Re: [asterisk-users] Paging systems?

[Bryant Zimmerman]

Michael

Based on how I read the manual if you connect the TIP and RING of the ATA to 
the right pairs you should be able to send a call to the paging box. It looks 
that when the page call is picked up by the paging system you would then press 
a zone 1-9 or 0 for all.  The page would then bridge to the desired zone. the 
page would complete when the call is hung up. You would likely need to make 
sure the ATA is using current loop disconnect or reverse to ensure hang-up.

I think it should be the PABX config using the Figure 3 configuration.

Best of luck

Bryant Zimmerman

Sr. Systems Architect
Grand Dial Communications, A ZK Tech Inc. Company
616-299-5607 (mobile)
616-855-1030 Ext. 2003 (office)


From: Michael Munger 
Sent: 3/21/19 7:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
, John Novack 
Subject: Re: [asterisk-users] Paging systems?
Excellent point.

This is it: https://www.valcom.com/pdf/v-1109rthf.pdf

Get Outlook for Android

On Thu, Mar 21, 2019 at 7:22 PM -0400, "John Novack"   
wrote:

Michael Munger wrote:

Does anyone have an (overhead) paging system that they like that works with SIP?



We’ve got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t 
auto-answer the call, so paging never happens.


Does it expect to see a POTS line with battery on it?
Then a Cisco or other ATA that would work to supply service to a POTS phone 
should work
OR:
Does it expect to see a POTS connection from a PBX trunk, and supply battery TO 
the trunk?
Then you would need a Cisco or other ATA with an FXO connection.

Both types of paging systems have been made and both styles of connections have 
existed through the last 30 + years, and since you haven't revealed the brand 
and model of paging system it makes troubleshooting difficult.
Using the existing system can be made to work
I use a very old Harris PagePak VS that was used with a Western Electric 
Horizon system back in the dark ages with Asterisk

John Novack

--  Dog is my Co-Pilot


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Re: [asterisk-users] Paging systems?

[Bryant Zimmerman]

Michael

We use Bogan UTI1 box in conjunction with an ATA to patch to any overhead 
paging system. You patch the box directly into the amp line in for the 
overhead. When you call the extension it answers and puts the audio on the line 
to the PA.

If you only have a few speakers the UTI1 can even handle being the amp for a 
few speakers.

Bryant Zimmerman

Sr. Systems Architect
Grand Dial Communications, A ZK Tech Inc. Company
616-299-5607 (mobile)
616-855-1030 Ext. 2003 (office)


From: Darryl Moore 
Sent: 3/21/19 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Paging systems?
For a paging system? No you don't. A number of SNOM PA1's and a few grandstream 
phones and you're golden. If you do need FXO or FXS, they are just as easy to 
setup as well, and there are lots to choose from.

On Thu, Mar 21, 2019, 4:45 PM Ryan, Travis,  wrote:

You need more than an ATA. You need something with an FSO and FXO. I’ve used 
Linksys/SPA3102-3.3.6 and been happy with it.







From: asterisk-users  On Behalf Of 
Sebastian Nielsen
Sent: Thursday, March 21, 2019 3:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 

Subject: Re: [asterisk-users] Paging systems?



How did the page system answer the call when it was used with the analog system?

You could propably ”fake” those signals from inside asterisk, and cause it to 
answer.



Från: asterisk-users  För Michael 
Munger
Skickat: den 21 mars 2019 20:00
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] Paging systems?



Does anyone have an (overhead) paging system that they like that works with SIP?



We’ve got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t 
auto-answer the call, so paging never happens.





Michael J. Munger, dCAP, MCPS, MCNPS, MBSS

Microsoft Certified Professional

Microsoft Certified Small Business Specialist

Digium Certified Asterisk Professional

High Powered Help, Inc.

p:

678-905-8569

w:

hph.io  e: m...@hph.io





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[asterisk-users] Asterisk 13.18.4 - New Error PJLIB_UTIL_EDNS_REFUSED

[Bryant Zimmerman]

We just updated from 13.17.1 to 13.18.4 and are noticing a new error
  
 [2017-12-21 10:12:48] ERROR[32343]: res_pjsip.c:3850 endpt_send_request: 
Error 320055 'DNS "Refused" (PJLIB_UTIL_EDNS_REFUSED)' sending OPTIONS 
request to endpoint 6162480909.8009
 
 The DNS on the system seems to be working find. Anyone have an idea what 
could be triggering this issue?

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003

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Re: [asterisk-users] user-agent access from pjsip

[Bryant Zimmerman]

I am trying to get the user-agent from extensions registered via pjsip. 
 With sip we could do a sip show peer peername and it would list the 
user-agent string. 
 In a pjsip deployment it looks like this info is likely in the contact. I 
know we can access it from the dialplan, but this is only works when a call 
occurs. How can we get the user-agent for extensions from the console. We 
need this for firmware version checking of extensions as many providers 
include that in the user-agent.  Any ideas as the pjsip show contact 
contactname does not return any real helpful info to the command line. 
  
 Please advise if you are able. 
  
 Thanks
 Bryant

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[asterisk-users] PJSIP Asteirks 13 - Audio Jitter in one direction only

[Bryant Zimmerman]

?We have upgraded a system from Asterisk 11 to Asterisk 13 with pjsip.
 We are experiencing random Jitter on outbound calls. This was not occurring 
when running asterisk 11.

 We have two IP's bound to pjsip one on the private vlan network the phones are 
on and the asterisk one on the asterisk wan vlan. We record the calls on the 
asterisk switch so we have the call legs. It appears that the audio is making 
it to the switch fine, but is being garbled before it leaves asterisk to the 
destination carrier. We have all media running through the server and this is 
happening when there is only 1 to 2 calls on the line. The cpu, and memory are 
not even being pushed. We are running G711 as the codec so there should be no 
real transcoding occurring..

 What could be causing this. The users are very upset. This is a very transient 
issue so the breakup is can occur for two to four seconds and then goes away. 
It is like asterisk and pjsip are screwing with the audio. Please advise.

 zktech

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Re: [asterisk-users] PJSIP add header not working

[Bryant Zimmerman]

Andre
  
 For this to work we have had to go to using the b() option in the dial 
legs for the calls that are pasting up.
 You call a context that gets run before the calls are made on each 
channel. This allows you to add headers to the new pjsip channels. 
 It works well. You can also set variables with the _ option to trigger 
which headers you want to add..
  
 The example below would add "ThisHeader", "ThatHeader" and "Call-Info" to 
the new channel created in the dial. You could use combinations of other 
variables and augment these methods to meet almost any need. 
  
 Exp
  
 [OutboundDial]
 exten => _XX,1,NoOp(Dial Exp)
 exten => _XX,n,Set(_var1setinparrent=1) ;;Set Variable so that 
when you call the b() option context in your dial the first header is 
added
 exten => _XX,n,Set(_var2setinparrent=1) ;;Set Variable so that 
when you call the b() option context in your dial the second header is 
added
   exten => _XX,n,Set(_varAddSessionInparrent=1) ;;Set Variable so 
that when you call the b() option context in your dial the second header is 
added

 exten => 
_XX,n,Dial(pjsip/333222@vendortrunk,b(AddpjsipHeaders^s^1))
  
  
 [AddpjsipHeaders]
  exten =>s,1,Gosubif({"$[var1setinparrent}}"="1"]?ThisHeader,1)

 exten =>s,n,Gosubif({"$[var2setinparrent}}"="1"]?ThatHeader,1)
 exten 
=>s,n,Gosubif({"$[varAddSessionInparrent}}"="1"]?addSessionCallInfo,1)
  
 exten => ThisHeader,1,Set(PJSIP_HEADER(add,ThisHeader)=ValueToSet)
 exten => ThisHeader,n,Return()
  
 exten =>  ThatHeader,1,Set(PJSIP_HEADER(add,ThatHeader)=ValuetoSet)
 exten =>  ThatHeader,n,Return()
  
 exten => 
addSessionCallInfo,1,Set(PJSIP_HEADER(add,Call-Info)=\;answ
er-after=0)
exten => addSessionCallInfo,n,Return()
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: "Andre Gronwald" <andregronwal...@gmail.com>
Sent: Monday, October 2, 2017 11:07 AM
To: "asterisk-users" <asterisk-users@lists.digium.com>
Subject: [asterisk-users] PJSIP add header not working   

Hi,
I am trying to add a custom header to my calls to map several call-legs 
into a global call for viewing.

For this to work I read the call-id from pjsip-channel and write it into 
X-CID:

##
-- Executing [s@macro-dialout-trunk-predial-hook:4] 
Set("PJSIP/10-0006", 
"pjsipCallId=313530363933383438363436353930-1gh0bjceo933") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:5] 
Set("PJSIP/10-0006", 
"PJSIP_HEADER(add,X-CID)=313530363933383438363436353930-1gh0bjceo933") in 
new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/10-0006", 
"0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/10-0006", 
"1?Set(CONNECTEDLINE(num,i)=0xx)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/10-0006", 
"1?Set(CONNECTEDLINE(name,i)=CID:3x)") in new stack
-- Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/10-0006", 
"0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)3x)") in new stack
-- Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/10-0006", 
"0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:23] Dial("PJSIP/10-0006", 
"PJSIP/0xx@3x,300,T") in new stack
-- Called PJSIP/0xx@3x
<--- Transmitting SIP request (991 bytes) to UDP:217.23.24.100:5060 --->
INVITE sip:0xxx...@sip.provid.er:5060 SIP/2.0
Via: SIP/2.0/UDP 
192.168.253.185:15070;rport;branch=z9hG4bKPj453d15e0-de58-4945-8b95-d05b16b9
e4c3
From: 
<sip:+49...@sip.provid.er>;tag=080788ac-7c10-4cf3-86b3-359764ffb5a2


To: <sip:0xxx...@sip.provid.er>
Contact: <sip:+49x@192.168.253.185:15070>
Call-ID: de41b93b-51d8-44b5-9c34-f2c0928192b0
CSeq: 1519 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-14.0.1.10(14.6.2)
Content-Type: application/sdp
Content-Length:   308

v=0
o=- 1719768133 1719768133 IN IP4 192.168.253.185
s=Asterisk
c=IN IP4 192.168.253.185
t=0 0
m=audio 55112 RTP/AVP 107 9 8 3 101
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Received SIP response (559 bytes) from UDP:217.23.24.100:5060 --->
[...]

##

But I can't see that header anywhere in my call-legs. What am I missing?

kind regards,
andre 

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[asterisk-users] Bug in func_odbc module

[Bryant Zimmerman]

Hey all
  
 I have code we are moving from an early asterisk 13 system to the latest 
build. 
  
 The issue we are having is func_odbc calls are acting incorrectly. 
 We have tables that have fields with null values in them. 
  
 On the new system when we read a field with a null value it is copying the 
value from the previous filed into the value and not leaving the filed as 
null or blank string. So what is happening is we get variables inside of 
our dialplan that have values from other variables fields. 
 As soon as the system hits a value with a non null field, even a filed 
with an empty string it self corrects until it hits another field value 
with a null string in it. 
  
 Any thoughts on how and when this could get fixed?
  
 Thanks

Bryant 

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Re: [asterisk-users] Asterisk pjsip registration issues - Solved

[Bryant Zimmerman]

Dave
  
 from_user  fixed the issue. 
  
 Thank You Thank You Thank You 
  
 I was about ready to chuck pjsip. The lack of good / complete 
documentation is a real problem. 
 Man you saved me another late night. 

Thanks

Bryant
  


 From: "Dave Platt" 
Sent: Tuesday, September 26, 2017 3:28 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk pjsip registration issues   

> Hey all
>
> I am trying to register a PJSIP server on our office to an Asterisk 11
> chan_sip server in a datacenter.
>
> I keep getting
> WARNING[18084]: res_pjsip_outbound_authenticator_digest.c:178
> digest_create_request_with_auth_from_old: Host: 'XXX.XXX.XXX.XXX:5060':
> Unable to create request with auth. No auth credentials for realm(s)
> 'asterisk' in challenge.
>
> Any insights would be appreciated I have been banging my head for
several
> days now.

I ran into a very similar problem when I tried to switch my PJSIP
service with Vitelity from "fixed IP address" to "registration-based".
I would try to place a call, and it would simply time out and then get
a "busy here" error from Vitelity.

Calls to a similar Vitelity sub-account from a Zoiper soft-phone worked
just fine.

I wiresharked the sessions and found that the critical difference seemed
to be in the From: and Contact: headers. Zoiper set these to the
Vitelity sub-account name (the registration name) while PJSIP just set
them to "asterisk".

I checked the PJSIP wizard file, and found that the outbound
authentication object had the right username information in it,
so that wasn't the problem.

After stumbling around for hours, I found that it's necessary to
set the "from_user" parameter in the endpoint object to match the
username in the outbound authentication object. This causes PJSIP
to send this value (rather than "asterisk") in the From and Contact
fields of the INVITE, and this apparently gives the far end the
information it needs to issue a proper credentials challenge.

Once I added this one line to my definition and restarted, outbound
calls worked like a charm.

So, in pjsip_wizard, one would write something like

[peername]
type = wizard
transport = transport-udp
remote_hosts = outbound.peer.com
sends_auth = yes
endpoint/context = outbound
endpoint/from_user = MYNAME
outbound_auth/username = MYNAME
outbound_auth/password = MYPASSWORD

Modify and embellish as required. If you're writing your PJSIP
objects individually rather than via the wizard, just set the
fields in those objects appropriately.

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[asterisk-users] Asterisk pjsip registration issues

[Bryant Zimmerman]

Hey all
  
 I am hoping someone can assist I have now spent over a week trying to 
figure out what is going on with PJSIP registrations. 
  
 I am able to register handsets against an asterisk 13 server running 
pjsip, but I am not able to get pjsip to register out to an older chan_sip 
asterisk server. 
 If I drop the registration I can make things work, but when I have to 
register the asterisk - pjsip server against another server the 
registration completes, but I can not send any calls across the 
registration, nor will it handle options correctly as well. 
  
 We keep getting ... No auth credentials for realm(s) 
'aster...@xxx.xxx.xxx.xxx' in challenge.
in one form or another, and I have been unable to find any definitive 
documentation on what is at cause for this. In some areas I have seen 
responses saying it is an issue with realms so I have tried with and 
without but no success. 
  
 I really need some direction on this. This is the last issue I know of 
that is holding up us from moving to pjsip. If I can't get asterisk / pjsip 
to register and send authenticated  messages than it can't work for 
replacing chan_sip in all situations.   
  
 What am I doing wrong. 
  
  
  [zktech_trunk]
type=registration
 endpoint=zktech_trunk
transport=udp-nat
outbound_auth=zktech_trunk
server_uri=sip:acct.8...@xxx.xxx.xxx.xxx
client_uri=sip:acct.8...@xxx.xxx.xxx.xxx
contact_user=zktech_trunk
retry_interval=60
forbidden_retry_interval=600
expiration=3600
line=yes
  
 [zktech_trunk]
type=auth
auth_type=userpass
password=rossi72v8qr
username=ACCT.8009
realm=aster...@xxx.xxx.xxx.xxx
  
 [zktech_trunk]
 type=aor
max_contacts=1
contact=sip:acct.8...@privxxx.xxx.xxx.xxx:5060
qualify_frequency=60

  
 Thanks

Bryant 

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Re: [asterisk-users] Registering Asterisk 13 server PJSIP to Asterisk 11 SIP

[Bryant Zimmerman]

Hey all
  
 I am trying to register a PJSIP server on our office to an Asterisk 11 
chan_sip server in a datacenter. 
  
 I keep getting 
  WARNING[18084]: res_pjsip_outbound_authenticator_digest.c:178 
digest_create_request_with_auth_from_old: Host: 'XXX.XXX.XXX.XXX:5060': 
Unable to create request with auth. No auth credentials for realm(s) 
'asterisk' in challenge.
  
 Any insights would be appreciated I have been banging my head for several 
days now. 

Thanks

bryant

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Re: [asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

 
  Original Message 
> From: "Joshua Colp" <jc...@digium.com>
> Sent: Friday, September 15, 2017 11:31 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Realtime pjsip issues
>
> On Fri, Sep 15, 2017, at 12:18 PM, Bryant Zimmerman wrote:
> > Joshua
> >
> > We are using MariaDB as the database storage.
> > We have recreated the database tables with alembic.
> >
> > Test 1:
> > We enable tables for aors, auths and endpoints only. With cache turned
> > off the end point registers successfully We have no way to get any
> > feed
> > back as pjsip show/list returns no objects found. pjsip send notify
> > cmd
> > endpoint -- does not work as it says there is no endpoint. endpoint
> > can
> > send a call as it appears to be registered, we have no way to confirm
> > this
> > form the console but calls come in.
>
> 
>
> The show and list commands are supposed to work, even without caching
> being enabled. Your problem is therefore at the realtime level. Calls
> coming in should appear on the console, and the endpoint name will be in
> the channel name. Enabling caching just masks it some because things
> exist in the cache for a bit.
>
> >
> > I can offer the following:
> > A dump of the database schema that alembic is creating.
> > extconfig.config
> > sorcery.conf
>
> Feel free to provide these and me (or another individual) may pick out
> what is wrong.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
  
 I have linked to a zip file containing a dump of my sql schema (MySQL), 
extconfig.conf, sorcery.conf
  
 dumps.zip
  
 Hopefully someone can see what might be causing our issues with the pjsip 
realtime system. 
  
