup SRTP session. [Aug 20 17:24:02] WARNING[22737][C-0009]: chan_sip.c:9974 process_sdp: Can't provide secure audio requested in SDP offer jcvmasterisk1*CLI Trying to do the reverse... X-Lite stays in Calling... - in sipml5, the right pane, with the local webcam thumbnailm, pops up, but no Answer button. Only Call and Hangup. Also, after a lng time, I get a ringing tone in X-Lite. And the webcam thing never goes away in sipml5. What I get in the log is just this: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 -- Executing [2010@demo:1] Dial(SIP/2012-0004, SIP/2010) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/2010 jcvmasterisk1*CLI sipml5 to sipml5: Not acceptable here. And the destination extension is totally inert. Log: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 [Aug 20 17:30:58] ERROR[22747][C-000c]: chan_sip.c:32140 setup_srtp: No SRTP module loaded, can't setup SRTP session. [Aug 20 17:30:58] WARNING[22747][C-000c]: chan_sip.c:9974 process_sdp: Can't provide secure audio requested in SDP offer jcvmasterisk1*CLI Meh, same thing as simpl5-to-plain-SIP. Juan On Mon, Aug 20, 2012 at 4:00 PM, Andrew Latham lath...@gmail.com wrote: On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro jcas...@instant.com.br wrote: Hoo-hah. It registers. Progress! Now... media. Or not. On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote: - Original Message - The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. Well, I did the following changes in sipml5 and now I get a Bad Request on REGISTER, instead of 404. Clearly, I'm still missing something. Here are the changes I made: You are probably getting hit by a bug in Asterisk 11 that has been fixed. It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support along with a work around via configuration. -- Joshua Colp Digium, Inc. | Software 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 -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- Juan Matt just opened https://issues.asterisk.org/jira/browse/ASTERISK-20267 to document some of this. Feel free to pipe in. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.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 -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- _ -- 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] Extensions mask as variable?
Hi, How to define a extension mask as global variable in Ast 1.8? For example: [globals] MYVARIABLE = _15[7-9]X I tried this way but it did not work. Thanks Att, Rafael Saraiva -- _ -- 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] Extensions mask as variable?
I don’t fool with 1.8 but I don’t think this is supposed to work. The extension mask and variable processing are two different routines. I assume you are trying to do “real-time” filtering where you would set a variable to control a group of extensions? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos Saraiva Sent: Monday, August 20, 2012 3:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Extensions mask as variable? Hi, How to define a extension mask as global variable in Ast 1.8? For example: [globals] MYVARIABLE = _15[7-9]X I tried this way but it did not work. Thanks Att, Rafael Saraiva -- _ -- 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 as TLS server as well as TLS client
On 20/08/12 21:11, Danny Nicholas wrote: This is all nice and good but the documentation all assumes that you are on a Debian box and use MYSQL. What about us SUSE/Postgresql folks? They are both good questions, and there are partial answers: SUSE: reSIProcate can be built from source on a large number of platforms. I recently converted the upstream project to autotools, this should make it straightforward to build (and even package it) for SUSE. There has been some mention of RPM packaging on the resiprocate dev email list. I'm even working on it for OpenCSW at the moment. Postgresql: This is a bigger challenge. - Scott recently added the MySQL support for the 1.8 release, before that there was no working DB support, just BDB files. - It should probably be generalised for UNIXODBC or something like that, I actually used that approach in dynalogin. However, it will probably need someone to volunteer or present a commercial opportunity to enhance it like that. As for the guides: to make it easy, they talk about what exists today. Once the RPM packages appear in Fedora or SUSE, I will definitely update the guides, there is no hidden agenda to force people onto Debian. -- _ -- 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 as TLS server as well as TLS client
