Re: [asterisk-users] PJSIP logging fails

2017-04-13 Thread Trey Hilyard 
On Wed, Apr 12, 2017, 4:14 PM Sree Harsha Totakura 
wrote:

> Did you try setting the debug verbosity to a number > 3?
>
> Alternatively, if you want to see a register packet, try running
> wireshark on the server and capture the request packets.
>
> Sree
> On 04/12/2017 08:55 PM, Saint Michael wrote:
> > I am trying to log my SIP registration attempts.
> > PJSIP is in logger mode, and I can see INVITES comingh, my SIP Register
> > does not show, especially the packet I send.
>

Are you logging everything or only a specific host/peer? If you are running
it for a specific host, maybe the absence of the message points to your
problem and the REGISTER is going to the wrong place.

> The only thing shown is:
> > res_pjsip_outbound_registration.c: No response received from 'snet'
> on
> > registration attempt to 'sip:7866314772-xnet', retrying in '60'
> > How do I see my own packet?
>

I confirmed I see REGISTERs in the logger, so I don't know any other reason
you aren't seeing it. What version are you on?

>
> >
> >
>
>
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[asterisk-users] Adding Subscribe Handlers in PJSIP

2017-03-01 Thread Trey Hilyard 
Is there any "easy" way to add a custom subscribe handler? I have a set of
users with Polycom phones that attempt to Events that Asterisk/PJSIP
doesn't recognize, "call-info" and "as-feature-event". It just generates a
warning, but it got me wondering if I could add my own handlers for those
that didn't actually do anything but simply responded with a 200 OK.

Yes, I can probably stop the phones from subscribing, but this is more
academic at this point. I assume there are things that I could do if I
wanted to make changes and recompile, but is there an easier way to add a
handler? I am a little confused about whether there is a subscribe handler
in front of res_pjsip_pubsub, or if that is the first place that a
SUBSCRIBE could get caught.

-Trey
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Re: [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP

2016-05-27 Thread Trey Hilyard 
If you are using PJSIP, you should be able to define a different transport
for each source IP that you want to use and simply tie the endpoint to your
provider to the appropriate transport. Obviously, you'd need to define the
IP on the interface as well. You could use the same interface with multiple
IPs or, if you need to separate the traffic over unique links (not likely,
I imagine), you could always tie them to different interfaces if needed.

On Fri, May 27, 2016 at 1:09 AM Attila Megyeri 
wrote:

>
> If I had just two such users, I would be fine.
> But having 10-20 or even more? It is not a nice, scalable solution...
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand
> Sent: Thursday, May 26, 2016 11:14 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Sending Calls via SIP trunk from several
> different IP addresses from same Asterisk Machine, to the same destination
> IP
>
> How about running a second asterisk instance on the same box with
> different IP/Port combo
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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

2016-03-20 Thread Trey Hilyard 
On Mar 18, 2016 8:27 PM, "Steve Edwards" <asterisk@sedwards.com> wrote:
>>
>> On Fri, 18 Mar 2016, Trey Hilyard wrote:
>>
>>> I thought this would be as easy as
>>> exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})
>
>
> How about something like:
>
> [parse-lrn]
> exten = _x.,1,  verbose(1,[${EXTEN}@${CONTEXT}])
> same = n,   set(DID=${CUT(EXTEN,\;,1)})
> same = n,   set(LRN=${CUT(EXTEN,\;,2):3:12})
> same = n,   execif($["${LRN:0:1}" =
"+"]?set(LRN=${LRN:1}))
> same = n,   execif($["${LRN:0:1}" =
"1"]?set(LRN=${LRN:1}))
> same = n,   goto(${LRN},${DID},1)
> same = n,   hangup()

That's a good one. One thing it doesn't do is actually validate that the
LRN is mine, but that shouldn't be tough to add now the the LRN is in its
own variable. Thanks for the help!