 Thanks 
 Bryant
 



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Re: [asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

Joshua
  
 We are using MariaDB as the database storage. 
 We have recreated the database tables with alembic. 
  
 Test 1:
We enable tables for aors, auths and endpoints only.With cache 
turned 
off the end point registers successfullyWe have no way to get any feed 
back as pjsip show/list returns no objects found.   pjsip send notify cmd 
endpoint -- does not work as it says there is no endpoint.  endpoint can 
send a call as it appears to be registered, we have no way to confirm this 
form the console but calls come in.  
  
 Test 2: 
We enable cache on the endpoints, auth and aors in the sorcery.conf 
 
endpoint/cache = 
memory_cache,object_lifetime_stale=600,object_lifetime_maximum=1800,expire_o
n_reload=yes,full_backend_cache=yes 
auth/cache=memory_cache,expire_on_reload=yes
aor/cache = 
memory_cache,object_lifetime_stale=1500,object_lifetime_maximum=1800,expire_
on_reload=yes,full_backend_cache=yes
We now get an error:[2017-09-15 
11:02:04] WARNING[3375]: 
res_pjsip_registrar.c:744 registrar_on_rx_request: AOR '6162480909-300' has 
no configured max_contacts. Endpoint '6162480909-300' unable to register 
The aors entry has the max_contacts set to 1 but the error 
still occurs.  

pjsip show/list shows the endpoint shows endpoints, aors, 
auths  but 
registration fails 
  
  Test 3: 
We enable cache on the endpoints, auth and aors in the sorcery.conf 
 
endpoint/cache = 
memory_cache,object_lifetime_stale=600,object_lifetime_maximum=1800,expire_o
n_reload=yesauth/cache=memory_cache,expire_on_reload=yes
aor/cache = 
memory_cache,object_lifetime_stale=1500,object_lifetime_maximum=1800,expire_
on_reload=yes   
Endpoint registers  pjsip show/list endpoints works the 
first time and 
fails there after.  UBNTU-ROSSI-GUEST*CLI> pjsip 
show endpoints
 Endpoint:
  
I/OAuth:  


Aor:

  Contact:
 <RTT(ms)..>
  Transport:  

   Identify:  


Match:  
Channel:
  
Exten:   CLCID: 


===
===
 Endpoint:  6162480909-300   Not in 
use0 of inf
 InAuth:  6162480909-300/6162480909-300
Aor:  6162480909-300 1
  Contact:  6162480909-300/sip:6162480909-300@192.168. 0475d46ff2 
Unknown nan
  Transport:  udp-nat   udp  0  0  0.0.0.0:5060

Objects found: 1
UBNTU-ROSSI-GUEST*CLI> pjsip show endpoints
No objects found.

pjsip show/list shows the endpoint fails ever time after the 
first. 

 Test 4: 
Test 1: with the addition of the contacts entry as realtime in 
sorcery.confWe get error on registration attempt:   
[2017-09-15 
11:16:07] WARNING[3591]: res_config_odbc.c:120 custom_prepare: SQL Prepare 
failed! [INSERT INTO ps_contacts (id, via_addr, qualify_timeout, call_id, 
reg_server, path, endpoint, via_port, authenticate_qualify, uri, 
qualify_frequency, user_agent, expiration_time, outbound_proxy) VALUES (?, 
?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?)] [2017-09-15 11:16:07] 
ERROR[3591]: res_pjsip_registrar.c:432 register_aor_core: Unable to bind 
contact 'sip:6162480909-300@192.168.201.105:59758' to AOR '6162480909-300' 

Registration has failed at this point.  
  
 I can offer the following:
 A dump of the database schema that alembic is creating.
 extconfig.config
 sorcery.conf
  
 Thanks
 Bryant
  


 From: "Joshua Colp" <jc...@digium.com>
Sent: Friday, September 15, 2017 9:56 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime pjsip issues   
On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote:
> Joshua
>
> That is the interesting part of it. We took our configs and database
> tables from our working 13.12.2 deployments and tried to use them with
> our
> new 13.17.1 deployments and we are having issues where the tables are 
not
> working. On the new server asterisk keeps saying it can't find the AORS
> entries when we purge the sorcery memory cache it starts finding the 
aors
> but then it says it cant find the auths.
>
> The wired thing is when it says it can't find the aors and auths entries
> it does not show it is looking for the values in the aors and auth 
fields
> from the endpoints tables. It keeps putting the value from the endpoints
> id
> field as the entries it can't find.
>
> One point of

Re: [asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

Joshua
  
 We have completed more testing this morning and when we remove the 
realtime cache options from the sorcery file the endpoints complete 
registration, but we pjsip show/list does not offer any feed back at all, 
We also can't send any pjsip send notify commands as they say they don't 
have an endpoint there. Something has changed in the cache part of the 
system that is breaking the system in some manner for us with the current 
version and we are out of ideas. 
  
 Thanks
 Bryant
  
  
  
  
  Joshua
  
 That is the interesting part of it. We took our configs and database 
tables from our working 13.12.2 deployments and tried to use them with our 
new 13.17.1 deployments and we are having issues where the tables are not 
working. On the new server asterisk keeps saying it can't find the AORS 
entries when we purge the sorcery memory cache it starts finding the aors 
but then it says it cant find the auths. 
  
 The wired thing is when it says it can't find the aors and auths entries 
it does not show it is looking for the values in the aors and auth fields 
from the endpoints tables. It keeps putting the value from the endpoints id 
field as the entries it can't find. 
  
 One point of note the tables we used and created for pjsip back when we 
setup the 13.12.2 version are not what is currently being created when we 
run alembic now.. Also the contact table from alembic creation process does 
not work we get insert errors inside of asterisk when contact entry 
attempts are being crated. It shows a number of fields that are not there 
in the created tables. 
  
 This is the foundation of my issues. I really have to resolve them in some 
manner so I can mover forward with getting these new systems into 
production. 
 Any assistance is appreciated. 
  
 Thanks
 Bryant
  


 From: "Joshua Colp" <jc...@digium.com>
Sent: Thursday, September 14, 2017 4:34 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime pjsip issues
 On Thu, Sep 14, 2017, at 05:27 PM, Bryant Zimmerman wrote:
> This appears to be some kind of cache issue.
> We have been doing caching with earlier versions of asterisk 13 on the
> pjsip realtime, but now for some reason
> The items only show up the first time we use pjsip list/show and then
> they
> are wiped. I see a new full cache option and that appears to make a
> difference, but it is unclear what is going on. In effect it appears 
that
> items loaded from a database for pjsip must be fully cached or you can't
> look up any data.
>
> Why has a change of this magnitude been put into an LTS?
> What is the best practices. I see in some of the wikis cache
> suggestions.
> What are others really seeing?

There haven't been any changes made except for bug fixes to the sorcery
memory cache, certainly no behavior changes. In fact the implementation
is the same between 13 and 14 except for a single line addition. What is
your sorcery.conf for both?

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

Joshua
  
 That is the interesting part of it. We took our configs and database 
tables from our working 13.12.2 deployments and tried to use them with our 
new 13.17.1 deployments and we are having issues where the tables are not 
working. On the new server asterisk keeps saying it can't find the AORS 
entries when we purge the sorcery memory cache it starts finding the aors 
but then it says it cant find the auths. 
  
 The wired thing is when it says it can't find the aors and auths entries 
it does not show it is looking for the values in the aors and auth fields 
from the endpoints tables. It keeps putting the value from the endpoints id 
field as the entries it can't find. 
  
 One point of note the tables we used and created for pjsip back when we 
setup the 13.12.2 version are not what is currently being created when we 
run alembic now.. Also the contact table from alembic creation process does 
not work we get insert errors inside of asterisk when contact entry 
attempts are being crated. It shows a number of fields that are not there 
in the created tables. 
  
 This is the foundation of my issues. I really have to resolve them in some 
manner so I can mover forward with getting these new systems into 
production. 
 Any assistance is appreciated. 
  
 Thanks
 Bryant
  


 From: "Joshua Colp" <jc...@digium.com>
Sent: Thursday, September 14, 2017 4:34 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime pjsip issues   
On Thu, Sep 14, 2017, at 05:27 PM, Bryant Zimmerman wrote:
> This appears to be some kind of cache issue.
> We have been doing caching with earlier versions of asterisk 13 on the
> pjsip realtime, but now for some reason
> The items only show up the first time we use pjsip list/show and then
> they
> are wiped. I see a new full cache option and that appears to make a
> difference, but it is unclear what is going on. In effect it appears 
that
> items loaded from a database for pjsip must be fully cached or you can't
> look up any data.
>
> Why has a change of this magnitude been put into an LTS?
> What is the best practices. I see in some of the wikis cache
> suggestions.
> What are others really seeing?

There haven't been any changes made except for bug fixes to the sorcery
memory cache, certainly no behavior changes. In fact the implementation
is the same between 13 and 14 except for a single line addition. What is
your sorcery.conf for both?

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

This appears to be some kind of cache issue. 
 We have been doing caching with earlier versions of asterisk 13 on the 
pjsip realtime, but now for some reason
 The items only show up the first time we use pjsip list/show and then they 
are wiped. I see a new full cache option and that appears to make a 
difference, but it is unclear what is going on. In effect it appears that 
items loaded from a database for pjsip must be fully cached or you can't 
look up any data. 
  
 Why has a change of this magnitude been put into an LTS?
 What is the best practices. I see in some of the wikis cache suggestions. 
What are others really seeing?

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: "Bryant Zimmerman" <brya...@zktech.com>
Sent: Thursday, September 14, 2017 2:43 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Realtime pjsip issues   
 We are having an issue where on the latest version of asterisk when 
configuration pjsip via realtime. 

   we do a pjsip list endpoints  it shows our endpoints but lists them as 
invalid. 
   When we do the pjsip list endpoints again it shows no objects. 

   This applies to pjsip list aors as well.  We did not have this issue on 
our older asterisk 13 installs. My guess is something has changed with 
pjsip and realtime. Anyone have any ideas where I can start. We have tried 
a number of things already and would love some suggestions. 


Thanks
zktech


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[asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

We are having an issue where on the latest version of asterisk when 
configuration pjsip via realtime. 
  
 we do a pjsip list endpoints  it shows our endpoints but lists them as 
invalid. 
 When we do the pjsip list endpoints again it shows no objects. 
  
 This applies to pjsip list aors as well.  We did not have this issue on 
our older asterisk 13 installs. My guess is something has changed with 
pjsip and realtime. Anyone have any ideas where I can start. We have tried 
a number of things already and would love some suggestions. 
  

Thanks
zktech

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Re: [asterisk-users] softphone instead of desktop phones

[Bryant Zimmerman]

Thomas
  
 Bria is by counterpath

Bryant


 From: "Matt Riddell (lists)" 
Sent: Saturday, April 29, 2017 11:50 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] softphone instead of desktop phones   
 I use Bria on all of the above. 

Kind regards,   
 Matt

On Apr 29, 2017, at 10:35 AM, Thomas  wrote:
 
  Hello,
Iam lookong for an Softphone for iPhor oder Android smartphone using 
togehter 
with an headset.
I tried Zoiper and CSipSimple but quality was bad compared to an desktop 
SIP 
phone.

Is there an better softphone?

Or are there softphone solutions for PC desktop MAC or Android with an 
headset?
I want to save cost for desktop phones.

thanks Thomas

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Re: [asterisk-users] softphone instead of desktop phones

[Bryant Zimmerman]

Thomas
  
 I have found you will likely end up spending more with softphones. You 
have to purchase a good headset and quality does not come inexpensive with 
headsets. We find we will pay $80 to $100 dollars plus for a fair quality 
headset. You also have to support deployment to devices. Which means in 
many cases you must touch the device for maintenance.
  
 The best softphone I have found is by counter path. They have versions for 
PC, MAC, Android and IOS. You will spend about $60.00 / device for the 
software. They have a fee edition for some devices, but not as feature 
rich.
  
 I have found it much more successful to deploy something like a Grand 
Stream Phone $100.00 for a nice feature phone, and I get convenient remote 
provisioning, Users never have username and passwords so security is 
ensured, Quality is good, I don't have to worry about phone issues if their 
pc or smartphone is acting up for the most part they just work. There are 
phones for as low as $50.00 from grand stream with high quality audio just 
fewer features.
  
 Softphones can work well, but when looking at trouble tickets over the 
last 13 years. I get more from my softphone users then from my desk phone 
users. And I can hear the quality difference when I talk to the softphone 
users it is such a wild card. How important is consistent quality to you? 
Users change headsets, move mic positions and the caller on the other end 
gets the short end of it. Also remember virus scans, patches, updates, 
system reboots, wifi signal strength, cell signal (cell ip calls) cheep 
Bluetooth headsets and the list goes on and on.   
  
 The counterpath software is as good as I have found, but there are 
additional variables you have to look at. Which is the best for your use 
case and the cheapest? You have to figure out your formula make sure to 
look at all the costs and factors how much is support time worth?
  
 In very controlled environments where you can have consistent control and 
good quality headsets such as call centers soft clients can work well, but 
that is not how you started out you are looking for quality on the cheep.
  
 Desk phones are cheep and in most cases just work and offer consistent 
quality.
  
 If others have found different I look forward to seeing their responses. 
This is a great question thanks for asking it Thomas.

Best of luck

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: "Thomas" <thomasit...@gmail.com>
Sent: Saturday, April 29, 2017 11:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] softphone instead of desktop phones   
Hello,
Iam lookong for an Softphone for iPhor oder Android smartphone using 
togehter
with an headset.
I tried Zoiper and CSipSimple but quality was bad compared to an desktop 
SIP
phone.

Is there an better softphone?

Or are there softphone solutions for PC desktop MAC or Android with an
headset?
I want to save cost for desktop phones.

thanks Thomas

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Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED]

[Bryant Zimmerman]

In most instances the company being called is not charging the caller for 
their phone serves. That is the callers service provider, and once the 
answer is issued the call is up.  
  
 This only makes senses if the company being called is providing services 
and charging a per min rate for that service. They would not charge the 
customer for the hold time waiting for a rep to come on the line. 
  
 This could all be done by creating billing records from cel logs. These 
can log events such as channel start and answer by an extension, transfers 
and hangups.  
  
 As Samy Go stated a good way to reduce charges to the caller would be to 
offer call back options. So when a rep is available the system would call 
the original caller back. 
  
 Telecom networks around the world are just not designed to offer delayed 
billing. Legislating that requirement would require world wide overhauls of 
the networks as well as treaties. 
  
 In some areas you also have to pay ring time. That is a novel idea to 
actually pay for a resource you are using when you use it. That is a little 
too capitalistic for some. 

Bryant


 From: "SamyGo" 
Sent: Wednesday, March 29, 2017 9:52 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] How to have callers not being billed when in 
waiting queue ? [SOLVED]   
 Hi,   Just trying to figure out how is this solved ? by involving multiple 
telcos in the loop and asking them to not charge based on 200 OK/Answer!?
 As far as I know people have designed Queue/CallCenter platforms who upon 
entering a number in queue just state them their number in queue and approx 
time before they'll be contacted and drop the call. This all can be done 
within Progress.
  
 As soon as their turn comes the CallCenter platform automatically triggers 
the call to them and get them connected with an agent. This is the way I 
can understand as nobody waiting in the queue but people in the 
waiting-list. Since Queue has to "answer" the call first before doing 
anything once the signal to Answer is triggered technically that marks the 
start of billing for everyone.
  
 Regards,
 Sammy

   On Wed, Mar 29, 2017 at 7:25 AM, Olivier  wrote: 
 Thank you very much, Max, for this valuable and informative answer.
 
Offline billing must be quite complex to set up as several telco may be 
involved (or origination,transit or termination).
Moving to normal landline fare seems much simpler !
 
Thanks again2017-03-28 21:41 GMT+02:00 Max Grobecker 
:  Hi,

in Germany, this kind of regulation is in effect for phone numbers which 
cost more than a normal landline call.
The regulation states, that the waiting time must not be charged to the 
customer.

Most companies implemented this by simply switching their telephone numbers 
to those, which are charged per call
(so there's no difference in price between waiting for someone to pick up 
or being connected to someone) ;-)
Or they decided to use a normal landline phone number for which this 
regulation does not apply.

The second method was to not answer the call before really connected to a 
person on the queue and using Early Media as you mentioned.
But: The maximum length of this Early Media stream is in most telephone 
networks limited to somewhat around 90 to 180 seconds,
then the call gets disconnected by the network.

I'm not very familiar with regulations and numbering plans in France, but 
maybe there's also something called "offline billing".
Using this, your call is not billed by the caller's telephone company until 
you send them the amount of time that should be billed for a specific 
call.