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Pocock Sent: Monday, August 20, 2012 3:49 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk as TLS server as well as TLS client SUSE: reSIProcate can be built from source on a large number of platforms. I recently converted the upstream project to autotools, this should make it straightforward to build (and even package it) for SUSE. There has been some mention of RPM packaging on the resiprocate dev email list. I'm even working on it for OpenCSW at the moment. Postgresql: This is a bigger challenge. - Scott recently added the MySQL support for the 1.8 release, before that there was no working DB support, just BDB files. - It should probably be generalised for UNIXODBC or something like that, I actually used that approach in dynalogin. However, it will probably need someone to volunteer or present a commercial opportunity to enhance it like that. As for the guides: to make it easy, they talk about what exists today. Once the RPM packages appear in Fedora or SUSE, I will definitely update the guides, there is no hidden agenda to force people onto Debian. I'm fond of the tar-config-make method that Asterisk uses. Is this possible for reSIPprocate? If so can you provide a link? -- _ -- 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 as TLS server as well as TLS client
On 20/08/12 22:53, Danny Nicholas wrote: I'm fond of the tar-config-make method that Asterisk uses. Is this possible for reSIPprocate? If so can you provide a link? http://www.resiprocate.org/ReSIProcate_1.8_Release You can access the download directory (use the 1.8.5 tarball) or SVN from there Any feedback is welcome, there is a link there to the reSIProcate community email lists -- _ -- 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 11 - BLF on Custom devices
On 08/20/2012 03:20 PM, Noah Engelberth wrote:
And after 303 tries to call 302 while 301 302 are still on a call
(301 302 on a call, plus 303 calling 302):
-= Registered Asterisk Dial Plan Hints =-
_3XX@hints : Custom:${EXTEN} State:Idle
Watchers 0
302@hints : Custom:302
State:InUseRinging Watchers 2
303@hints : Custom:303 State:InUse Watchers 2
301@hints : Custom:301 State:InUse Watchers 2
But despite the above, the BLF fields on my phones (Cisco SPA 509G for
all 3 extensions) only update for Idle or InUse -- they do not show
the Ringing or InUseRinging statuses. I have verified the SPA phones
BLFs do still show the correct Ringing and InUseRinging statuses if
they subscribe directly to a SIP device's state with the hint -- the
issue only seems to be effecting Custom devices. Can anyone think of
anything else I should check?
I'd do a packet capture -- ideally from the phone, or using your switch
to mirror the phone's port -- and look for a SIP NOTIFY. Then we can
know if a NOTIFY is not being sent, or if it's just not being processed
as desired by your Cisco SPA 509G. If it's not there, do the same on
your asterisk server, and if we see it on the asterisk server but not at
the endpoint, we can suspect network configuration.
You can also get some more detail about what endpoints are subscribed
with sip show subscriptions in the asterisk console, but since it says
Watchers 2, that suggests the subscription has been made. I'd just
verify that specifically the device you are using for testing is subscribed.
Also, I know if you turn the verbosity up high enough (core set
verbosity 3) you will get messages in the console about notifications
sent. You can also set sip debug on or something along those lines and
have Asterisk print all the SIP traffic it's attempting to send. Should
help you narrow the possible causes.
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Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
Hi, you need to build Asterisk with SRTP support... *wget http://sourceforge.net/projects/srtp/files/latest/download -O srtp-latest.tgz tar -zxvf srtp-latest.tgz ./configure --prefix=/libsrtp make make install* *And for Asterisk...* *./configure --with-srtp=/libsrtp* * * *this should work...* * * *I did some changes to the sipml5 client and wanted to share this with you guys... Actually only 2 simple changes...* https://github.com/mailsvb/sipml5 *- The main config section has been splitted and made a little more flexible, see *http://i45.tinypic.com/10x59o7.png - Main call.html file has been renamed to .php and some code has been added that will replace the something.invalid with the actual IP of your client PC. Currently I am able to register and at least make my softphone ring ;-) As soon as I answer the outgoing call from sipml5 in the softclient, I get an error in sipml5... You can find my console output here http://pastebin.com/jdkXSMSD I will continue investigating tomorrow... best regards, Sven 2012/8/20 Juan Castro jcas...@instant.com.br Put my sipml5 changes there. By the way, this is what happens when I try to call a X-Lite extension from a sipml5 extension: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 [Aug 20 17:24:02] ERROR[22737][C-0009]: chan_sip.c:32140 setup_srtp: No SRTP module loaded, can't setup SRTP session. [Aug 20 17:24:02] WARNING[22737][C-0009]: chan_sip.c:9974 process_sdp: Can't provide secure audio requested in SDP offer jcvmasterisk1*CLI Trying to do the reverse... X-Lite stays in Calling... - in sipml5, the right pane, with the local webcam thumbnailm, pops up, but no Answer button. Only Call and Hangup. Also, after a lng time, I get a ringing tone in X-Lite. And the webcam thing never goes away in sipml5. What I get in the log is just this: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 -- Executing [2010@demo:1] Dial(SIP/2012-0004, SIP/2010) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/2010 jcvmasterisk1*CLI sipml5 to sipml5: Not acceptable here. And the destination extension is totally inert. Log: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 [Aug 20 17:30:58] ERROR[22747][C-000c]: chan_sip.c:32140 setup_srtp: No SRTP module loaded, can't setup SRTP session. [Aug 20 17:30:58] WARNING[22747][C-000c]: chan_sip.c:9974 process_sdp: Can't provide secure audio requested in SDP offer jcvmasterisk1*CLI Meh, same thing as simpl5-to-plain-SIP. Juan On Mon, Aug 20, 2012 at 4:00 PM, Andrew Latham lath...