>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
>
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[asterisk-users] Incoming INVITE with Portability Info and LRN

2016-03-19 Thread Trey Hilyard 
I am trying to set up my Asterisk server so that it will recognize an
incoming call to the Asterisk's own Location Routing Number (LRN),
validating the "rn" in the INVITE and then using the Called Number from the
INVITE as the extension in the dialplan.

The INVITE R-URI looks like:
INVITE 
sip:+19135041291;rn=+1913663;npdi@12.4.240.200:5060;user=phone;transport=udp
SIP/2.0

The +1913663000 is the LRN of the Asterisk box, so I would want to have the
dialplan validate that the "rn" is that number. The +19136631291 is the
extension within the system that they are trying to reach, that extension
will vary, and will have an exten defined in the dialplan.

I assume that this is just going to require that I do some matching and
substring-type variable replacement to hit a context with just the Called
Number part of the request, but I wondered if anyone had a working example
of this before I started putting too much effort into it.

As a PBX, Asterisk doesn't have to worry about portability, but I am using
it to simulate a full-blown Class 5 switch, so I have to have an LRN
assigned to it to allow users to port to that switch.

-Trey
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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

2016-03-19 Thread Trey Hilyard 
On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <ad...@tootai.net>
wrote:

> Le 18/03/2016 16:20, Trey Hilyard a écrit :
> > I am trying to set up my Asterisk server so that it will recognize an
> > incoming call to the Asterisk's own Location Routing Number (LRN),
> > validating the "rn" in the INVITE and then using the Called Number from
> > the INVITE as the extension in the dialplan.
> >
> > The INVITE R-URI looks like:
> > INVITE
> > sip:+19135041291;rn=+1913663;npdi@12.4.240.200
> :5060;user=phone;transport=udp
> > SIP/2.0
> >
> > The +1913663000 is the LRN of the Asterisk box, so I would want to have
> > the dialplan validate that the "rn" is that number. The +19136631291 is
> > the extension within the system that they are trying to reach, that
> > extension will vary, and will have an exten defined in the dialplan.
> >
> > I assume that this is just going to require that I do some matching and
> > substring-type variable replacement to hit a context with just the
> > Called Number part of the request, but I wondered if anyone had a
> > working example of this before I started putting too much effort into it.
>
> Use the SIP_HEADER function
>
> http://www.voip-info.org/wiki/view/Asterisk+func+sip_header


I am not sure that this is needed here. The Request URI has all of the
values that I need. I agree that I might need to CUT part of the R-URI, but
I don't need access to any other header to find the info I need.

When the call arrives at the Asterisk right now, this is the exten/context
that it is hitting, so it already has the info I need:
Executing [9135041291;rn=+1913663;npdi@from_pstn:1]

As far as I can tell, I think that I just need to figure out how to make an
extension entry that matches on the "rn=+1913663\;npdi" and then moves
to another context (or same one) with ${EXTEN,0,10}.

I just can't get that first extension to match on the RN value.



>
>
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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

2016-03-19 Thread Trey Hilyard 
I thought this would be as easy as

exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})

But it appears that the pattern match doesn't work once I get to the "r" in
"rn". I am assuming that the pattern match doesn't like dealing with
characters without taking the entire URI.

I am working on a plan using a lot more CUTs than I think I should need,
but we'll see if it works.

On Fri, Mar 18, 2016 at 10:58 AM Trey Hilyard <kct...@gmail.com> wrote:

> On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <ad...@tootai.net>
> wrote:
>
>> Le 18/03/2016 16:20, Trey Hilyard a écrit :
>> > I am trying to set up my Asterisk server so that it will recognize an
>> > incoming call to the Asterisk's own Location Routing Number (LRN),
>> > validating the "rn" in the INVITE and then using the Called Number from
>> > the INVITE as the extension in the dialplan.
>> >
>> > The INVITE R-URI looks like:
>> > INVITE
>> > sip:+19135041291;rn=+1913663;npdi@12.4.240.200
>> :5060;user=phone;transport=udp
>> > SIP/2.0
>> >
>> > The +1913663000 is the LRN of the Asterisk box, so I would want to have
>> > the dialplan validate that the "rn" is that number. The +19136631291 is
>> > the extension within the system that they are trying to reach, that
>> > extension will vary, and will have an exten defined in the dialplan.
>> >
>> > I assume that this is just going to require that I do some matching and
>> > substring-type variable replacement to hit a context with just the
>> > Called Number part of the request, but I wondered if anyone had a
>> > working example of this before I started putting too much effort into
>> it.
>>
>> Use the SIP_HEADER function
>>
>> http://www.voip-info.org/wiki/view/Asterisk+func+sip_header
>
>
> I am not sure that this is needed here. The Request URI has all of the
> values that I need. I agree that I might need to CUT part of the R-URI, but
> I don't need access to any other header to find the info I need.
>
> When the call arrives at the Asterisk right now, this is the exten/context
> that it is hitting, so it already has the info I need:
> Executing [9135041291;rn=+1913663;npdi@from_pstn:1]
>
> As far as I can tell, I think that I just need to figure out how to make
> an extension entry that matches on the "rn=+1913663\;npdi" and then
> moves to another context (or same one) with ${EXTEN,0,10}.
>
> I just can't get that first extension to match on the RN value.
>
>
>
>>
>>
>> --
>> Daniel
>>
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Re: [asterisk-users] SIP URI set 'telephone-context='

2016-02-17 Thread Trey Hilyard 
Agree. All you have to do is:

Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10\;user=phone)

I am actually surprised that the dialplan reload would work without it...

On Wed, Feb 17, 2016 at 5:51 AM A J Stiles 
wrote:

> On Wednesday 17 Feb 2016, imperium broadcast wrote:
> > I kinda have it working with chan_sip.
> >
> > Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10;user=phone)
> > But it doesn't include the user=phone at the end when dialling out.
> >
> > "To: ".
> >
> > even adding
> > usereqphone=yes
> > to the sip.conf doesn't add the user=phone to the end unless I remove the
> > the sip uri stuff out of the dial string.
> >
> > Ideally I would like it to look like this
> > INVITE sip:118099;phone-context=+44@10.10.10.10:5060;user=phone
> > Or
> > INVITE sip: 118099@10.10.10.10:5060; user=phone; phone-context=+44
> >
> > It doesn't matter which way I do it I can only include one extra
> parameter
> > and not the two (user=phone;phone-context) as Asterisk ignores the second
> > one.
>
> That's because in the Asterisk dialplan, a semicolon is used to denote a
> comment  (on account of the comment mark being a valid DTMF digit).  So you
> will have to insert a backslash before the semicolon before user=phone .
>
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Re: [asterisk-users] SIP URI set 'telephone-context='

2016-02-16 Thread Trey Hilyard 
Are you using res_pjsip or chan_sip?

For PJSIP, it's as easy as passing the parameters to the Dial. For example:
Dial(PJSIP/${ARG1}\;phone-context=mydomain.com@pjsippeer,60)

I am pretty sure it was easy in chan_sip, too. If you are using chan_sip,
I'll try and find an example.

On Tue, Feb 16, 2016 at 11:03 AM imperium broadcast <
imperium.broadc...@gmail.com> wrote:

> Hi all, I am currently using asterisk 11, and I am trying to figure out
> how to set the uri parameter telephone-context.
> I need to set it for outbound calls for a specific carrier when making
> emergency calls and don't seem able to find the option to set it.
>
> Regards
> Impy
> aka Mick
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Re: [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)

2016-02-11 Thread Trey Hilyard 
I am stumped so far. What is most interesting to me is that Asterisk is
actually sending two BYE transactions for the same dialog, at basically the
same time. I am still going through your traces again, but maybe someone
else has suggestions on how to add more debug to the res_pjsip logging that
would prove useful.