Your best choice will be, that - if you ever get those regulations - you 
should rely on what your telephone number provider tells you to do ;-)

Greetings
 Max

Am 28.03.2017 um 15:24 schrieb Olivier:
> Hello,
>
> In France, years ago, there was some discussions about a new regulation 
forcing some providers to not charge anything to callers while those are 
waiting for a call center agent to become available.
> Once caller and agent are on call with each other, nominal charging 
applies.
>
> No matter if those discussions ever did or didn't change current 
regulation, I wonder which dialplan statements could technically comply 
this dual billing requirement ?
>
>
> same = n,Progress()
> same = n,Queue(whatever,...,macro-option, ...)
>
> To me, coupling Progress app with Queue's  macro or gosub option like 
above, would let a sysadmin answer a queued call.
> Doing so, time spent before connection with queue agent should not be 
billed to anyone (caller nor callee), while time spent after connection is 
billed normaly.
>
> 1. Should this work ? Am I missing something ?
>
> 2. Is there an alternative way to implement this ?
>
> 3. Comments ? Suggestions ?
>
> Regards
>
>
 

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Re: [asterisk-users] WebRTC - Transport Issues. - Solved

[Bryant Zimmerman]

Josh
  
 Thank you for the confirmation on this. The captures do confirm that I am 
using the wss. 
 What was throwing me was I have only udp and wss in the transports and 
then the Primary once connected was showing the ws. 
 At first I thought I was doing something wrong and the traffic was flowing 
unencrypted.  You confirmed what I had hoped that the wss was just showing 
the underlying ws transport.
  
 A big thanks. We are excited to finally getting our webrtc test 
application out to some customers. 
  
 Have a great week. 
 Bryant

From: "Joshua Colp" <jc...@digium.com>
Sent: Sunday, March 12, 2017 7:35 PM   

On Sat, Mar 11, 2017, at 09:52 PM, Bryant Zimmerman wrote:
> Hey all. I have webrtc up and running with asterisk 11. All is going 
well
> with TLS now working.
> At least I hope it is using TLS and wss. Based on what I am seeing I
> have
> UDP, WSS listed in the Allowed transports, but every time I connect the
> Primary transport shows WS.. Why is this? Am I actually running ws in
> wss
> mode?

You are using WSS (the Contact line has transport=wss which indicates
it). Both WS and WSS will show "WS" for the Primary Transport. Another
way to tell is to look at the SIP traffic and check the Via header for
WSS. You can also check a packet capture.
   

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[asterisk-users] WebRTC - Transport Issues.

[Bryant Zimmerman]

Hey all. I have webrtc up and running with asterisk 11. All is going well 
with TLS now working.
 At least I hope it is using TLS and wss. Based on what I am seeing I have 
UDP, WSS listed in the Allowed transports, but every time I connect the 
Primary transport shows WS..  Why is this?  Am I actually running ws in wss 
mode?
   
   Prim.Transp. : WS
  Allowed.Trsp : UDP,WSS
  Def. Username: 6167761066.2011
  SIP Options  : (none)
  Codecs   : (ulaw)
  Codec Order  : (ulaw:20)
  Auto-Framing : No
  Status   : OK (71 ms)
  Useragent: SIP.js/0.7.7
  Reg. Contact : sip:fed97qgu@192.0.2.35;transport=wss
  Any Insights would be appreciated.
  
 Thanks
 Bryant

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Re: [asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code. - Solved

[Bryant Zimmerman]

I figured this out.

 I had to set the outofcall_message_context = messages on the actual peer.
 It was not good enough to set in the sip.conf

 Thanks
Bryant


 From: "Bryant Zimmerman" <brya...@zktech.com>
Sent: Friday, March 10, 2017 11:39 AM
  Jean

   Thank you for your response. I have the options you suggested already set, 
and I am still not getting the dialplan to trigger. The message is being sent 
but nothing. I have tried with the auth both set to no and yes as well.

   accept_outofcall_message = yes
outofcall_message_context = messages
auth_message_requests = yes

Thanks
Bryant




   From: "Jean Aunis" <jean.au...@prescom.fr>
Sent: Friday, March 10, 2017 2:24 AM


This is not a SIP NOTIFY but a SIP MESSAGE (the first line in the logs is not 
related to the few next ones).

If you are using chan_sip, you have to activate out of call messages in 
sip.conf :

accept_outofcall_message=yes
outofcall_message_context=messages

Then in extensions.conf, define a context "messages" with the appropriate 
extensions (to stick to your example, it will be 16162995607) and use the 
function MESSAGE to retrieve the SMS content.

Best regards

Jean Aunis      Le 10/03/2017 à 00:21, Bryant Zimmerman a écrit :
I am trying to send SMS from my grandstream GXV3240
 Asterisk receives the message in a NOTIFY block.

 How can I get asterisk to run dialplan code when receiving these Notify SMS 
Message Blocks.
 I can then route them to my SMS provider.

 Any ideas are appreciated. Below is debug of a message sent from the phone 
when received no dialplan code is triggered.
 I am wounding if I need to modify some setting in sip.conf or the peer config. 
 Incomming SMS from my vendor works without issue and is transmitted to the 
phone.


<->
--- (10 headers 0 lines) ---
Really destroying SIP dialog 
'3927411c7fe967886df6c8d0410d4...@xxx.xxx.xxx.xxx:5060' Method: NOTIFY

<--- SIP read from UDP:XXX.XXX.XXX.XXX:57568 --->
MESSAGE sip:16162995...@vgw0005.granddial.net SIP/2.0
Via: SIP/2.0/UDP 192.168.201.104:20093;branch=z9hG4bK1738682353;rport
From: <sip:6167761066.2...@vgw0005.granddial.net>;tag=1683585926
To: <sip:16162995...@vgw0005.granddial.net>
Call-ID: 1662412698-20093-...@bjc.bgi.cab.bae
CSeq: 9430 MESSAGE
Contact: <sip:6167761066.2003@192.168.201.104:20093>
Max-Forwards: 70
User-Agent: Grandstream GXV3240 1.0.3.158
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, 
UPDATE, MESSAGE
Content-Type: text/plain; charset=UTF-8
Content-Length: 5

Test Message SMS
<->

Thanks

Bryant




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Re: [asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code.

[Bryant Zimmerman]

Jean

 Thank you for your response. I have the options you suggested already set, and 
I am still not getting the dialplan to trigger. The message is being sent but 
nothing. I have tried with the auth both set to no and yes as well.

 accept_outofcall_message = yes
outofcall_message_context = messages
auth_message_requests = yes



Thanks
Bryant




 From: "Jean Aunis" <jean.au...@prescom.fr>
Sent: Friday, March 10, 2017 2:24 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Trying to get SMS from GXV3240 to trigger 
dialplan code.

This is not a SIP NOTIFY but a SIP MESSAGE (the first line in the logs is not 
related to the few next ones).

If you are using chan_sip, you have to activate out of call messages in 
sip.conf :

accept_outofcall_message=yes
outofcall_message_context=messages

Then in extensions.conf, define a context "messages" with the appropriate 
extensions (to stick to your example, it will be 16162995607) and use the 
function MESSAGE to retrieve the SMS content.

Best regards

Jean Aunis    Le 10/03/2017 à 00:21, Bryant Zimmerman a écrit :
  I am trying to send SMS from my grandstream GXV3240
 Asterisk receives the message in a NOTIFY block.

 How can I get asterisk to run dialplan code when receiving these Notify SMS 
Message Blocks.
 I can then route them to my SMS provider.

 Any ideas are appreciated. Below is debug of a message sent from the phone 
when received no dialplan code is triggered.
 I am wounding if I need to modify some setting in sip.conf or the peer config. 
 Incomming SMS from my vendor works without issue and is transmitted to the 
phone.


<->
--- (10 headers 0 lines) ---
Really destroying SIP dialog 
'3927411c7fe967886df6c8d0410d4...@xxx.xxx.xxx.xxx:5060' Method: NOTIFY

<--- SIP read from UDP:XXX.XXX.XXX.XXX:57568 --->
MESSAGE sip:16162995...@vgw0005.granddial.net SIP/2.0
Via: SIP/2.0/UDP 192.168.201.104:20093;branch=z9hG4bK1738682353;rport
From: <sip:6167761066.2...@vgw0005.granddial.net>;tag=1683585926
To: <sip:16162995...@vgw0005.granddial.net>
Call-ID: 1662412698-20093-...@bjc.bgi.cab.bae
CSeq: 9430 MESSAGE
Contact: <sip:6167761066.2003@192.168.201.104:20093>
Max-Forwards: 70
User-Agent: Grandstream GXV3240 1.0.3.158
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, 
UPDATE, MESSAGE
Content-Type: text/plain; charset=UTF-8
Content-Length: 5

Test Message SMS
<->

Thanks

Bryant



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[asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code.

[Bryant Zimmerman]

I am trying to send SMS from my grandstream GXV3240
 Asterisk receives the message in a NOTIFY block.
  
 How can I get asterisk to run dialplan code when receiving these Notify 
SMS Message Blocks.
 I can then route them to my SMS provider.
  
 Any ideas are appreciated. Below is debug of a message sent from the phone 
when received no dialplan code is triggered.
 I am wounding if I need to modify some setting in sip.conf or the peer 
config.  Incomming SMS from my vendor works without issue and is 
transmitted to the phone.
  

<->
--- (10 headers 0 lines) ---
Really destroying SIP dialog 
'3927411c7fe967886df6c8d0410d4...@xxx.xxx.xxx.xxx:5060' Method: NOTIFY  

<--- SIP read from UDP:XXX.XXX.XXX.XXX:57568 --->
MESSAGE sip:16162995...@vgw0005.granddial.net SIP/2.0
Via: SIP/2.0/UDP 192.168.201.104:20093;branch=z9hG4bK1738682353;rport
From: ;tag=1683585926
To: 
Call-ID: 1662412698-20093-...@bjc.bgi.cab.bae
CSeq: 9430 MESSAGE
Contact: 
Max-Forwards: 70
User-Agent: Grandstream GXV3240 1.0.3.158
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, 
UPDATE, MESSAGE
Content-Type: text/plain; charset=UTF-8
Content-Length: 5  

Test Message SMS
<-> 

Thanks

Bryant 

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Re: [asterisk-users] fail2ban Asterisk 13.13.1

[Bryant Zimmerman]

John V

 Are you using pjsip? We are have several test servers and  I just checked my 
/etc/fail2ban/filter.d/asterisk.conf and it is not updated for pjsip 
implementations.  Looking at the security log files and the regex I noticed 
that some items are being banned but others are not due to changes in the 
messages for pjsip.
 Anyone got an updated asterisk.conf for fail2ban.

 Bryant



 From: "Telium Technical Support" 
Sent: Wednesday, March 1, 2017 9:54 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] fail2ban Asterisk 13.13.1

If this is a small site, I recommend you download the free version of SecAst 
(www.telium.ca) and replace fail2ban.  SecAst does NOT use the log file, or 
regexes, to match etc.instead it talks to Asterisk through the AMI to extract 
security information.  Messing with regexes is a losing battle, and the lag in 
reading logs can allow an attacker 100+ registration attempts before fail2ban 
even does anything (assuming the IP is exposed in the Asterisk log).



If this is a large install then post in the commercial list for more 
information.



-Raj-



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support
Sent: Wednesday, March 1, 2017 2:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 

Subject: Re: [asterisk-users] fail2ban Asterisk 13.13.1



It's possible that you need to increase the value of 'findtime' to 
something greater than 300 secs. You also may want to set "timestamp = yes" in 
asterisk.conf so each line in the CLI will be time stamped. Time stamping it 
will be the definitive determination on whether or not the 'findtime' is the 
culprit.

Regards;

John V.



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motty Cruz
Sent: Wednesday, March 01, 2017 01:29 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] fail2ban Asterisk 13.13.1



Hello, fail2ban does not ban offending IP.



NOTICE[29784] chan_sip.c: Registration from 
'"user3"' failed for 'offending-IP:53417' - Wrong 
password

NOTICE[29784] chan_sip.c: Registration from 
'"user3"' failed for 'offending-IP:53911' - Wrong 
password





# A host is banned if it has generated "maxretry" during the last "findtime"

# seconds.

findtime  = 300



[asterisk-iptables]

enable = true

port = 5060,5061

filter   = asterisk

action   = iptables-allports[name=ASTERISK, protocol=all]

  sendmail[name=ASTERISK, dest=mo...@email.com, 
sender=fail2...@asterisk-ip.com]

#action   = %(banaction)s[name=%(__name__)s-tcp, port="%(port)s", 
protocol="tcp", chain="%(chain)s", actname=%(banaction)s-tcp]

   %(banaction)s[name=%(__name__)s-udp, port="%(port)s", 
protocol="udp", chain="%(chain)s", actname=%(banaction)s-udp]

   %(mta)s-whois[name=%(__name__)s, dest="%(destemail)s"]

logpath  = /var/log/asterisk/messages

maxretry = 3

findtime  = 300

bantime  = -1





in filter.d

asterisk.conf

failregex = ^%(__prefix_line)s%(log_prefix)s Registration from '[^']*' failed 
for '(:\d+)?' - (Wrong password|Username/auth name mismatch|No matching 
peer found|Not a local domain|Device does not match ACL|Peer is not supposed to 
register|ACL error \(permit/deny\)|Not a local domain)$

^%(__prefix_line)s%(log_prefix)s Call from '[^']*' \(:\d+\) 
to extension '[^']*' rejected because extension not found in context

^%(__prefix_line)s%(log_prefix)s Host  failed to authenticate 
as '[^']*'$

^%(__prefix_line)s%(log_prefix)s No registration for peer '[^']*' 
\(from \)$

^%(__prefix_line)s%(log_prefix)s Host  failed MD5 
authentication for '[^']*' \([^)]+\)$

^%(__prefix_line)s%(log_prefix)s Failed to authenticate 
(user|device) [^@]+@\S*$

^%(__prefix_line)s%(log_prefix)s hacking attempt detected ''$

^%(__prefix_line)s%(log_prefix)s 
SecurityEvent="(FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPassword)",EventTV="([\d-]+|%(iso8601)s)",Severity="[\w]+",Service="[\w]+",EventVersion="\d+",AccountID="(\d*|)",SessionID=".+",LocalAddress="IPV[46]/(UDP|TCP|WS)/[\da-fA-F:.]+/\d+",RemoteAddress="IPV[46]/(UDP|TCP|WS)//\d+"(,Challenge="[\w/]+")?(,ReceivedChallenge="\w+")?(,Response="\w+",ExpectedResponse="\w*")?(,ReceivedHash="[\da-f]+")?(,ACLName="\w+")?$

^%(__prefix_line)s%(log_prefix)s "Rejecting unknown SIP connection 
from "$

^%(__prefix_line)s%(log_prefix)s Request (?:'[^']*' )?from '[^']*' 
failed for '(?::\d+)?'\s\(callid: [^\)]*\) - (?:No matching endpoint 
found|Not match Endpoint(?: Contact)? ACL|(?:Failed|Error) to authenticate)\s*$



failregex = NOTICE.* .*: Registration 

Re: [asterisk-users] pjsip realtime - endpoints not loading - Solved

[Bryant Zimmerman]

It appears that res_odbc.so does not always load fast enough to allow the 
realtime mappings in the extconfig.conf to complete successfully at startup 
thus stopping the first load of the pjsip endpoints and other pjsip values. 

  
 The resolution for this is to preload the res_odbc.so and 
res_config_odbc.so in the modules.conf.  The realtime mappings then appear 
to complete correctly during startup and allows all the pjsip data to load 
correctly.

Thanks

Bryant



Sent: Wednesday, December 21, 2016 9:12 AM
Subject: [asterisk-users] pjsip realtime - endpoints not loading.   
 We are continuing to test our asterisk 13 pjsip deployments.

   I am running into an issue that I am assuming is a configuration 
problem, and am hoping someone can point me in the right direction. We are 
running pjsip in real-time mode using a database to store all the endpoint 
records. Our endpoint records with our carrier do not support 
registration.



   The issue I am having is when asterisk starts none of the non 
registration endpoints become available. They will not allow calls inbound 
or acknowledge qualify's. To get them to come on line we have to do a pjsip 
show endpoints, and then all works until asterisk is restarted.



   Is there any way to get the endpoints to load without manually doing the 
pjsip show endpoints?



   Any input is appreciated.

   
Thanks

   Bryant


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[asterisk-users] pjsip realtime - endpoints not loading.

[Bryant Zimmerman]

We are continuing to test our asterisk 13 pjsip deployments.
 I am running into an issue that I am assuming is a configuration problem, 
and am hoping someone can point me in the right direction. We are running 
pjsip in real-time mode using a database to store all the endpoint records. 
Our endpoint records with our carrier do not support registration.
  
 The issue I am having is when asterisk starts none of the non registration 
endpoints become available. They will not allow calls inbound or 
acknowledge qualify's. To get them to come on line we have to do a pjsip 
show endpoints, and then all works until asterisk is restarted.
  
 Is there any way to get the endpoints to load without manually doing the 
pjsip show endpoints?
  
 Any input is appreciated.

Thanks
 Bryant

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[asterisk-users] Fax faling on PJSip

[Bryant Zimmerman]

I am working on moving from version 11 to version 13 for my fax 
applications.
 We are bumping into an issue where the bulk of the T38 faxes are failing.

The sending test switch is reporting COMREC_ERR_TRANSMIT_PHASE
  
 These same faxes succeed on the 11 version of asterisk.
 I am wondering if there are any ideas? COMREC_ERR_TRANSMIT_PHASE
  
 Both servers are running the same version of spandsp. The dialplan code is 
the same on both.
  
 The only difference is the versions of asterisk and pjsip on the 13 
platform.
  
 Any ideas would be appreciated.
  
 Thanks

Bryant

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Re: [asterisk-users] Asterisk 13 T.38 Version 3?