@gmail.com wrote: On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro jcas...@instant.com.br wrote: Hoo-hah. It registers. Progress! Now... media. Or not. On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote: - Original Message - The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. Well, I did the following changes in sipml5 and now I get a Bad Request on REGISTER, instead of 404. Clearly, I'm still missing something. Here are the changes I made: You are probably getting hit by a bug in Asterisk 11 that has been fixed. It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Supportalong with a work around via configuration. -- Joshua Colp Digium, Inc. | Software 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 -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- Juan Matt just opened https://issues.asterisk.org/jira/browse/ASTERISK-20267 to document some of this. Feel free to pipe in. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.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 -- Juan Carlos Castro y
Re: [asterisk-users] BLF and Call Queues
So I can just add these 4 lines to app_queue.c and this will give me the ability to use : exten = 566,hint,Queue:voipq1 ?? Yes, then I assume you know that you need to compile etc. ./configure make menuselect make make install Alec -- _ -- 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] BLF and Call Queues
2012/8/19 Alec Davis siva...@paradise.net.nz Do you also know why it hasn't been accepted ? Seems like this functionality is asked for on different forums. Wanting to watch a queue for calls is not that strange. Not sure why? Maybe I didn't promote it enough. Maybe my examples aren't simple enough. Sometimes things just slip off the radar and need to get pushed back to the top of the stack. Bumping the review or pinging people in #asterisk-dev usually works. In this particular case, the following comment might have made people think that you were going to put a new version up for review: This is slightly flawed, in that a queue with all agents on a call (that don't allow multiple calls), when another call comes in, the hint will flash, but cannot be picked up, as there isn't a ringing extension. The patch works perfectly to see if a person is in a queue. Taking that a little further, picking up the queue only works if an agent is ringing. However, when no agent is ringing, that's where the pickup will fail. Directed Pickup can only pickup devices that are ringing. Is this senario a queue issue, or a directed pickup issue? I was unable to decide. In our use case, it hardly ever happens. The main use for us is the night bell queue, which is a dahdi extension with just a ringer, no handset, so always needs to be picked up. Alec -- _ -- 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, What about Queue logs ? How is a picked-up call logged ? Giving agents the capability to easily pickup a call, without beeing logged-in, is a big change with both positive and negative side effects. I would be curious to read opinions about that. -- _ -- 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 hangs while starting in OpenSuse 12.2
Ok Thanks Bryant, let me try with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman brya...@zktech.comwrote: I have the current version of 8.x and 10.x on systems. I am using OpenSuse 12.1, We are working on getting a 12.2 boxs up just running out of time. Asterisk on all of our boxes are complied from source. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Gopalakrishnan N gopalakrishnan...@gmail.com *Sent*: Monday, August 20, 2012 10:11 AM *To*: Bryant Zimmerman brya...@zktech.com *Subject*: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 It's really glad that asterisk is installed at your machine in open suse. Can you let me know which version you are using and the architecture. Regards. On Aug 20, 2012 6:22 PM, Bryant Zimmerman brya...@zktech.com wrote: I compile from source.. Sent from my Verizon Wireless Phone - Reply message - From: Gopalakrishnan N gopalakrishnan...@gmail.com Date: Mon, Aug 20, 2012 8:15 am Subject: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com From the forum I understand OpenSuse 12.2 is pre-relase and better to use OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Its really weird working with OpenSuse. I am not sure how others are using with OpenSuse. Through Yast also I tried to install Asterisk package, it didn't find. Now I am clueless to work with OpenSuse. Regards. On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.comshould resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- _ -- 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