On Thu, Feb 11, 2016 at 1:33 AM Dmitriy Serov <serov@gmail.com> wrote:

> The call initiated from internal extension.
>
> I have made two test call:
> Successful call from device on res_pjsip via endpoint on chan_sip:
> http://pastebin.com/LWeDYstj
> Unsuccessful call from device on res_pjsip via endpoint on res_pjsip:
> http://pastebin.com/hepVb6Nu
>
> And ones again i don't see anything that would make asterisk send BYE.
>
> I would be grateful for any ideas.
>
> 11.02.2016 1:47, Trey Hilyard пишет:
>
> How are you initiating the call out to that server? Are you dialing from
> an internal phone or doing it from the CLI? It looks like it is from an
> internal extension, if I were guessing, but that side of the call isn't in
> your log.
>
> If it is from an internal extension, I think a SIP trace on that side
> would help.
>
> On Wed, Feb 10, 2016, 3:20 PM Dmitriy Serov <serov@gmail.com> wrote:
>
>> Please help find the cause of strange behavior res_pjsip.
>>
>> Making outgoint call to other sip server (CommuniGatePro), my asterisk
>> suddenly sends BYE after picking up!
>> Partial log of an outgoing call with full debug is attached and on web:
>> http://pastebin.com/tLNCpx4d
>>
>> No diagnostic messages why asterisk suddenly decided to hangup i don't
>> found :(
>>
>> There are suggestions or strong belief about the reasons of such behavior?
>>
>> Thanks.
>>
>> Dmitriy.
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Re: [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)

2016-02-10 Thread Trey Hilyard 
How are you initiating the call out to that server? Are you dialing from an
internal phone or doing it from the CLI? It looks like it is from an
internal extension, if I were guessing, but that side of the call isn't in
your log.

If it is from an internal extension, I think a SIP trace on that side would
help.

On Wed, Feb 10, 2016, 3:20 PM Dmitriy Serov  wrote:

> Please help find the cause of strange behavior res_pjsip.
>
> Making outgoint call to other sip server (CommuniGatePro), my asterisk
> suddenly sends BYE after picking up!
> Partial log of an outgoing call with full debug is attached and on web:
> http://pastebin.com/tLNCpx4d
>
> No diagnostic messages why asterisk suddenly decided to hangup i don't
> found :(
>
> There are suggestions or strong belief about the reasons of such behavior?
>
> Thanks.
>
> Dmitriy.
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[asterisk-users] PJSIP Returning 421 Extension Required

2016-01-13 Thread Trey Hilyard 
I am turning up a PJSIP Endpoint and am having problems when they send an
INVITE to my server. Asterisk is returning a 421 Extenstion Required. Since
"extension" means different things in the SIP stack versus Asterisk, I
don't know what it is complaining about.

I have attached the trace below. Nothing else shows up with core verbose or
core debug enabled, so I am assuming it has to be dying at the PJSIP
module. The INVITE does come from an abnormal UDP Port, which is also shown
in the Via header, but the fact that the PBX is responding makes me think
that isn't the culprit.

Any thoughts?

SIP Logger:
INVITE sip:+18165116504@12.4.240.200:5060;user=phone SIP/2.0
v: SIP/2.0/UDP 10.77.27.103:20065
;branch=z9hG4bK0020C575A392E895C39051;oc-accept
Max-Forwards: 70
t: 
f: ;tag=10847511385389740959
i: 117620342110831512016142@10.77.27.103
CSeq: 1 INVITE
d: no-fork
Privacy: none
P-Asserted-Identity: 
Require: 100rel
Accept: application/sdp
k: histinfo,resource-priority
c: application/sdp
m: 
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,UPDATE
l:   228

v=0
o=PVG 1452710812870 1452710812870 IN IP4 10.77.160.55
s=-
c=IN IP4 10.77.160.55
t=0 0
m=audio 37700 RTP/AVP 0 101
b=AS:80
b=RR:0
b=RS:0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20