[Bryant Zimmerman]

Does anyone know if Asterisk 13 will support T.38 Version 3?
 ?
 Thanks
 Bryant

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Re: [asterisk-users] pjsip transports from database.

[Bryant Zimmerman]

 
 On Friday, November 4, 2016 10:20 AM - Joshua Colp wrote:
 
 >>On Fri, Nov 4, 2016, at 10:26 AM, Bryant Zimmerman wrote:
>> Hey all
>>
>> I am trying to configure all my pjsip transports form a database table.
>> The issue I am running into is that pjsip is auto binding to 
0.0.0.0:5060
>> before it reads my list of transports from the database. This means 
that
>> my
>> entries for port 5060 are already bound and the settings in the 
database
>> are not loaded.
>>
>> When loading the transport form the .conf file it works as expected and
>> does not do an auto binding, but uses what is in the .conf
>>
>> Is there a way to have asterisk pjsip hold the default binding override
>> until after it has checked the database when sourcery .conf configures 
a
>> transport location other then pjsip.conf?

>>PJSIP has no auto binding or default binding. It will only bind to what
>>Is configured. Do you have it in both .conf and in realtime? Do you also
>>have chan_sip loaded?

Joshua
  
 You were correct. There was an old chan_sip.so in the bin folder that was 
being auto loaded. It was binding to 0.0.0.0:5060 causing the transports 
from the database for pjsip to fail. I forced down asterisk and deleted the 
chan_sip.so from the bin folder and the issue resolved. Looks like I need 
to go through and clean up old garbage from an earlier build so I don't get 
caught in the future. I also added a noload for chan_sip.so just incase one 
ever gets dropped back in the folder. 
  
 Much thanks for the direction here I spent a lot of time trying to figure 
out where the binding was coming from.
  
 Bryant

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[asterisk-users] pjsip transports from database.

[Bryant Zimmerman]

Hey all
  
 I am trying to configure all my pjsip transports form a database table. 
The issue I am running into is that pjsip is auto binding to 0.0.0.0:5060 
before it reads my list of transports from the database. This means that my 
entries for port 5060 are already bound and the settings in the database 
are not loaded.
  
 When loading the transport form the .conf file it works as expected and 
does not do an auto binding, but uses what is in the .conf
  
 Is there a way to have asterisk pjsip hold the default binding override 
until after it has checked the database when sourcery .conf configures a 
transport location other then pjsip.conf?

Thanks

Bryant

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Re: [asterisk-users] PJSIP - State of the art

[Bryant Zimmerman]

I agree the multi-domain environment is a nice idea, but too many endpoints 
don't properly support.
We to use a prefix in the SIP username for multi-domain environments.

Thanks
Bryant
  


 From: "Ludovic Gasc" 
Sent: Sunday, July 17, 2016 5:20 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] PJSIP - State of the art   
   2016-07-17 14:30 GMT+02:00 Annus Fictus :

The main idea of the new channel was working on a multi-domain environment 

  For now, to my experience, it's more future-proof compliant to use a 
prefix in the SIP username than multi-domain environment.
 Even if the multi-domain support was perfect in Asterisk, we tested some 
crappy SIP endpoints where in fact, even if you configure a domain name 
everywhere in the configuration, you have only IPs in SIP packets.
  
  We have that on production for our cloud plateform, it works pretty well 
and also simplify whitelabel handling.
 Moreover, if you have a good provisioning support, it will be invisible 
for your users.
   

  
 When I see the time needed to really use on production the SNI feature in 
SSL, and you have only 5 majors HTTP endpoints (aka Web browsers).
 In the SIP world, I'm not sure you can use multi domain except if you can 
force the SIP endpoints used by your clients.

, have more then one device registered with same credentials and have more 
stability. 
  Since 13.9.1, we have a better experience of pjsip.
 Nevertheless, not yet massively used on production for now, we planned to 
migrate endpoint by endpoint to minimize the risk. 

   

Be Better still with Asterisk 1.11.X? 
  Maybe you could use Asterisk 13 with chan_sip to start, it works pretty 
well and already think to support chan_pjsip in the same time.
 The benefit to think about that if one day you need to use an alternative 
channel like chan_iax2, it should be easier to implement for you.

   

Regards  

   

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Re: [asterisk-users] Call File - CPU spikes

[Bryant Zimmerman]

I am working on a project that we are seeing a 100% CPU spike when we move 
50 calls files to the folder.
  
 We are running pjsip and asterisk 13..It holds the spike for several 
minutes Are there any tunable that may help with this?
  

Thanks
Bryant

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Re: [asterisk-users] ? Re: Recommendations for free virtual server tech and Asterisk? (Ikka Tirtawidjaja)

[Bryant Zimmerman]

Hyper-V works well we run both OpenSuse and Debian with asterisk on it is rock 
solid,
 and it is free if you use the Hyper-V Server Version.

 Bryant


 From: "Saint Michael" 
Sent: Saturday, April 9, 2016 1:23 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: [asterisk-users] ? Re: Recommendations for free virtual server tech 
and Asterisk? (Ikka Tirtawidjaja)
  ?OpenVZ is useless for Asterisk or any other resource intensive application. 
OpenVZ was built from a hosting provider point if view, and if you exceed any 
of the counters, dozens of them, they system will
 kill your app immediately. It is almost impossible to build a VPS that will 
use all the resources of the machine.  The only container technology that works 
for Asterisk is LXC, better implemented by Ubuntu on the server side. Centos 7 
is behind in the LXC version and it is part of the core OS, but found in a 
repository. Yo need kernel 4.X to make it works flawlessly.
 ?


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[asterisk-users] Asterisk 13 - Call Bridge issue.

[Bryant Zimmerman]

Even when using the U option just issuing the Answer does not seem to 
always work. I end up having to play a prompt of some sort to force the 
answer.. There has to be some kind of bug going on here.
  
 Thanks

Bryant 
  


 From: "Bryant Zimmerman" <brya...@zktech.com>
Sent: Thursday, March 31, 2016 6:54 PM
To: brya...@zktech.com, "Asterisk Users Mailing List - Non-Commercial 
Discussion" <asterisk-users@lists.digium.com>
Subject: re: [asterisk-users] Asterisk 13 - Call Bridge issue.   
  

--------

From: "Bryant Zimmerman" <brya...@zktech.com>
Sent: Thursday, March 31, 2016 6:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk 13 - Call Bridge issue.   
 I have the following scenario.



   Call file calls 1st party.

   When connected give called party option to connect to second party.



   Issue Dial to second party. Caller answers and the two are bridged 
together.

   My issue is that 4 out of 5 calls fail to bridge the audio.



   Am I  missing something or is there some kind of bug? Here is my test 
dialplan 





;Dialer Base Code Files.
;Variables are sent in from .call file
[calluser-intake]
exten => s,1,NoOp(Start Call Intake)
exten => s,2,NoOp(Setup any vars)
exten => s,n,Set(_g_pmtPath=/vapp/dialerprompts/)
exten => s,n,NoOp(What is Path = ${g_pmtPath})
exten => s,n,NoOp(Read Call File Vars)
exten => s,n,NoOp(Dial To - ${l_DialTo})
exten => s,n,NoOp(Proxy - Proxy.${l_Proxy})
exten => s,n,NoOp(Carrier Trunk - ${l_Carrier})
exten => s,n,Set(_l_CallerIDnum=${CALLERID(num)})
exten => s,n,Set(CALLERID(num)=${g_SIPUser})
exten => 
s,n,Dial(PJSIP/${l_DialTo}@proxy_${l_Proxy},30,b(dialer-header^s^1)G(dialer-
playmsg^s^1))  

[dialer-header]
exten => s,1,Set(PJSIP_HEADER(add,X-Carrier)=${l_Carrier})
same => n,Set(PJSIP_HEADER(add,X-CallerID)=${l_CallerIDnum})
same => n,NoOp(X-Carrier = ${PJSIP_HEADER(read,X-Carrier)})
same => n,Set(CONNECTEDLINE(number,i)=vap_002)
same => n,DumpChan(1)
same => n,Return()  

[dialer-playmsg]
exten => s,1,Goto(hold,1)
same => n,NoOp(Enter Play Message)
same => n,NoOp(Path = ${g_pmtPath})
same => n,SayAlpha(${g_SIPUser})
same => n,BackGround(${g_pmtPath}Intro)
same => n,WaitExten(60)  

exten => 2,1,NoOp(Dial Through)
same => n,Set(_l_CallerIDnum=616831)
same => n,Set(_l_Carrier=0001)
same => n,Set(l_DialTo=6167761066)
same => n,Set(l_Proxy=002)
same => n,Dial(PJSIP/${l_DialTo}@proxy_001,30,b(dialer-header^s^1))  

exten => _X,1,NoOp(Digit Entry)
exten => _X,n,NoOp(Log Response)
exten => _X,n,Playback(${g_pmtPath}YouPressed)
exten => _X,n,SayNumber(${EXTEN})  

exten => hold,1,NoOp(Park Called)
exten => hold,n,While($[1 < 5])
exten => hold,n,Wait(90)
exten => hold,n,EndWhile 



   Any ideas on why the media would not flowing after it sates they bridge 
has completed



   Another point. If I use a b option in the second dial. to call another 
context on connect of the second call. I get audio played on that both 
caller and callee channels.



   Thanks

Bryant 
  Ok it appears that the channel is not answering when it bridges the two 
calls together. 
 If I use the U option to gosub to a context to force an Answer() before 
the bridge then things seem to work. I also tried the lower case "a" option 
to force the answer and nothing happens with it appears to be ignored. .. 
So the U option with a gosub to an Answer seems to be the only way to get 
this to work...
  
 This seems like a bug. Should the called channel answer when a call is 
made with the Dial() function? Can anyone chime in on this one. 
  
 Note: Current systems are on Asterisk 13.5.0 (So if this was a bug has it 
been fixed in the latest release.) I did not see anything in the change 
logs that I would attribute to this.
  
 Thanks 
 Bryant


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Re: [asterisk-users] Asterisk 13 - Call Bridge issue.

[Bryant Zimmerman]

 



From: "Bryant Zimmerman" <brya...@zktech.com>
Sent: Thursday, March 31, 2016 6:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk 13 - Call Bridge issue.   
 I have the following scenario.



   Call file calls 1st party.

   When connected give called party option to connect to second party.



   Issue Dial to second party. Caller answers and the two are bridged 
together.

   My issue is that 4 out of 5 calls fail to bridge the audio.



   Am I  missing something or is there some kind of bug? Here is my test 
dialplan 





;Dialer Base Code Files.
;Variables are sent in from .call file
[calluser-intake]
exten => s,1,NoOp(Start Call Intake)
exten => s,2,NoOp(Setup any vars)
exten => s,n,Set(_g_pmtPath=/vapp/dialerprompts/)
exten => s,n,NoOp(What is Path = ${g_pmtPath})
exten => s,n,NoOp(Read Call File Vars)
exten => s,n,NoOp(Dial To - ${l_DialTo})
exten => s,n,NoOp(Proxy - Proxy.${l_Proxy})
exten => s,n,NoOp(Carrier Trunk - ${l_Carrier})
exten => s,n,Set(_l_CallerIDnum=${CALLERID(num)})
exten => s,n,Set(CALLERID(num)=${g_SIPUser})
exten => 
s,n,Dial(PJSIP/${l_DialTo}@proxy_${l_Proxy},30,b(dialer-header^s^1)G(dialer-
playmsg^s^1))  

[dialer-header]
exten => s,1,Set(PJSIP_HEADER(add,X-Carrier)=${l_Carrier})
same => n,Set(PJSIP_HEADER(add,X-CallerID)=${l_CallerIDnum})
same => n,NoOp(X-Carrier = ${PJSIP_HEADER(read,X-Carrier)})
same => n,Set(CONNECTEDLINE(number,i)=vap_002)
same => n,DumpChan(1)
same => n,Return()  

[dialer-playmsg]
exten => s,1,Goto(hold,1)
same => n,NoOp(Enter Play Message)
same => n,NoOp(Path = ${g_pmtPath})
same => n,SayAlpha(${g_SIPUser})
same => n,BackGround(${g_pmtPath}Intro)
same => n,WaitExten(60)  

exten => 2,1,NoOp(Dial Through)
same => n,Set(_l_CallerIDnum=616831)
same => n,Set(_l_Carrier=0001)
same => n,Set(l_DialTo=6167761066)
same => n,Set(l_Proxy=002)
same => n,Dial(PJSIP/${l_DialTo}@proxy_001,30,b(dialer-header^s^1))  

exten => _X,1,NoOp(Digit Entry)
exten => _X,n,NoOp(Log Response)
exten => _X,n,Playback(${g_pmtPath}YouPressed)
exten => _X,n,SayNumber(${EXTEN})  

exten => hold,1,NoOp(Park Called)
exten => hold,n,While($[1 < 5])
exten => hold,n,Wait(90)
exten => hold,n,EndWhile 



   Any ideas on why the media would not flowing after it sates they bridge 
has completed



   Another point. If I use a b option in the second dial. to call another 
context on connect of the second call. I get audio played on that both 
caller and callee channels.



   Thanks

Bryant 
  Ok it appears that the channel is not answering when it bridges the two 
calls together. 
 If I use the U option to gosub to a context to force an Answer() before 
the bridge then things seem to work. I also tried the lower case "a" option 
to force the answer and nothing happens with it appears to be ignored. .. 
So the U option with a gosub to an Answer seems to be the only way to get 
this to work...
  
 This seems like a bug. Should the called channel answer when a call is 
made with the Dial() function? Can anyone chime in on this one. 
  
 Note: Current systems are on Asterisk 13.5.0 (So if this was a bug has it 
been fixed in the latest release.) I did not see anything in the change 
logs that I would attribute to this.
  
 Thanks 
 Bryant


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[asterisk-users] Asterisk 13 - Call Bridge issue.

[Bryant Zimmerman]

I have the following senerio.
  
 Call file calls 1st party.
 When connected give called party option to connect to second party.
  
 Issue Dial to second party. Caller answers and the two are bridged 
together.
 My issue is that 4 out of 5 calls fail to bridge the audio.
  
 Am I  missing something or is there some kind of bug? Here is my test 
dialplan 
  

;Dialer Base Code Files.
;Variables are sent in from .call file
[calluser-intake]
exten => s,1,NoOp(Start Call Intake)
exten => s,2,NoOp(Setup any vars)
exten => s,n,Set(_g_pmtPath=/vapp/dialerprompts/)
exten => s,n,NoOp(What is Path = ${g_pmtPath})
exten => s,n,NoOp(Read Call File Vars)
exten => s,n,NoOp(Dial To - ${l_DialTo})
exten => s,n,NoOp(Proxy - Proxy.${l_Proxy})
exten => s,n,NoOp(Carrier Trunk - ${l_Carrier})
exten => s,n,Set(_l_CallerIDnum=${CALLERID(num)})
exten => s,n,Set(CALLERID(num)=${g_SIPUser})
exten => 
s,n,Dial(PJSIP/${l_DialTo}@proxy_${l_Proxy},30,b(dialer-header^s^1)G(dialer-
playmsg^s^1))  

[dialer-header]
exten => s,1,Set(PJSIP_HEADER(add,X-Carrier)=${l_Carrier})
same => n,Set(PJSIP_HEADER(add,X-CallerID)=${l_CallerIDnum})
same => n,NoOp(X-Carrier = ${PJSIP_HEADER(read,X-Carrier)})
same => n,Set(CONNECTEDLINE(number,i)=vap_002)
same => n,DumpChan(1)
same => n,Return()  

[dialer-playmsg]
exten => s,1,Goto(hold,1)
same => n,NoOp(Enter Play Message)
same => n,NoOp(Path = ${g_pmtPath})
same => n,SayAlpha(${g_SIPUser})
same => n,BackGround(${g_pmtPath}Intro)
same => n,WaitExten(60)  

exten => 2,1,NoOp(Dial Through)
same => n,Set(_l_CallerIDnum=616831)
same => n,Set(_l_Carrier=0001)
same => n,Set(l_DialTo=6167761066)
same => n,Set(l_Proxy=002)
same => n,Dial(PJSIP/${l_DialTo}@proxy_001,30,b(dialer-header^s^1))  

exten => _X,1,NoOp(Digit Entry)
exten => _X,n,NoOp(Log Response)
exten => _X,n,Playback(${g_pmtPath}YouPressed)
exten => _X,n,SayNumber(${EXTEN})  

exten => hold,1,NoOp(Park Called)
exten => hold,n,While($[1 < 5])
exten => hold,n,Wait(90)
exten => hold,n,EndWhile 
   
 Any ideas on why the media would not flowing after it sates they bridge 
has completed
  
 Another point. If I use a b option in the second dial. to call another 
context on connect of the second call. I get audio played on that both 
caller and callee channels.
  
 Thanks

Bryant 

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Re: [asterisk-users] 2 devices same *actual* extension - can it be done

[Bryant Zimmerman]

With Asterisk 13 you may be able to do it with PJSIP using two separate 
connections on the same AOR
 I believe you would have two separate endpoints that would register under the 
same user and auth. If I understand it correctly when you send a call to the 
AOR both registered endpoints would be rung.  I have not tried inbound ring 
yet, but when I have registered for out bound multiple connections and it seems 
to work well.