<--- Transmitting SIP response (495 bytes) to UDP:10.77.27.103:20065 --->
SIP/2.0 421 Extension Required
Via: SIP/2.0/UDP 10.77.27.103:20065
;rport=20065;received=10.77.27.103;branch=z9hG4bK0020C575A392E895C39051;oc-accept
Call-ID: 117620342110831512016142@10.77.27.103
From: ;tag=10847511385389740959
To: ;tag=z9hG4bK0020C575A392E895C39051
CSeq: 1 INVITE
Require: 100rel
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 13.3.0-rc1
Content-Length:  0

PJSIP Endpoint:
zeus*CLI> pjsip show endpoint erc905

 Endpoint:  
   
I/OAuth:
 
Aor:  
 
  Contact:  
   
  Transport:
 
   Identify:
 
Match:  
Channel:  
   
Exten:   CLCID: 
 
=

 Endpoint:  erc905   Invalid
0 of inf
Aor:  erc905 0
  Contact:  erc905/sip:10.77.27.103:5060 Avail
 32.887
  Transport:  ngvn  udp  0 40  12.4.240.200:5060
   Identify:  erc905_1/erc905
Match: 10.77.27.103/32


 ParameterName : ParameterValue
 
 100rel: required
 accountcode   :
 aggregate_mwi : true
 allow : (ulaw)
 allow_subscribe   : true
 allow_transfer: true
 aors  : erc905
 auth  :
 call_group:
 callerid  : 
 callerid_privacy  : allowed_not_screened
 callerid_tag  :
 connected_line_method : invite
 context   : from_pstn
 cos_audio : 0
 cos_video : 0
 device_state_busy_at  : 0
 direct_media  : true
 direct_media_glare_mitigation : none
 direct_media_method   : invite
 disable_direct_media_on_nat   : false
 dtls_ca_file  :
 dtls_ca_path  :
 dtls_cert_file:
 dtls_cipher   :
 dtls_fingerprint  : SHA-256
 dtls_private_key  :
 dtls_rekey: 0
 dtls_setup: active
 dtls_verify   : No
 dtmf_mode : rfc4733
 fax_detect: false
 force_avp : false
 force_rport   : true
 from_domain   :
 from_user :
 ice_support   : false
 identify_by   : username
 inband_progress   : false
 language  :
 mailboxes :
 media_address :
 media_encryption  : none
 media_encryption_optimistic   : false
 media_use_received_transport  : false
 message_context   :
 moh_suggest   : default
 mwi_from_user :
 named_call_group  :
 named_pickup_group:
 one_touch_recording   : false
 outbound_auth :
 outbound_proxy:
 pickup_group  :
 record_off_feature: automixmon
 record_on_feature : automixmon
 rewrite_contact   : false
 rtp_engine: asterisk
 rtp_ipv6

Re: [asterisk-users] How to use TRUNK only if IAX fails?

2015-05-31 Thread Trey Hilyard 
I would especially look at the CHANUNAVAIL dial status Since it sounds like
you are probably qualifying your IAX trunk, that status will be the
quickest way to overflow from IAX to TDM.

On Sat, May 30, 2015, 11:35 PM Ashwin Surendran 
ashwin.surend...@now-health.com wrote:

  Hi Matt,



 I was a bit concerned on the delay if there might  be any when my iax link
 is down?

 It would be two dial steps right when my iax link is down.

 But I’m more than happy to try.



 Many Thanks,

 Ashwin.

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Matt Riddell
 (lists)
 *Sent:* 30 May 2015 16:55
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] How to use TRUNK only if IAX fails?



 The command he gave you was in Asterisk. Why do you not want to call it to
 try it?



 Then you can fail over to the other trunk if the IAX link is down.

 Kind regards,



   Matt


 On May 30, 2015, at 2:03 AM, Ashwin Surendran 
 ashwin.surend...@now-health.com wrote:

  Many Thanks Carlos, I was hoping to check whether the remote server is
 available before I issue the dial in my dial plan.

 Is there a better way to do it in asterisk without using unix commands?