 Bryant

 From: "Kevin Long" 
Sent: Wednesday, March 9, 2016 1:42 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: [asterisk-users] 2 devices same *actual* extension - can it be done


Hello,

My company has invested heavily in Counterpath's Stretto provisioning platform 
for Mobile and Desktop VoIP clients .

At this time their system allows 2 devices (for example iPhone + desktop 
computer) using the same software license per user , which many of our users 
require.

Their provisioning system assumes that both devices will use the same SIP 
extension for auth however.

Normally we would use separate extensions and a follow-me , but if there is any 
way to use the same extension, I need to figure it out.

Thank you,

Kevin Long--
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Re: [asterisk-users] Grandstream Early Dial

[Bryant Zimmerman]

Jean

 If you moved the exten => _.   Lines to the bottom of the context then you 
should like be able to get away from having to have two separate contexts. I 
use that method quiet often, but was in a hurry to get you a response and did 
not think remember that nuance.

 I will have to try this as we are a heavy grandstream shop. It has been 
something on the list.

 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003



 From: "Jean-Denis Girard" <jd.gir...@sysnux.pf>
Sent: Friday, February 19, 2016 11:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Grandstream Early Dial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi Bryant,

Thanks for your reply.

It didn't work immediately, I had to create a second context, or else it
was looping between the second and first line. This seems to work:

[earlydial] ; Test Early Dial
exten => _.,1,Set(l_Extension=${EXTEN})
exten => _.,n,Goto(earlydial2,${l_Extension},1)

[earlydial2]
exten => _.,n,Goto(noMatch,1)
exten => noMatch,1, Incomplete(n)

exten => i,1,Goto(noMatch,1)
exten => t,1,Goto(noMatch,1)

exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
same => n,Playback(extension)
same => n,SayDigits(${EXTEN})
same => n,Hangup()

Best regards,
- --
Jean-Denis Girard

SysNux Systèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27

Le 19/02/2016 03:31, Bryant Zimmerman a écrit :
> Jean-Denis Girard
>
> I have not used the Incomplete yet, but you might be able to do
> something like this.
>
> [earlydial]
>
> exten => _.,1,Set(l_Extension = ${EXTEN})
> exten => _.,n,Goto(${l_Extension},1)
> exten => _.,n,Goto(noMatch,1)
>
> exten => i,1,Goto(noMatch,1)
>
> exten => noMatch,1, Incomplete(n)
>
> exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
> same => n,Playback(extension)
> same => n,SayDigits(${EXTEN})
> same => n,Hangup()
>
>
> I wrote this in this message and have not tested this so use with
> caution. There may be syntactical issues, but the concept might work f
or
> you.
>
> Bryant
>
> --
- --
> *From*: "Jean-Denis Girard" <jd.gir...@sysnux.pf>
> *Sent*: Thursday, February 18, 2016 8:02 PM
> *To*: asterisk-users@lists.digium.com
> *Subject*: Re: [asterisk-users] Grandstream Early Dial
>
> Le 18/02/2016 11:03, Richard Mudgett a écrit :
>> I've been using Grandstream phones for more than 10 years, but onl
> y
>> yesterday tried to use Early Dial... and I failed. What is needed
> on the
>> Asterisk side to reply 484 to INVITE? Phones are talking to chan_p
> jsip
>> on Asterisk-13.7.1.
>
>
>> Look into the Incomplete application.
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_In
c
> omplete
>
> Thanks for prompt answer Richard.
>
> Actually I had already tried the Incomplete application, but failed to
> add the "n" option, and this seems mandatory for SIP. I find the help
> text misleading : "NOTE: Most channel types need to be in Answer state
> in order to receive DTMF".
>
> This is my test dialplan:
>
> [earlydial] ; Test Early Dial
> exten => 1,1,Verbose(2,Incomplete 1 test)
> same => n,Incomplete(n)
>
> exten => _1X,1,Verbose(2,Incomplete 1X test)
> same => n,Incomplete(n)
>
> exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
> same => n,Playback(extension)
> same => n,SayDigits(${EXTEN})
> same => n,Hangup()
>
> It works, but seems a bit complicated: is this the correct way to use
> Incomplete ?
>
>
> Thanks,
>
> --
> _
> -- 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
>
>
>

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Re: [asterisk-users] Grandstream Early Dial

[Bryant Zimmerman]

Jean-Denis Girard

 I have not used the Incomplete yet, but you might be able to do something like 
this.

 [earlydial]

 exten => _.,1,Set(l_Extension = ${EXTEN})
 exten => _.,n,Goto(${l_Extension},1)
 exten => _.,n,Goto(noMatch,1)

 exten => i,1,Goto(noMatch,1)

 exten => noMatch,1, Incomplete(n)

 exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
  same => n,Playback(extension)
  same => n,SayDigits(${EXTEN})
  same => n,Hangup()


 I wrote this in this message and have not tested this so use with caution. 
There may be syntactical issues, but the concept might work for you.

 Bryant



 From: "Jean-Denis Girard" 
Sent: Thursday, February 18, 2016 8:02 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Grandstream Early Dial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Le 18/02/2016 11:03, Richard Mudgett a écrit :
> I've been using Grandstream phones for more than 10 years, but onl
y
> yesterday tried to use Early Dial... and I failed. What is needed
on the
> Asterisk side to reply 484 to INVITE? Phones are talking to chan_p
jsip
> on Asterisk-13.7.1.
>
>
> Look into the Incomplete application.
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Inc
omplete

Thanks for prompt answer Richard.

Actually I had already tried the Incomplete application, but failed to
add the "n" option, and this seems mandatory for SIP. I find the help
text misleading : "NOTE: Most channel types need to be in Answer state
in order to receive DTMF".

This is my test dialplan:

[earlydial] ; Test Early Dial
exten => 1,1,Verbose(2,Incomplete 1 test)
same => n,Incomplete(n)

exten => _1X,1,Verbose(2,Incomplete 1X test)
same => n,Incomplete(n)

exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
same => n,Playback(extension)
same => n,SayDigits(${EXTEN})
same => n,Hangup()

It works, but seems a bit complicated: is this the correct way to use
Incomplete ?

Thanks,
- --
Jean-Denis Girard

SysNux Systèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
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=g2Jy
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Re: [asterisk-users] How to execute a macro after dial but before connect

[Bryant Zimmerman]

Phillip

 Check out the b and B options one of them should do what you want.

 https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial

 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003



 From: "Saint Michael" <vene...@gmail.com>
Sent: Friday, February 19, 2016 8:05 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] How to execute a macro after dial but before connect
  ?Dear friends:
Is there a way to execute a macro or sub-routine after we send the invite 
before we receive anything like a 200 OK, 183, etc??
 Philip


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Re: [asterisk-users] Voicemail issue on Grandstream GXP2000 phones

[Bryant Zimmerman]

Richard
  
 Check both the DTMF settings, and the DialPlan string for account 3 on the 
phone.
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: "Richard Schroeder" <rsch...@gmail.com>
Sent: Tuesday, February 9, 2016 12:58 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] Voicemail issue on Grandstream GXP2000 phones   
Perhaps this is not limited to Grandstream GXP 2000 phones, but those 
are the phones we are using.
  
 Using FreePBX.
  
 Retrieving a voice message (*97) works fine from Line 1.
 Retrieving a voice message (*98) and picking the extension (Comedian mail) 
works fine from Line 1.
  
 From Line 3, it does not recognize the password. (*97 or *98). The 
extension is installed on Line 3. Retrieving Line 3's voice messages can 
only be done from Line 1 (on any extension on the PBX). Line 3 seems to 
work fine otherwise.
  
 Is this a limitation, or is it some kind of setup issue?
 I can't seem to find anything in the documentation for the phone or 
FreePBX related to this issue.
  
 Anyone? This is frustrating and I will be grateful for any help.
  
 Thank you!
  
 Richard
  
  

--  Richard C. Schroeder
rsch...@gmail.com
rsch...@optonline.net
516-859-1129 - Cell


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Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

[Bryant Zimmerman]

Sonny
  
 We use a real-time database for adding pjsip users. If you want to do it 
from the pjsip.conf you would have to write to the file from a script of 
some sort and then trigger a reload.   There is a real-time implementation 
for the extensions.conf as well. I personally use scripts for most of my 
dialplan, but in some cases I write to files included in my dialplan from a 
script and force a reload. 
  
 To directly answer you question I do not believe there is an API baked 
into asterisk to update the pjsip.conf and extensions.conf directly from 
the dialplan.
  
 Thanks

Bryant
  


 From: "Sonny Rajagopalan" 
Sent: Thursday, January 28, 2016 7:35 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP 
users and dialplans programmatically using API   
 Hi,  
 I am using Asterisk 13.6.0 and was wondering if I can programmatically add 
users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk 
server using API of some sort.
  
 Please do let me know.
  
 Thanks,
 Sonny.


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Re: [asterisk-users] PJSIP Stun/ICE

[Bryant Zimmerman]

George

 Reloading transports is one critical part and it sounds like you are making 
headway on that.  I have yet to be able to get transports to load from a 
real-time table using sorcery.conf
 If I would get the transports pulling from real-time as the (documentation 
says is possible but I have found no working examples yet) and then be able to 
reload any changes without forcing a compete asterisk restart. This would allow 
for a host of options for detecting and updating IP addresses.  In the long run 
it would be nice to be able to tie some kind of stun support for updating the 
external media and signaling IP addresses.

 Thanks

Bryant



 From: "George Joseph" 
Sent: Thursday, January 28, 2016 9:12 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] PJSIP Stun/ICE
  On Thu, Jan 28, 2016 at 6:58 PM, James Cloos  wrote:
   > "AS" == A J Stiles  writes:

AS> If you are paying for a business-grade Internet connection, you
AS> should get a static IP address -- or a block of them -- as
AS> standard.  Maybe you need to change your ISP?

In some places (including here) static ip is not affordable.
  ?Please create a JIRA issue and let me know what the number is.  I've just 
posted a patch for review that allows reloading transports from the command 
line.?  I'd like to know what else you actually need.



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Re: [asterisk-users] PJSIP Stun/ICE

[Bryant Zimmerman]

Joshua
  
 So once a transport is pulled from the transports table in realtime during 
asterisk startup it can't get any updates?
 Can a new transport be added to the table and the associated endpoints be 
updated to use the new transport, or are transport types only read at 
startup across the board?
  
 Thanks

Bryant
  


 From: "Joshua Colp" <jc...@digium.com>
Sent: Tuesday, January 26, 2016 8:10 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] PJSIP Stun/ICE   
Bryant Zimmerman wrote:
> Joshua
> Since there is no automated way currently built in to update the
> external signaling and media address information.
> Does the realtime pjsip support having the transport contexts section
> being pulled from a database table?
> I was thinking a cron script updating the table and forcing a reload
> each time an IP address changed might a workable solution.

No, once loaded the transports can not be changed.

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[asterisk-users] PJSIP - Realtime - Transports module?

[Bryant Zimmerman]

Does anyone know which module the type=transport loads under.
 I am trying to set up transports to load from a realtime table. I added 
the following under [res_pjsip] and it does not poll the associated 
database.
  
 [res_pjsip]
transport=realtime,vap002_ps_transports
  
 We also set the associated values in extconfig.conf as well. My best guess 
is that transports are loaded under a different module's context. Anyone 
have an idea?
  
 Thanks

Bryant

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Re: [asterisk-users] PJSIP Stun/ICE

[Bryant Zimmerman]

Daniel
  
 Thank you for your response. I was considering this as well. I have a 
script that monitors the IP Address now. I was hoping to use the real-time 
transports table now that alembic creates. I am trying to figure out which 
pjsip module is responsible for the transports contexts as I need to now 
configure it in the sorcery.conf file. I thought it would be under the 
[res_pjsip] context, but it is not even trying to pull from my transports 
table when it is there.  I am hoping someone will know what module it is in 
so I can move my configuration under the correct context.
  
 Thanks

Bryant
  


 From: "Daniel Heckl" <daniel.he...@gmail.com>
Sent: Tuesday, January 26, 2016 10:15 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] PJSIP Stun/ICE   
Bryant,

I have the same problem with dynamic public IPs and PJSIP. What is your 
idea to solve the problem?

My suggestion would be to write a script that monitors the change, 
pjsip.transports.conf updated and Asterisk restarts?

Daniel

> Am 26.01.2016 um 14:21 schrieb Joshua Colp <jc...@digium.com>:
>
> Bryant Zimmerman wrote:
>> Joshua
>> So once a transport is pulled from the transports table in realtime
>> during asterisk startup it can't get any updates?
>> Can a new transport be added to the table and the associated endpoints
>> be updated to use the new transport, or are transport types only read 
at
>> startup across the board?
>
> Transports can only be loaded at startup. This stems from PJSIP not being 
dynamic with transports (it doesn't like its environment changed to that 
degree while in use). I'm afraid if your IP changes you'd have to restart 
Asterisk when you are using PJSIP.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
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Re: [asterisk-users] PJSIP Stun/ICE

[Bryant Zimmerman]

Joshua
  
 I look forward to improvements as time goes on with PJSIP.
 I have been trying all day to get the Transport objects to pull from a 
real-time table. The documentation says it is possible, but does not show 
any examples. I am hoping to have the Transports pulled from the table at 
asterisk startup and then add additional as necessary. Using reloads to 
make the new Transports available. I understand the limitation of not being 
able to change existing and can live with that for now.   
  
 Do you know if there is anything special I have to do in the sorcery.conf 
to make the Transports pull from the real-time side of things. All my other 
tables are working.
  
 I disagree with the user that things PJSIP is worthless. There are some 
issues to work out long term, and documentation will get better over time 
as more of us work with it and contribute back.  Thanks for all you have 
assisted with around PJSIP.
  
 Bryant 


 From: "Joshua Colp" 
Sent: Tuesday, January 26, 2016 8:40 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] PJSIP Stun/ICE   
James Cloos wrote:
>> "JC" == Joshua Colp writes:
>
> JC> This stems from PJSIP not being dynamic with transports (it
> JC> doesn't like its environment changed to that degree while
> JC> in use). I'm afraid if your IP changes you'd have to restart
> JC> Asterisk when you are using PJSIP.
>
> Wow.
>
> I say this having voted for pjsip over the listed alternatives back when
> the plan to depricate chan_sip was first floated:
>
> That should have excluded pj from the options. Which of course means
> there were no reasonable options.

PJSIP doesn't like changing existing transports, the NAT functionality
is provided by the Asterisk implementation and can't be reloaded as a
side effect due to the heavy handed restriction. With work it could be
changed to allow the non low level things to be changed. What you can't
do with PJSIP is create a UDP transport, reload, and have it removed.
Once it's there it is there unless you restart.

>
> Can ari get around that bug?

ARI is a REST interface to Asterisk, it doesn't have anything to do with
this.

>
> Lack of full support for traversing nat makes pjsip worthless for a
> large number of users. And the whole point of realtime is to have all
> of the rt config fully dymanic.

I disagree that it makes it worthless for a large number of users. It's
only within the last few days that a few people have run into this
particular issue where they have a public IP address that is changing a
lot and PJSIP does not support changing it without a restart. If it were
a huge sweeping issue we'd be seeing it more often. If it continues to
show up a community member or us (heck maybe even myself in my spare
time) may look into implementing it.

>
> If ari cannot avoid that limitation, chan_sip should get full ongoing
> maintainance until pjsip is fixed.

The support level for chan_sip has already been changed and was
announced long ago. Patches will continue to be accepted for it and
community members can support it. We (Digium) are putting our effort
towards PJSIP.

--
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] PJSIP Stun/ICE

[Bryant Zimmerman]

Joshua
  
 Since there is no automated way currently built in to update the external 
signaling and media address information.
 Does the realtime pjsip support having the transport contexts section 
being pulled from a database table?
 I was thinking a cron script updating the table and forcing a reload each 
time an IP address changed might a workable solution.
  
 Thanks
 Bryant
  


 From: "Joshua Colp" <jc...@digium.com>
Sent: Tuesday, January 26, 2016 7:39 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] PJSIP Stun/ICE   
Bryant Zimmerman wrote:
> I have an asterisk 13 server behind NAT on a dynamic IP Address. It is
> running the PJSIP Stack
> It is registering to another asterisk 13 server that is on a Static IP
> outside of the firewall at a different location (also on the PJSIP 
Stack).
> How do we implement STUN/ICE on the server behind the dynamic Address.
> It does not appear to be registering properly without knowing the NAT
> pubic address. When I manually add external_media_address and
> external_signaling_address to the pjsipconfig registration seams to
> work, but knowing that the IP could change really means I need some kind
> of STUN/ICE similar to what we ran with chan_sip.
> I can find limited documentation on this, and what I have found does not
> show how to set a stun server to make the ice_support field work on an
> endpoint.
> Can anyone advise where I could find an answer to this.
> Thanks in advance for any ideas you can offer.
> Bryant

The res_pjsip module does not currently support an auto-updating
mechanism for the external signaling and media address information.

--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
 

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[asterisk-users] PJSIP Stun/ICE

[Bryant Zimmerman]

I have an asterisk 13 server behind NAT on a dynamic IP Address. It is 
running the PJSIP Stack
 It is registering to another asterisk 13 server that is on a Static IP 
outside of the firewall at a different location (also on the PJSIP Stack).
  
 How do we implement STUN/ICE on the server behind the dynamic Address. It 
does not appear to be registering properly without knowing the NAT pubic 
address.  When I manually add external_media_address and 
external_signaling_address to the pjsipconfig registration seams to work, 
but knowing that the IP could change really means I need some kind of 
STUN/ICE similar to what we ran with chan_sip. 
 I can find limited documentation on this, and what I have found does not 
show how to set a stun server to make the ice_support field work on an 
endpoint.
  
 Can anyone advise where I could find an answer to this.
  
 Thanks in advance for any ideas you can offer.
  
 Bryant

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[asterisk-users] PJSIP NAT traversal.

[Bryant Zimmerman]

 
 I have two servers running pjsip they are both on NAT. The proxy has a 
static public address.
 I set the ;external_media_address=203.0.113.1 and 
;external_signaling_address=203.0.113.1  to the actual IP address in the 
transport section on the proxy.
  
 The issue I am having is on the server with only a dynamic IP address. I 
can not figure out how to get ice support working so the public ip address 
is written into the registration.. The nat dynamic server is trying to 
register to the proxy.
  
 One issue I am seeing is only the private IP address are showing in the 
contact table on the proxies contact record.
  
 What could I be missing?
  
 Thanks

Bryant 

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Re: [asterisk-users] Using external RTP proxy for res_pjsip

[Bryant Zimmerman]

Dmitrity
  
 What kind of volume are you running?
 You can use asterisk as a proxy if you set it up correctly. The choice 
would fall on the volume and the operational needs.
 To use an external proxy you would either need to register to the proxy or 
have a trusted IP to IP relationship. If your carrier allows for endpoint 
registration then you could attached your asterisk server directly, and 
would not need a proxy. If you have an IP for a proxy you could also do a 
NAT translation from that IP directly to the Asterisk server and negate the 
need for a proxy all together.
  
 As far as connecting to a proxy. You would need a pjsip endpoint either 
with a trusted IP or with a registration to your proxy server.
  
 As far as exactly what should go in your pjsip.conf that depends on your 
final implementation. You have not given enough detail of your network 
situation, and the reasons for a proxy to adequately advise you any 
deeper.
 
 Good luck. I hope some of the above is helpful to you.
  
 Bryant
  


 From: "Dmitriy Serov" 
Sent: Monday, November 2, 2015 9:10 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: [asterisk-users] Using external RTP proxy for res_pjsip   
The asterisk server has a permanent IP address, but the provider cannot 
ensure stable quality traffic for RTP.

There is a desire to use an external server, the address of which shall be 
specified in the SDP, through which flowing media.
I use asterisk 13.6 and res_pjsip.

Prompt, please:
1. what software can be used on an external RTP proxy?
2. What settings need to be done in pjsip.conf to use this external RTP 
proxy?

Preferably specifies the external RTP proxy to specify a specific endpoint, 
not globally. If only globally valid, the suit and the decision.

I would be grateful for any clues.

Dmitriy Serov.
 

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Re: [asterisk-users] Remote UNIX connection / disconnected.

[Bryant Zimmerman]

Anyone know how to suppress the -- Remote UNIX connection / disconnected 
messages.
 I have a monitoring application that calls asterisk from the command line 
to verify some uptime stats. I would like to not have the console log the 
connections.. Any ideas are appreciated.
  
 Thanks

Bryant

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Re: [asterisk-users] pjsip show xxxx like endpoint?

[Bryant Zimmerman]

George, and Matthew
  
 I can open an issue later today, but if you want to do it that would be 
awesome as well. Please post the issue number back to this thread so I can 
follow it.
  
 Ideally the Like would work with all pjsip show commands   so we can 
reduce the list and drill down just like we could with sip show 
commands  This is a big missing for me right now and is really stopping 
me from going production along with the realtime performance issues already 
being talked about.
  
 Thanks for your assistance. I greatly appreciate it.
  
 Thanks

Bryant 


 From: "Matthew Jordan" 
Sent: Sunday, October 18, 2015 7:08 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] pjsip show  like endpoint?   
On Sun, Oct 18, 2015 at 12:39 PM, George Joseph
 wrote:
> Did you open a Jira issue for this yet? I can actually work on this this
> week.
>

I think it'd be pretty cool.

George: want me to open an issue?

--
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Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] pjsip show xxxx like endpoint?

[Bryant Zimmerman]

George and Mat
  
 Here is the link to the Jar Issue.
  
 https://issues.asterisk.org/jira/browse/ASTERISK-25477
  
 Thanks

Bryant
  


 From: "George Joseph" 
Sent: Sunday, October 18, 2015 10:17 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] pjsip show  like endpoint?   
 On Sun, Oct 18, 2015 at 5:07 PM, Matthew Jordan  
wrote:  On Sun, Oct 18, 2015 at 12:39 PM, George Joseph
 wrote:
> Did you open a Jira issue for this yet?  I can actually work on this 
this
> week.
>

I think it'd be pretty cool.

George: want me to open an issue?
   Thanks Matt.   Bryant said he'd do it tonight.
  
  


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Re: [asterisk-users] pjsip database error when using MS SQL via ODBC

[Bryant Zimmerman]

 I have a project that is requiring the use of MS SQL from asterisk. I get 
an error when the pjsip contact tries to update the contact table.
  
 [Oct 15 21:34:55] WARNING[3033]: res_odbc.c:649 
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 22018: 
[FreeTDS][SQL Server]Conversion failed when converting the varchar value 
'3.00' to data type int. (101)
  
 The datatype in MySQL is integer and in MS SQL is integer. What could be 
the cause of this? Is it likely some kind of FeeTDS conversion issue?  If I 
change the MS SQL type to double the error goes away, but I am unsure of 
the long term issues associated with this.
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003

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[asterisk-users] pjsip show xxxx like endpoint?

[Bryant Zimmerman]

Is there a way to limit the items returned by pjsip show [type] using like
 chan_sip allowed for sip show peers like , but I can't seem to figure 
out how to lookup or limit my returns with pjsip
  
 Thanks

Bryant

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Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

[Bryant Zimmerman]

 


 From: "Ryan, Travis" 
Sent: Monday, October 5, 2015 8:20 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from 
ODBC   
 Ah ok, thanks.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Monday, October 05, 2015 8:20 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from 
ODBC

On 15-10-05 09:16 AM, Ryan, Travis wrote:

[snip]

>
>
> So should anyone using realtime PJSIP be using the registrations line? 
Even if it's not used for any trunking?

A registrations line in sorcery.conf for res_pjsip would do absolutely 
nothing. If you put it under res_pjsip_outbound_registration and have no 
outbound registrations it will execute some queries against your database 
but otherwise do nothing.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: 
www.digium.com & www.asterisk.org

--
I am working on a step by step and some internal documentation on pjsip.
 Where would the best place to post this information be.  I am willing to 
share what I have it may help others in the same boat I was in.
 I have been running pjsip internally for several 6 plus months and still 
hit config snags occasionally, but our config process is almost complete 
and is going production by the end of the month.  
  
 Bryant
 


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Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

[Bryant Zimmerman]

From: "Joshua Colp" <jc...@digium.com>
Sent: Monday, October 5, 2015 9:20 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from 
ODBC   
 On 15-10-05 10:15 AM, Bryant Zimmerman wrote:



>
> --
> I am working on a step by step and some internal documentation on pjsip.
> Where would the best place to post this information be. I am willing to
> share what I have it may help others in the same boat I was in.
> I have been running pjsip internally for several 6 plus months and still
> hit config snags occasionally, but our config process is almost complete
> and is going production by the end of the month.

There's a section on the wiki[1] for configuring PJSIP. You can take a
look and see what is missing/could be improved. If you'd like you can
add a comment to an initial page and if your suggestions look good then
Rusty can provide you wiki edit access.

[1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

--
Joshua
  
 That sounds good. I will do this as soon as we know our documentation is 
good.
  
 Thanks
 Bryant
 


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Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

[Bryant Zimmerman]

I have a pjsip install that is not pulling it's realtime registrations from 
an ODBC database.
 When I have them directly pull from a MySQL database everything seems to 
work.
  
 I am having trouble finding a requirements document for the pjsip realtime 
registrations.
 Is there some kind of special config that has to be used to trigger the 
connection for realtime registrations over ODBC?
  
 My realtime connections to aors, auths,contacts, and endpoints via ODBC 
are working as expected.
  
 Any ideas are appreciated.
  
 Asterisk v 13.5.0
  
 Registrations line from sorcery.conf
 registrations=realtime,px1_ps_registrations
 
 Line for database from extconfig.conf
 px1_ps_registrations => odbc,pjsipRealtime
 
 pjsip show registration
 show no return records - even though there are records in the database 
table.
  
  
  
 Thanks

Bryant

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Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

[Bryant Zimmerman]

On 15-10-04 09:54 AM, Bryant Zimmerman wrote:
> I have a pjsip install that is not pulling it's realtime registrations
> from an ODBC database.
> When I have them directly pull from a MySQL database everything seems to
> work.
> I am having trouble finding a requirements document for the pjsip
> realtime registrations.
> Is there some kind of special config that has to be used to trigger the
> connection for realtime registrations over ODBC?
> My realtime connections to aors, auths,contacts, and endpoints via ODBC
> are working as expected.
> Any ideas are appreciated.
> Asterisk v 13.5.0
> Registrations line from sorcery.conf
> /registrations=realtime,px1_ps_registrations/

This should be:
registration=realtime,px1_ps_registrations

> Line for database from extconfig.conf
> /px1_ps_registrations => odbc,pjsipRealtime/
> pjsip show registration
> show no return records - even though there are records in the database
> table.
> Thanks
>
> Bryant
>
>

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

--
  
 Joshua
  
 Thanks for your reply. It thought the same thing, but when I change the 
line in the corcery.conf to:
 registration=realtime,px1_ps_registrations
  
 Asterisk crashes and won't start. Here is what the log loop.
  
 [Oct  4 16:04:18] WARNING[64823] config_options.c: Cannot update type 
'registration' in module 'res_pjsip' because it has no existing 
documentation!
  
 If I switch to "registrations=realtime,px1_ps_registrations" the error 
stops, but I get now calls from the px1_ps_registrations table from the 
database.
 What could be missing?
  
 Thanks
 Bryant
 

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Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

[Bryant Zimmerman]

 


 From: "Joshua Colp" <jc...@digium.com>
Sent: Sunday, October 4, 2015 12:12 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from 
ODBC   
 On 15-10-04 01:09 PM, Bryant Zimmerman wrote:
> --
> Joshua
> Thanks for your reply. It thought the same thing, but when I change the
> line in the corcery.conf to:
> registration=realtime,px1_ps_registrations
> Asterisk crashes and won't start. Here is what the log loop.
> [Oct 4 16:04:18] WARNING[64823] config_options.c: Cannot update type
> 'registration' in module 'res_pjsip' because it has no existing
> documentation!
> If I switch to "registrations=realtime,px1_ps_registrations" the error
> stops, but I get now calls from the px1_ps_registrations table from the
> database.
> What could be missing?

Outbound registrations are done in res_pjsip_outbound_registration, as a
result the registration= needs to be in a section for that module instead.

--
  
 Joshua
  
 That seems to have been the issue.  Is there a documentation page out 
there that highlights which options goes under which modules.
 I have not run across this yet and am wondering if I am going to bump into 
any more that need to be pushed under their own config context.
  
 Also is there a trick to what should be used in the client_uri field to 
make the connection?
 I am trying to connect to a sip vendor and I am trying to use 
sipacco...@venderhostname.vendordomain.net?
 Now that the registration table is coming up it is stating I have an 
invalid client URI.. I put the same thing in for a text based registration 
and it worked.

Thank you for your assistance.
  
 Bryant


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Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

[Bryant Zimmerman]

 


 From: "Joshua Colp" <jc...@digium.com>
Sent: Sunday, October 4, 2015 12:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from 
ODBC   
 On 15-10-04 01:42 PM, Bryant Zimmerman wrote:
> 
> *From*: "Joshua Colp" <jc...@digium.com>
> *Sent*: Sunday, October 4, 2015 12:12 PM
> *To*: asterisk-users@lists.digium.com
> *Subject*: Re: [asterisk-users] pjsip realtime registrations not pulling
> from ODBC
> On 15-10-04 01:09 PM, Bryant Zimmerman wrote:
>> --
>> Joshua
>> Thanks for your reply. It thought the same thing, but when I change the
>> line in the corcery.conf to:
>> registration=realtime,px1_ps_registrations
>> Asterisk crashes and won't start. Here is what the log loop.
>> [Oct 4 16:04:18] WARNING[64823] config_options.c: Cannot update type
>> 'registration' in module 'res_pjsip' because it has no existing
>> documentation!
>> If I switch to "registrations=realtime,px1_ps_registrations" the error
>> stops, but I get now calls from the px1_ps_registrations table from the
>> database.
>> What could be missing?
>
> Outbound registrations are done in res_pjsip_outbound_registration, as a
> result the registration= needs to be in a section for that module 
instead.
>
> --
> Joshua
> That seems to have been the issue. Is there a documentation page out
> there that highlights which options goes under which modules.
> I have not run across this yet and am wondering if I am going to bump
> into any more that need to be pushed under their own config context.

I don't think there's a page that describes it unfortunately.

> Also is there a trick to what should be used in the client_uri field to
> make the connection?
> I am trying to connect to a sip vendor and I am trying to use
> sipacco...@venderhostname.vendordomain.net?
> Now that the registration table is coming up it is stating I have an
> invalid client URI.. I put the same thing in for a text based
> registration and it worked.

You need to put sip: in front to make it a valid SIP URI.

--
Joshua
  
 Much thanks for all your direction today. I am sending good karma your 
way.
 Thank You Thank You
  
 Bryant


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[asterisk-users] Asterisk Qualify to pjsip

[Bryant Zimmerman]

I am running a pjsip test between two servers one running pjsip and one running 
chan_sip

 The chan_sip side is sending requests based on qualify=yes.

 The pjsip side is showing notices.. Exp
 ?[Oct  4 18:09:02] NOTICE[5982]: res_pjsip/pjsip_distributor.c:347 
log_unidentified_request: Request from '"asterisk" 
' failed for 'xxx.xxx.xxx.xxx:5060' (callid: 
2d65aa1a2f162b075486d21b661c9...@xxx.xxx.xxx.xxx:5060) - No matching endpoint 
found

 Why is asterisk chan_sip sending it's qualify requests as from 
sip:aster...@xxx.xxx.xxx.xxx?
 Is there a way to get the chan_sip side to send it's qualify requests from the 
actual registered peers so these messages would not look like requests being 
sent from an unauthorized endpoint?

 Is there an easy way to supress these notices if there is no way to shut them 
down.

 Thanks

 Bryant

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Re: [asterisk-users] Asterisk AMI events filtering

[Bryant Zimmerman]

Sam
  
 Based on my experience you need to write a middle tier that has what you 
want exposed to the users.. AMI was not really designed to offer direct 
multi-tenant access. That is for your middle tier to handle.
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: "Sam Basan" <sba...@bluebe.net>
Sent: Thursday, September 17, 2015 7:21 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] Asterisk AMI events filtering   

   

Hi folks,  

   

I have one server with multiple companies (multi-tenant).  

>From AMI I get all events of all extensions so any one that connect can see 
other extensions, from different company (context).  

How can I limit specific user to get just specific context?  

   

Sam  

  


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Re: [asterisk-users] Asterisk SMS

[Bryant Zimmerman]

On Friday 10 Jul 2015, Thyda ENG wrote:
 Dear Sir,

 Does the asterisk support SMS feature ?
 If it does how can we config that ?
 I am waiting for your reply,Thank.
 
 Thyda
  
 Yes asterisk supports SMS on both cell card and sip trunk.
  
 Checkout this link as a starting point. 
  
 http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  
   

 

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[asterisk-users] Distributed Device States - Best Option

[Bryant Zimmerman]

We have used AIS for disturbed Device State in the past, BLF and MWI, We 
are in the process of an update on one of our clustered systems, We are 
looking at XMPP and I found a few discussions on a Corosync with has 
OpenAIS built in. 
  
 My question is which should I be looking at to replace my current AIS 
option I currently have.  XMPP or Corosync? 
  
 It looks like the Corosync is just the AIS option more nicely packaged. Is 
XMPP a better solution as I grow my network? Are there down sides to XMPP 
that AIS/Corosync does better... 
  
 Can anyone recommend where I can find some up to date documentation that 
would cover up through Asterisk 13 on Distributed Device State. 
  
 Thanks for any feed back. 
  
 Bryant

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Re: [asterisk-users] adding area code

[Bryant Zimmerman]

 
 Thanks for your reply,

[globals]
AREACODE=381

[outbound]
exten = _9XX,1,Dial(SIP/SIP-Provider/1${AREACODE}${EXTEN-1},80)

did not work for me, any ideas?

Thanks,
 
  On 04/27/2015 01:59 PM, Phil Reynolds wrote:

On 27 April 2015 21:32:42 BST, Motty Cruz motty.c...@gmail.com wrote:
Hello,

I would like to add area code if clients dial 7 digits, it that
possible? currently clients dial prefix 9 plus local number, however my

SIP provider is requiring to dial 10 digits. is it possible to add area

code?r

Quite simple - you need to match on NXXX and when passing it to the SIP 
provider, present ${AREACODE}${EXTEN}, having first defined AREACODE in 
[globals].

--
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Re: [asterisk-users] adding area code

[Bryant Zimmerman]

Motty
  
 Yes
  
 From your dial plan accept 9 + 7 digits then concat your dialed number 
together with your areacode.
  