 Many Thanks,
 Ashwin

 On 5/30/15, 2:06 AM, Carlos Chavez cur...@telecomabmex.com wrote:


  On 5/29/15 1:16 PM, Ashwin Surendran wrote:

  Hi,

   I have multiple Asterisk servers in various parts of the world all

   connected using dedicated VPN¹s.



   Each of these servers have iax and dahdi TRUNK configured on them.



   Occasionally the VPN¹s fail.



   What I want to be able to do is on my dial plan, use IAX if the asterisk

   server can reach the remote server using the internet OR, use TRUNK only

   if it can¹t use IAX.



   Any ideas on how this can be implemented on the dial plan?





 Check the DIALSTATUS variable to see if the IAX trunk failed and

  then dial via DAHDI.



  https://wiki.asterisk.org/wiki/display/AST/Dial+Channel+Variables



  --

  Telecomunicaciones Abiertas de México S.A. de C.V.

  Carlos Chávez

  +52 (55)9116-91161





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Re: [asterisk-users] PJSIP Sends BYE with Wrong IP

2015-04-02 Thread Trey Hilyard 
On Thu, Apr 2, 2015 at 10:43 AM, Rusty Newton rnew...@digium.com wrote:

 On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard kct...@gmail.com wrote:

 Hello -

 I am trying to decide if I have stumbled across a bug in PJSIP or I am
 just missing something. My Asterisk has two interfaces, an internal eth0
 and an external eth1. In pjsip.conf, I define the following transports:

 [trusted]
 type=transport
 protocol=udp
 bind=10.xx.yy.zz:5060

 [untrusted]
 type=transport
 protocol=udp
 bind=12.4.aa.bb:5060

 My internal endpoints use transport=internal and external endpoints use
 transport=external. I guess that's obvious.

 You show transports trusted and untrusted, you don't show any transports
 named internal and external... so that is confusing.



You are right. That is my fault that I was sanitizing the configuration for
the purpose of this email and uses different names.


 Everything works fine, most of the time. INVITEs, 1XX, 2XX are sent to the
 right interface using the right source IP. But, when Asterisk tries to send
 a BYE to any internal endpoint, it sends using the external IP, but it is
 sent of the correct internal interface eth0. Only the IP layer is
 incorrect. The SIP layer has the correct IP in the Via header. From what I
 can tell, only BYE is affected.

 I didn't have this problem with chan_sip. Am I just missing some
 configuration?

 This sounds like improper configuration, or a bug.

 If you can pastebin a full (sanitized) pjsip.conf as well as an Asterisk
 log with verbose turned up[1], plus a SIP packet trace then we can take a
 look at it.

 [1]:
 https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information


 I actually got the issue resolved by upgrading to 13.3.rc-1, since this is
just my development system. I assume that the problem was resolved between
the two releases.
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[asterisk-users] PJSIP Sends BYE with Wrong IP

2015-04-01 Thread Trey Hilyard 
Hello -

I am trying to decide if I have stumbled across a bug in PJSIP or I am just
missing something. My Asterisk has two interfaces, an internal eth0 and
an external eth1. In pjsip.conf, I define the following transports:

[trusted]
type=transport
protocol=udp
bind=10.xx.yy.zz:5060

[untrusted]
type=transport
protocol=udp
bind=12.4.aa.bb:5060

My internal endpoints use transport=internal and external endpoints use
transport=external. I guess that's obvious.

My netstat shows both transports listening:
Proto Recv-Q Send-Q Local Address   Foreign Address State
PID/Program name
udp0  0 12.4.aa.bb:5060   0.0.0.0:*
  25494/asterisk
udp0  0 10.xx.yy.xx:5060  0.0.0.0:*
  25494/asterisk

Everything works fine, most of the time. INVITEs, 1XX, 2XX are sent to the
right interface using the right source IP. But, when Asterisk tries to send
a BYE to any internal endpoint, it sends using the external IP, but it is
sent of the correct internal interface eth0. Only the IP layer is
incorrect. The SIP layer has the correct IP in the Via header. From what I
can tell, only BYE is affected.