 This s a brief example.
  
 exten = _9XXX,1,Set(l_HomeAreaCode=555)
 exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) ;; This 
line should combine your area code and the last 7 digits of your dialed 
phone number
 exten = _9XXX,n,Dial(SIP/${dialnumber},35)
  
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: Motty Cruz motty.c...@gmail.com
Sent: Monday, April 27, 2015 4:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] adding area code   
Hello,

I would like to add area code if clients dial 7 digits, it that
possible? currently clients dial prefix 9 plus local number, however my
SIP provider is requiring to dial 10 digits. is it possible to add area
code?

Thanks,
Motty

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Re: [asterisk-users] FXO advice

[Bryant Zimmerman]

Alejandro
  
 All of the Grandstream devices can be remote provisioned if you know what 
you are doing.
  
 Bryant 
  


 From: Alejandro cdgr...@gmail.com
Sent: Wednesday, April 15, 2015 4:17 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] FXO advice   
 Hi All,   
 I'll like to know if exist some Basic FXO that support some type of 
automatic provisioning of configuration.
  
 Our idea is avoid the users need to go into WebPage and setup our SIP 
gateway.
  
 Some advice or recommendation?
  
 Thanks
 Alejandro


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Re: [asterisk-users] Grandstream GXP2140

[Bryant Zimmerman]

We use a lot of GXP21x phones. We have had issues with the GXP-2140 when 
using the side car as BLF's. The device becomes sluggish after about 45 
days of operation a reboot solves the issues. This has been reported but 
not resolved as of yet.. If you are not using the side car the issue does 
not seem to pop up.. We just put the latest firmware on the phone but it 
does not say it fixes the issue so we are not expecting it to..
  
 If you don't need the side car they are a good phone.
  
 Thanks

Bryant 


 From: dsi...@hcmr.gr
Sent: Wednesday, April 15, 2015 3:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Grandstream GXP2140   
I'm working with GXP2130.
About 12 phone on gigabit with PC after phone.
With Vlans on CISCO switch is stable and not so difficult.
This configuration running without problems since July 2013.

Quoting jg webaccounts...@jgoettgens.de:

 I have a customer looking to deploy about 20 Grandstream GXP2140
 phones. Normally they would deploy Yealink brand phones but they
 want a phone with gigabit pass through and the Yealinks with
 gigabit are too expensive for their budget.


 Does anyone on the list have experience with the GXP2140? Is it a
 reliable phone? Does anyone have recommendations for other phones
 with gigabit pass through?


 I'd be generally careful with the second ethernet connection. One
 should look at the chipset of the phone. I had pretty bad
 experiences with somewhat older TI based phones, regardless of the
 manufacturer. The problems became apparent in mixed environments,
 where some connections were gigabit and others not. It can be a
 nightmare, if you have to offer support.

 The best bet is to buy one, and check the performance of the
 connections. I use some GrandStream products myself and the product
 quality is now much better compared to a couple of years ago.

 jg

 --
 This message has been scanned for viruses and
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 believed to be clean.



D. Sidirokastritis
NOC HCMR-Crete
tel. 2810-337709


Hellenic Center for Marine Research
This message was sent using IMP, the Internet Messaging Program.

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Re: [asterisk-users] Fidelio protocol and Mitel protocol

[Bryant Zimmerman]

Does anyone know anything about the Fidelio and Mitel protocol for hotel / 
motel?
  
 Are these industry standards or proprietary formats?
  
 Are there open standards for communication with Hotel management 
software's that could be used in conjunction with a custom asterisk 
deployment?
  
 Thanks

Bryant 

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Re: [asterisk-users] switching from SIP to Skype..or not

[Bryant Zimmerman]

Hey all
  
 We have been working with SIP for years. It has the potential to be better 
than Skype. It is really all in the implementation.
 Not all SIP soft clients are equal nor are the networks and computers they 
are running on.
 I will not bash Skype. We have tested it and in most cases choose not to 
use it. It has it's place and is good for the user that meets it's specific 
target demographic.  SIP is a sold communications protocol that can 
communication with codecs of differ audio and video quality levels, and 
supports industry standard software and hardware endpoints.
  
 With SIP you get to choose how good your quality is. With Skype Microsoft 
does. 
  
 It comes down to what do you want to achieve, how much resource do you 
want to put in to it, and are you committed to a bit more work for a lot 
more options and better quality, or do you want a quick and easy solution 
with differing limits. Both solutions have their place.  To me SIP vs Skype 
is like complaining apples and carrots do you want fruit or veggies you get 
to choose.
  
 You can choose to agree or disagree with my statements. I hope they are 
useful to some.
  
 Thanks

Bryant
  


 From: Ron Wheeler rwhee...@artifact-software.com
Sent: Thursday, March 12, 2015 9:40 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] switching from SIP to Skype..or not   
Your characterization may be true but Skype works much better than SIP
when it comes to sound quality.

I have SIP softphone with Asterisk server and Skype on the same
workstation.
Skype just works better over the same network.

Ron

On 12/03/2015 9:26 AM, A J Stiles wrote:
 On Thursday 12 Mar 2015, Thufir wrote:
 I'm testing Asterisk at home, crummy connection. Skype works fine for
 me, but every SIP client, even without using Asterisk, fails to 
connect.
 That's ok.

 Is swapping out SIP for Skype a big deal?
 Stay away from Skype! It is a toxic, proprietary product. The lack of
 interoperability by design is the antithesis of what a telecommunication
 system should be about -- and the extent to which they have gone to 
thwart any
 attempt at interoperability is truly shocking.

 For connecting two Asterisk installations to each other over the 
Internet, IAX
 is better than SIP -- that's what it was designed for.


--
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President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Unstable phone connection

[Bryant Zimmerman]

D'Arcy J.M. Cain 
  
 If the device is registering and then dropping there are several usual 
items.  
 The router may be closing the ports on the device. 
 The router may have a AGL SIP helper that is causing issues. 
  
 Make sure that the device is sending out keep alive packets.
 Shut down any AGL helpers on the router.
 Make sure that the site is not double NATing
  
 Try using a stun server and see if that helps at all.
 Watch you console on your sip serer to see how long the device runs before 
losing connection.
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: D'Arcy J.M. Cain da...@vex.net
Sent: Thursday, March 12, 2015 2:40 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Unstable phone connection   
This is driving me to distraction. I have a switch with multiple
clients who are all working fine except for one and I can't figure out
what makes them different. I have tried every NAT setting in the ATA
(SPA112 ATA with 2 x FXS, 1 x LAN), stun server on and off, different
sip ports, different RTP ports and it still fails. I have left the
location with it working only to have it fail later. He always gets
registered but when a call is sent it doesn't respond so the caller
hears no ring and the phone does not ring.

Yesterday he mentioned that when the phone is working the WiFi slows
down significantly. No idea why or if it is related.

He has a radio station streaming music. I wondered if that might be
interfering. That's why I tried changing the SIP port and the RTP
ports but that didn't seem to help.

It smells like a network problem to me but I am running the same ADSL
device here and other clients are working behind a NAT gateway so I am
at a loss as to what might be wrong. Could it be the streaming?

Cheers.

--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

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Re: [asterisk-users] GXP 1405 and asterisk

[Bryant Zimmerman]

 

SIPAddHeader(Alert-Info:\;info=ring3)  

In the phone config add the value ring3 and select Account # / Call 
Settings / Match Incoming Caller ID (Matching Rule)  

In the first rule place the word ring3 and select your ring tone.  

This will cause the selected ringtone to be used when calls with the info 
value of ring3 is matched  

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: ricky gutierrez xserverli...@gmail.com
Sent: Thursday, March 12, 2015 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Subject: [asterisk-users] GXP 1405 and asterisk   
Hi list, someone has successfully change different ringtone from
dialpan with asterisk with this model Granstream?

for example:

exten = 0,1,Playback(pls-wait-connect-call)
same= n,SIPAddHeader(Alert-Info:;info=ring3)
same= n,Dial(SIP/310SIP/318,30,t)

can not get it to work

any idea o tips?

regardss

--
rickygm

http://gnuforever.homelinux.com

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Re: [asterisk-users] func_odbc 123

[Bryant Zimmerman]

Thurfir
  
 We use a combination of func_odbc calls to drive static dialplan.
 Be aware that there is currently a bug if you use ODBC with MySQL, and 
your primary database is offline
 The system does not properly roll to backup database servers. This often 
causes asterisk to lock up requiring a restart.
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
  


 From: Thufir hawat.thu...@gmail.com
Sent: Tuesday, March 10, 2015 4:15 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] func_odbc 123   
with func_odbc, in the definitive asterisk guide, they were suggesting
the possibility that part, or perhaps all of, the dialplan could be
written as SQL statement!?

First off, that sounds like a good idea to me, but the tone of the
authors was suggesting not so much, but that it was a personal preference.

From a naive perspective, why SQL statements at all? Why not just
database config and data instead?

thanks,

Thufir

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Re: [asterisk-users] New Asterisk build

[Bryant Zimmerman]

John
  
 I will have to get one of these and give this a try. Thanks for sharing.
  
 Thanks

Bryant Zimmerman (Grand Dial Communications, a ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: John Novack SCII jnov...@stromberg-carlson.org
Sent: Friday, March 6, 2015 3:37 PM
To: Ira i...@extrasensory.com, Asterisk Users Mailing List - 
Non-Commercial Discussion asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] New Asterisk build   
Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux 
( instructions at AstLinux.org ) Move your Digium card and your confs , fix 
up any differences from your
older version of Asterisk to the fairly current version 11 currently 
available with AstLinux.
Use the GUI to edit and mage the system, as AstLinux has a somewhat 
different directory structure than you may be familiar with
You should be up and running in a couple of hours, have a low power  20 
watts, fanless flash based system that will just work in a real case.
The Pi is OK for a playtoy and some testing, but I much prefer the HP thin 
clients for a robust installation.
I assume you are not doing any fancy call center or heavy database work. 
For a home or home office it is a really good solution.
AstLinux is also used with other embedded installations on computers with 
multiple Ethernet ports, acting as router and firewall in addition.
I prefer the component solution personally, which makes the HP thin clients 
the way to go.

John Novack

I have built more than 30 of these systems on various HP Thin Clients, used 
off of eBay with no failures

Ira wrote:
 Hello Asterisk,

 Back in 2009 I built a small Intel Atom based computer running
 Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs
 line and six or so SIP numbers. So basically no load. I'm
 feeling like it's time to build another machine. It's probably
 silly, but it's been six years and I can't upgrade the OS
 which is falling behind. I'd likely just put it on a Raspberry
 Pi or something like that, but I need the one POTS line and
 all I have for that at the moment is a Digium card for a PCI
 slot.

 Are there any current thoughts on this?

 -- Ira



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Re: [asterisk-users] New Asterisk build

[Bryant Zimmerman]

Iran
  
 For the kind of loads and low cost you are talking with 2 FXO, 2FXS and 
SIP the Grandstream UMC6102 is low power feature rich and easy to maintain. 
Check it out - 
http://www.grandstream.com/index.php/products/ip-voice-telephony/ip-pbx-solu
tions/ucm61xx 
 If you do choose to use the UMC61xx the grandstream phones auto-provision 
with it well, but it works with any complaint SIP phone.  
  
 If you do want to go with an asterisk home brew. You could use a 
Grandstream GXW4104 (4 FXO) for your POTS line. It is a FXO gateway that 
would register as a SIP endpoint. (You could look at the HT503 which has 
one FXO port, but I find them to be less reliable then the GXW4104). The 
nice thing about using gateways is there are no drivers to load on your 
asterisk build as the gateway is just a SIP endpoint.
  
 I have built asterisk test systems on raspberry pi Rev B and have not been 
happy with their performance even in light loads. The new version 2 B looks 
like it might be better, In ether case the Gateways would be a good way to 
go to connect your lines. Watch your SD card speeds slow cards really gave 
me a lot of issues. Especially when you had someone leaving a voicemail and 
someone else was trying to listing to an IVR prompt, multiple users reading 
and writing at the same time just really have not worked well. We hooked up 
a SSD via USB and put our prompts and voicemail on it and it was a bit 
better still limited to USB2 speeds, but that increased the cost.
  
 The UMC6102 is the best value as buy the time you purchase a gateway, 
system and spend time loading it is hard to beat the price point and you 
can get support on it from Grandstream or a reseller.
  
 (To be open I am a Grandstream reseller, I am offering these recommending 
as they are good options. There are several other low cost asterisk like 
PBX's out there as well, Allo and several others, but I know the GS options 
work)
  
 Good Luck and I hope this info helps.
  
 Thanks

Bryant Zimmerman (Grand Dial Communications, a ZK Tech Inc.)
  
 P.S. Glen's post also offers some good points as well.
  

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Re: [asterisk-users] Investigating international calls fraud

[Bryant Zimmerman]

If you have not done so contact the carrier immediately. Report the fraud.
 Have them disable international on the account until you have your 
security issues addressed.
 Ask them to pull call logs containing Source and destination IP address. 
for the fraud calls.
 If you are sure they came from your systems IP address then verify the 
systems does not have any unauthorized registered endpoints.
 Verify the source and destination of the calls from any internal logs.
  
 If the calls came from your IP then you are likely on the hook for the 
calls. If they came from another IP that you don't own then if you can 
prove you had no access to that IP than there is some hope the carrier may 
work with you.
  
  
 Thanks

Bryant 
  

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Re: [asterisk-users] Return SIP 401 on hangup

[Bryant Zimmerman]

I am having and issue I hope someone can help with..
  
 I have calls that often come in that need to be blocked. We wish to do 
this without answering the call.
 The issue is our carriers have fail over servers and will try sending the 
call from each when we block the call.
  
 If we send a hangup with a Sip 401 they will stop the route advance on 
their end.
  
 The issues is we have been sending a hangup/cause code 21 (Call rejected)
  But they are receiving a 403 Forbidden..
  
 Is there any hangup code that we can send that will reply a 401.. We see 
the 401 on the inbound is converted to a cause code 21 but we do not see 
any cause codes listed to send a 401 out. Please advise as this a becoming 
an issue for us as we have multiple vendors expecting a 401 not a 403
  
 Any assistance is appreciated. 
  
 Thank you.
 Bryant

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Re: [asterisk-users] Confbridge

[Bryant Zimmerman]

I am doing dynamic conference bridges using confbridge in asterisk 11.
 Is there a way to toggle off an on recording of an ongoing conference 
call
 I have figured out how to record a conference if it is turned on when 
someone enters.
  
 Also I have noticed that when setting music_on_hold_class dynamically it 
does not override what is set on the channel.

exten = s,n,Set(CONFBRIDGE(user,music_on_hold_class)=latin) 
 Does anyone have any ideas on how I might fix this as well?
  
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003

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Re: [asterisk-users] Astricom 2014 presentations

[Bryant Zimmerman]

 
   On 10/29/2014 05:50 AM, Bogdan Cristea wrote:
 Hi

 Will the presentations made at Astricom 2014 be made public as recorded 
videos ?

 thanks
 Bogdan

I'll second the request for that, and also ask if the sessions on
Kamailio will be similarly available.

Cheers,

j

That would be awesome if they chose to do this.
  
 Bryant
 


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Re: [asterisk-users] unidata incom ICW-1000G - On asterisk

[Bryant Zimmerman]

I am trying to use an ICW-1000G wireless handset connected to an asterisk 
server remotely
  
 The user is working from an offsite location and it appears that the 
device is not sending out keep-alives or stun.
 The manufacture is not being of assistance at all.  I am wondering if 
anyone has worked with these units or has any ideas of what I could do to 
make them work.
  
 Anyone have a better alternative. The units must support wifi, be able to 
clip on belt and support an external headset (wired is fine) Battery life 
should support all day use in a standard biz env. Work thru a firewall from 
offsite.
  
 The ICW-1000G supports all of these requirements except the Work thru a 
firewall from off-site.
  
 If the MFG can't get a fix or someone does not have any ideas I would 
recommend people stay away from the ICW-1000G if you have to use them 
off-site or for hosted connections.
  
 Thanks
 Bryant

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Re: [asterisk-users] AGI scripts - delay issue.

[Bryant Zimmerman]

Hey All
  
 We have several AGI scripts that access databases. These work well most of 
the time.
  
 The issue we are having is that on rare occasion our script must fail to a 
backup database server.
 When this occurs it may take up to two seconds to do so.  The issue is 
when there is this delay the script loses access to read global channel  
variable values only after the delay.  This is driving me crazy is there 
some kind of  AGI timeout issue or bug that could be causing this.
  
 Thanks

Bryant

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Re: [asterisk-users] Dynamic Parking Lots. Music on Hold Class

[Bryant Zimmerman]

How can we set the music on hold class using the Dynamic Parking lots?
  
 The variables set the PARKINGLOT, PARKINGDYNAMIC, 
PARKINGDYNPOS,PARKINGEXT,PARKINGDYNCONTEXT
  
 I can't find a PARKINGMOH variable. This is becoming a big issue. We are 
using the current release 11. version
  
 We have to be able to set the MOH dynamically I just can't find the 
mechanism. Any ideas?
  
 Thanks

Bryant

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Re: [asterisk-users] AGI script VERBOSE cmd

[Bryant Zimmerman]

I am working on an AGI script and all is going well except I can not get 
any of my VERBOSE commands to display.
  