I didn't have this problem with chan_sip. Am I just missing some
configuration?

To test, I have set up the most simple extension I can think of, and it
duplicates the condition:
exten = _9090,1,Answer
 same = n,Wait(2)
 same = n,Hangup
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Re: [asterisk-users] PJSIP Endpoint AOR question

2015-04-01 Thread Trey Hilyard 
I just realized that you are asking about dynamic AORs, not static Contacts
in an AOR. That may be the difference. I have never actually tried giving a
dynamic AOR a different name. And you wouldn't want more than one dynamic
AOR, you'd just use an AOR that allowed more than 1 contact.

On Wed, Apr 1, 2015 at 2:59 PM Trey Hilyard kct...@gmail.com wrote:

 I don't know why you have issues using different names. I have multiple
 AORs assigned to a single endpoint and it works fine. I have to admit that
 my AORs do contain the endpoint name, though. For example, for endpoint
 myswitch I have two AORs, myswitch_1 and myswitch_2, and I assign
 them to the endpoint with aors=myswitch_1,myswitch_2.

 When you say that the first example fails, what specifically fails? Do the
 PJSIP modules load but then you get an error when trying to dial to an
 endpoint, or maybe receive a call from one?

 On Wed, Apr 1, 2015 at 2:53 PM Dan Cropp d...@amtelco.com wrote:

 I am running asterisk 13.1.0



 In pjsip.conf, the endpoint section has an aors and an auth field.



 I can name the auth field anything I want.  The key is to set the
 auth=field accordingly.

 However, when I try this with the aors field, it never works.  It seems I
 have to name the aors=field to match the name of the endpoint section.



 Is this correct?



 Would there ever be a need for multiple aors to a single endpoint?  Since
 the field is named aors, I thought this would be possible.  How would I do
 this if I have to name the aor the name of the endpoint?



 This fails...



 [transport1]

 type = transport

 bind = 0.0.0.0

 protocol = udp



 [aor3]

 type = aor

 max_contacts = 1

 remove_existing = yes



 [auth3]

 type = auth

 username = 1003

 password = Password



 [1003]

 type = endpoint

 context = Test

 transport = transport1

 auth = auth3

 aors = aor3

 dtmf_mode = inband

 device_state_busy_at = 1

 disallow = all

 allow = ulaw





 This succeeds...



 [transport1]

 type = transport

 bind = 0.0.0.0

 protocol = udp



 [1003]

 type = aor

 max_contacts = 1

 remove_existing = yes



 [auth3]

 type = auth

 username = 1003

 password = Password



 [1003]

 type = endpoint

 context = Test

 transport = transport1

 auth = auth3

 aors = 1003

 dtmf_mode = inband

 device_state_busy_at = 1

 disallow = all

 allow = ulaw


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Re: [asterisk-users] PJSIP Endpoint AOR question

2015-04-01 Thread Trey Hilyard 
I don't know why you have issues using different names. I have multiple
AORs assigned to a single endpoint and it works fine. I have to admit that
my AORs do contain the endpoint name, though. For example, for endpoint
myswitch I have two AORs, myswitch_1 and myswitch_2, and I assign
them to the endpoint with aors=myswitch_1,myswitch_2.

When you say that the first example fails, what specifically fails? Do the
PJSIP modules load but then you get an error when trying to dial to an
endpoint, or maybe receive a call from one?

On Wed, Apr 1, 2015 at 2:53 PM Dan Cropp d...@amtelco.com wrote:

 I am running asterisk 13.1.0



 In pjsip.conf, the endpoint section has an aors and an auth field.



 I can name the auth field anything I want.  The key is to set the
 auth=field accordingly.

 However, when I try this with the aors field, it never works.  It seems I
 have to name the aors=field to match the name of the endpoint section.



 Is this correct?



 Would there ever be a need for multiple aors to a single endpoint?  Since
 the field is named aors, I thought this would be possible.  How would I do
 this if I have to name the aor the name of the endpoint?