 Is there any undocumented reason for this to occur? I am able to set 
variables, call other commands ect..
  
 I am sending my verbose command in the following format (VERBOSE Message 
to send 4)
  
 Any ideas what I might be doing incorrect?
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003

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Re: [asterisk-users] AGI script VERBOSE cmd

[Bryant Zimmerman]

Hey all
  
 Please disregard my question. I was looking for the word Verbose to show 
up. I was just being dense.
 There was no real issue it is working just different than what I was 
expecting.
  
 Thanks

Bryant
  


 From: Bryant Zimmerman brya...@zktech.com
Sent: Friday, June 27, 2014 11:25 AM
  I am working on an AGI script and all is going well except I can not get 
any of my VERBOSE commands to display.

   Is there any undocumented reason for this to occur? I am able to set 
variables, call other commands ect..

   I am sending my verbose command in the following format (VERBOSE 
Message to send 4)

   Any ideas what I might be doing incorrect?

   Thanks

Bryant 


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Re: [asterisk-users] Live Recording on the Storage Server?

[Bryant Zimmerman]

A simple way that we use to do the move is to create a cron job that looks for 
a .move file.
 It has the same name as the recorded file. asterisk writes the .move file 
which is just a text file with some stats in it.
 The .move file is written from the dial plan  at the end of the recording.
 In the exten = h we write a .delete file for an abandon call.

 The cron then processes the .move and .delete files at a given interval. We 
actually write special instructions into our .move files that the cron parses 
and can then act accordingly. So we have a single smart cron job handling moves 
for each type of task. In some cases our .delete files are processed as moves 
to an abandon cache for recovery if a customer did not intend to abandon it.

 The sky's the limit on how complex you want to make it, but in the long run it 
is fairly simple and it just works.

 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003



 From: Chris Bagnall aster...@lists.minotaur.cc
Sent: Thursday, April 17, 2014 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Live Recording on the Storage Server?
On 17 Apr 2014, at 16:14, Paul Belanger paul.belan...@polybeacon.com wrote:
 hi. I would not do that due to network issues.
 My approach is to record everything locally and every hour or so to move
 everything to a storage.
 +1 save yourself the headache and do this.

I'll add another +1 to this. I've never been able to get multi-channel 
recording (even 3 or 4 channels) working reliably over an NFS link to another 
server. I suspect, with some tweaking of nfs options it might be possible, but 
if it ain't broke.

Just a cautionary note if you do use a cron job to move recordings to a storage 
device at regular intervals: make sure you use lsof or similar to check the 
recordings aren't actually open by asterisk at the time, otherwise interesting 
things will happen.

Kind regards,

Chris
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Re: [asterisk-users] func_odbc

[Bryant Zimmerman]

Hi All
  
 Anyone know how to do include files with func_odbc.conf?
  
 I now have several pages of functions in my func_odbc.conf and it is 
getting harder to maintain it.
 I would like to break them up into files by category. The standard method 
of using the #include does not seem to work .
  
 Ideas are appreciated.
  
  
 Bryant

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Re: [asterisk-users] Asterisk SIP server on windows

[Bryant Zimmerman]

On Wednesday 04 December 2013, Ruddy Gbaguidi wrote:
 Hi all,
 
 I need to build an application that will be an SIP server program that 
will
 run on Linux and Windows.
 
 The sip server need only some features such as be able to :
 
 -  Register sip endpoints
 
 -  Answer a call and play a local file
 
 -  Make a dial from one channel to another.
 
 
 
 I know asterisk can be stripped to exactly fit my needs. I would like to
 know if there is a way to build it on windows after it has been 
stripped.
 
 Or do I have other alternatives out there ?

Ruddy

  If you can use windows 8.1 Pro 64bit. You can use hyper-v and run a 
virtual linux machine (Or the Free Hyper-V Server 2012 R2), VM Ware also 
works well. Load asterisk on that and you are set. This is how we run it at 
a few very small customers as well as my development machines and it works 
great. Best linux builds for Hyper-V we currently have found to be Ubuntu 
and Suse. 

As both a windows and linux guy I have to concur that loading Asterisk on 
windows directly is like putting a V8 on a moped. You may get there, but it 
won't be pretty; It's a lot of work, and it would be hell to maintain. (We 
do not trust it for production applications) 

In all seriousness I have a Asterisk build running on windows and it is 
stable but it is a lot of work to get it there and since it is not 
maintained by the community it is a full task to keep it up to date. We use 
it for in process testing of code that we develop with MS visual studio. If 
not for that I would not bother with Asterisk on Windows there would be no 
value in it. Especially since the current version of Asterisk now works so 
well in virtual environments. 

Good luck
Bryant

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Re: [asterisk-users] Asterisk SIP server on windows

[Bryant Zimmerman]

Ruddy

If you can target windows 8.0 pro or 8.1 pro you can ship a Hyper-V image. 
(This same image would work with Hyper-V on, Hyper-V Server 2012/2012r2 and 
Windows Server 2012/2012r2. 

You would need to write some kind of configuration editor or documentation 
to customize the image to the users network environment.

You could automate the entire process if you were so inclined.
You can fully automate the Hyper-V side of things using PowerShell or 
Dot.Net
The Linux side of things can be automated in a number of ways. We 
personally wrote a windows program that collects information from the user 
and posts it to our databases. The default image then has a script that 
pulls the info down (images uses DHCP to start) and re-writes the asterisk 
configs.

This process is not a small task but if you have the time and budge it can 
work very well. 

Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003


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Re: [asterisk-users] Voicemail greeting playback issues?

[Bryant Zimmerman]

From: Doug Lytle supp...@drdos.info
Sent: Monday, November 25, 2013 6:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Voicemail greeting playback issues?

Bryant Zimmerman wrote:
 Hey all

 I believe I found the bug in Asterisk 11.xxx If someone can help me
 verify it.

Actually,

I wouldn't consider it a bug.  I've know for years that you need to
answer a channel before you play back audio or strange things can and
will happen.

Doug

-- 
Doug

The real issue here is that issuing an Answer() just before does not seem 
to solve the problem. To work around the issue I have to either put a 
Wait(1) or Dial() some extensions first. It is presenting like if you drop 
into the Voicemail() command too fast during call setup that you have 
issues. This did not occur in 1.8.x. I would be ok if just issuing an 
Answer() would resolve it as this would be normal, but having to slow down 
the dial plan seems off. 

Thanks
Bryant

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[asterisk-users] Voicemail greeting playback issues?

[Bryant Zimmerman]

Hey all

I have been beating on this all weekend long.
Any feed back would be appreciated. 

We stood up a 11.6 system. We tested everything we could think of. 
We moved over to it and all seemed to be working good than a customer told 
us that they were not hearing our vociemail greetings. 
When we call into the system and it drops to voicemail we just get a beep 
no greeting played.  We checked and the greeting files are there and play 
back from the voicemail ivr.  If no greeting is there it just plays The 
Pers.. beep

Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue goes 
away.

Any Ideas?

Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003
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Re: [asterisk-users] Voicemail greeting playback issues?

[Bryant Zimmerman]

From: Doug Lytle supp...@drdos.info
Sent: Monday, November 25, 2013 2:01 PM
To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Voicemail greeting playback issues?

 Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue 
goes away.

I don't see this under 11.5.1

Doug

---

Doug

Thank you for your response. It is good to hear that you are not having the 
issue. 
It gives me hope that there is a way to resolve this quickly. 

Do  you have an thing special around your voicemail configuration? We  
started with the 11.xx sample config and mapped our settings from  1.8.x.   
Both our 11.xx and 1.8.x systems are running on the same  virtual server. 
Both are reading and writing audio and vm files to and  from the local 
storage.  I forced off g729 to ensure that it was not  causing the issues.

Do you know of any way to force a higher level of debugging to see why the 
voicemail application would be having an issue?

Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003


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Re: [asterisk-users] Voicemail greeting playback issues?

[Bryant Zimmerman]

From: Bryant Zimmerman brya...@zktech.com
Sent: Monday, November 25, 2013 2:49 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Voicemail greeting playback issues?


From: Doug Lytle supp...@drdos.info
Sent: Monday, November 25, 2013 2:01 PM
To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Voicemail greeting playback issues?

 Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue 
goes away.

I don't see this under 11.5.1

Doug

---

Doug

Thank you for your response. It is good to hear that you are not having the 
issue. 
It gives me hope that there is a way to resolve this quickly. 

Do  you have an thing special around your voicemail configuration? We  
started with the 11.xx sample config and mapped our settings from  1.8.x.   
Both our 11.xx and 1.8.x systems are running on the same  virtual server. 
Both are reading and writing audio and vm files to and  from the local 
storage.  I forced off g729 to ensure that it was not  causing the issues.

Do you know of any way to force a higher level of debugging to see why the 
voicemail application would be having an issue?

Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003



Hey all

I believe I found the bug in Asterisk 11.xxx If someone can help me verify 
it.

My voice mail test scripts do not answer or wait they just drop you into 
the voicemail box. 

It appears that something with Asterisk 11.xx is causing the voicemail() 
command to drop in and ether not play or mess up the prompts. If you have 
not given it at least one second in the channel before passing it to the 
voicemail() command.
If you throw a wiat(1) just before the voicemail() command the prompts play 
correctly. So if you have rung extensions using dial() before going to 
voicemail that appears to be enough time. 

If you place an inbound call directly to voicemail() with no pause then you 
have an issue. 

Example Broken:
exten = _9XXX,1,Set(l_VMExt=${EXTEN:1})
exten = _9XXX,n,MailboxExists(${l_VMExt}@${siteVMContext})
exten = _9XXX,n,GotoIf($[${VMBOXEXISTSSTATUS}=FAILED]?doHangup)
exten = _9XXX,n,Voicemail(${l_VMExt}@${siteVMContext},u)
exten = _9XXX,n(doHangup),NoOp(Issue 9XXX Hangup)
exten = _9XXX,n,Hangup()

Example Works:
exten = _9XXX,1,Set(l_VMExt=${EXTEN:1})
exten = _9XXX,n,MailboxExists(${l_VMExt}@${siteVMContext})
exten = _9XXX,n,GotoIf($[${VMBOXEXISTSSTATUS}=FAILED]?doHangup)
exten = _9XXX,n,Wait(1)
exten = _9XXX,n,Voicemail(${l_VMExt}@${siteVMContext},u)
exten = _9XXX,n(doHangup),NoOp(Issue 9XXX Hangup)
exten = _9XXX,n,Hangup()

The code that is broken with Asterisk 11.xx worked in Asterisk 1.8.x
Can anyone confirm this?

Thanks
Bryant Zimmerman()

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Re: [asterisk-users] 11.6 voicemail message cropped off?

[Bryant Zimmerman]

Hey all

I am running 11.6 and when a caller is sent to vociemail the greeting is 
cropped off and the beep occurs quickly.
Incoming calls are g729 and this occurs where it is using the default 
greeting or a user provided greeting.

I really want to go production with this are there any ideas what could 
cause an issue like this we have never seen it in 1.4 - 1.8

Bryant
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Re: [asterisk-users] 11.6 voicemail message cropped off?

[Bryant Zimmerman]

Update

When no greeting is recorded the default you have reached ext # greeting is 
cropped. When there is a greeting it is just ignored and not played at all. 


Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003


From: Bryant Zimmerman brya...@zktech.com
Sent: Saturday, November 23, 2013 8:32 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 11.6 voicemail message cropped off?

Hey all

I am running 11.6 and when a caller is sent to vociemail the greeting is 
cropped off and the beep occurs quickly.
Incoming calls are g729 and this occurs where it is using the default 
greeting or a user provided greeting.

I really want to go production with this are there any ideas what could 
cause an issue like this we have never seen it in 1.4 - 1.8

Bryant

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Re: [asterisk-users] Movistar sip Mexico

[Bryant Zimmerman]

Can you funnel them through a specific inbound dial context. Then force a 
re-invite to g729?

Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003


From: Damian Gonzalez dgonza...@denwaip.com
Sent: Thursday, November 21, 2013 8:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Movistar sip Mexico

Any posible solution?

On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner k...@kriskinc.com 
wrote:
 It is possible that Asterisk requires an rtpmap even for static payload 
types (I'm not sure about this).  The INVITE from your provider omits 
rtpmap for payload type 18 (G729) which is perfectly valid.

On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez dgonza...@denwaip.com 
wrote:

Hello,

Thanks for the quickly response. I have only G729 in the peer but I have 
t38pt_udptl= yes  

If I put t38pt_udptl=no , asterisk reject the call with 488 code. 

The problem is that Movistar send T38 codec in all calls and I need ignore 
only if in the SDP I have G729 and T38 (18 and 101), but if I have only T38 
I have to negociate a fax call.

Thanks.

On Wed, Nov 20, 2013 at 4:46 PM, Alyed al...@vivoxie.com wrote:

Think you only need to make sure you have in your sip.conf file these 
configs:

   [your-device-name]
.
.
disallow=all
allow=g729

.
  .

Alyed

2013/11/20 Damian Gonzalez dgonza...@denwaip.com

Hello,

I have a problem with movistar in Mexico with a sip calls. Movistar send to 
me T38 and G729 in the INVITE and they say that I have to ignore T38 and 
use G729 in the voice call.  

When a fax call is made Movistar send only T38 in the INVITE. 

Invite example:

v=0
o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2  
s=sip call
c=IN IP4 192.168.1.2
t=0 0
m=audio 6370 RTP/AVP 18 101
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20  
m=image 6372 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy  

How can I  ignore T38 and use only G729 for this call?.

Thanks for your help.

Damian  

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Re: [asterisk-users] is g729 codec free? or under license???

[Bryant Zimmerman]

When calling between two g729 client endpoints you do not need any licenses as 
long as no audio prompts or voicemail is evolved. Also as a sip trunk provider 
we offer g729 as a source and destination codec this allows you to make calls 
in and out using g729 (most carrier grade providers offer this option)



You really only need to buy the number of g729 licenses that you will need for 
callers that require simultaneous transcoding. This is when a callers stream in 
or out will need to be converted to another codec format.  This occurs when 
callers are jumping from say g729 to g711 or g729 to g722, g729 to gsm. If you 
plan things right and make sure any audio prompts your system is using are 
recorded in g729 as well as g711 and g722 you will reduce the number of g729 
license considerable.



Process that use a lot of g729 transcodes. ConfBridge uses g722 so all g729 has 
to be converted to and from g722 so 10 g729 callers to a confbridge would 
likely require 10 codecs (**See confbridge trick below). If you have prompts 
that are not pre-encoded in g729 those would use a transcoder license while 
playing.  Voicemail would require a license as g729 has to be transcoded to one 
of the storage formats.



The real number is based on how you are using your system.



ConfBridge Trick - Have seen this used for voicemail as well, Make sure you 
test when using this method.

  If you can live with using higher bandwidth to the asterisk switch when using 
confbridges (endpoints also have to support in call reinvites correctly) you 
can force endpoints to re-invite to g722 before dropping into the conference 
bridge. This has the upside of not needing to transcode on the server thus 
improving performance and reducing g729 license requirements. This comes at the 
cost of needing higher bandwidth between the client endpoints and the phone.  
Figure about double the bandwidth when using this method. It may or may not be 
worth it to you depending on your scenario.



Please let us know if this information helps you.



Thanks

Bryant Zimmerman

Sr. Systems Architect
Grand Dial Communications , A ZK Tech Inc. Company
616-299-5607 (mobile)
616-855-1030 Ext. 2003 (office)


From: Don Kelly d...@donkelly.biz
Sent: Wednesday, October 2, 2013 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] is g729 codec free? or under license???

In your scenario, all the calls are from endpoints on 181 to endpoints on 183. 
If the endpoint devices are similar, it seems to me that there should be no 
need to transcode-you can use a codec common to the endpoints. 729 would not be 
required.

--Don

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of s m
Sent: Wednesday, October 02, 2013 2:34 AM
To: Dominik George
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] is g729 codec free? or under license???



thank you Dominik you help me a lot.

 and the last question is how many license key should i buy? i read that 
license for g729 is per-channel but i don't understand what channel exactly 
means here. this is my scenario :

10endpointspbx181...pbx182...pbx183...10endpoints

pbx181 and pbx183 has 10 endpoints connected to them. the call between these 
endpoints are established by pbx182. if i want to buy a license for pbx182, how 
many license key do i need? just one because i have just one connection on it?  
or two, because two trunks is defined on it? or as many as endpoints which are 
connected to each other via pbx182?

please help me to clarify channel concept in my mind.

thanks in advance

SAM



On Tue, Oct 1, 2013 at 11:34 AM, Dominik George n...@naturalnet.de wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512

Hi,

about g729, you mean if it get free g729 and all my systems (PBXs and
routers) use g729 codec for setting a call, call is set without any
problem?

Yes, if all systems use g729 directly, you are ready to go.

- -nik

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Re: [asterisk-users] Conf Bridge

[Bryant Zimmerman]

Hey all. 

RE: Conf Bridge.

I am looking into a project that would need 8 to 10 thousand parties in a 
single conference. 
Most would be on mute but 5 to 6 would be presenters. 

Is the new conf bridge solid enough to handle this kind of load?
Any ideas on hardware projections?

If not 8 to 10 thousand how many would be realistic?

If not asterisk any other suggestions.

Thanks for any input. 

zktech

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