 This fails...



 [transport1]

 type = transport

 bind = 0.0.0.0

 protocol = udp



 [aor3]

 type = aor

 max_contacts = 1

 remove_existing = yes



 [auth3]

 type = auth

 username = 1003

 password = Password



 [1003]

 type = endpoint

 context = Test

 transport = transport1

 auth = auth3

 aors = aor3

 dtmf_mode = inband

 device_state_busy_at = 1

 disallow = all

 allow = ulaw





 This succeeds...



 [transport1]

 type = transport

 bind = 0.0.0.0

 protocol = udp



 [1003]

 type = aor

 max_contacts = 1

 remove_existing = yes



 [auth3]

 type = auth

 username = 1003

 password = Password



 [1003]

 type = endpoint

 context = Test

 transport = transport1

 auth = auth3

 aors = 1003

 dtmf_mode = inband

 device_state_busy_at = 1

 disallow = all

 allow = ulaw


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[asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown

2015-03-26 Thread Trey Hilyard 
I found an issue with how PJSIP handles a typo in the Dial application. If
the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//...),
the Dial applications fails (obviously), but it also kills the server.

I put some code in my pbx_config to check for that string and not let the
dialplan reload, but it seems like there should be a better way to handle
in in the PJSIP stack or Dial app so that it doesn't take the server down
if it gets through.

I am not a developer, but I was hoping maybe someone who monitors this
mailing list might feel like taking this on as a bug fix.I haven't tried
with any other channel drivers, so it may cross to others.
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Re: [asterisk-users] outbound calls

2015-03-20 Thread Trey Hilyard 
I am making some assumptions, but assuming the 217.195.xx.xxx is your
provider, you are getting this back from them:

Got SIP response 556 No address found back from 217.195.xx.xxx:5060

Are you sure that 0033149xx is the format the provider is expecting?
You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what
the INVITE looks like, but normally a 556 indicates that your provider
didn't have routing for either the R-URI or they didn't recognize that is
was coming from you. You might compare the SIP INVITE coming from Asterisk
to the one from Z-Lite and see where the differences are.



On Fri, Mar 20, 2015 at 12:03 PM Salaheddine Elharit 
salah.elharit...@gmail.com wrote:

 hello list

 i have an issue related to outbound calls i can contact all the number
 except on number given by our provider in trunk

 the issue just when i configure my trunk in our server but when i
 configure the trunk directly in x-lite i can contact this number without
 issue

 below the cli

   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
 -- Executing [0149xx@from-internal:1] Macro(SIP/101-0103,
 user-callerid,LIMIT,EXTERNAL,) in new stack
 -- Executing [s@macro-user-callerid:1] Set(SIP/101-0103,
 TOUCH_MONITOR=1426869820.301) in new stack
 -- Executing [s@macro-user-callerid:2] Set(SIP/101-0103,
 AMPUSER=101) in new stack
 -- Executing [s@macro-user-callerid:3] GotoIf(SIP/101-0103,
 0?report) in new stack
 -- Executing [s@macro-user-callerid:4] ExecIf(SIP/101-0103,
 1?Set(REALCALLERIDNUM=101)) in new stack
 -- Executing [s@macro-user-callerid:5] Set(SIP/101-0103,
 AMPUSER=101) in new stack
 -- Executing [s@macro-user-callerid:6] GotoIf(SIP/101-0103,
 0?limit) in new stack
 -- Executing [s@macro-user-callerid:7] Set(SIP/101-0103,
 AMPUSERCIDNAME=101) in new stack
 -- Executing [s@macro-user-callerid:8] GotoIf(SIP/101-0103,
 0?report) in new stack
 -- Executing [s@macro-user-callerid:9] Set(SIP/101-0103,
 AMPUSERCID=101) in new stack
 -- Executing [s@macro-user-callerid:10] Set(SIP/101-0103,
 __DIAL_OPTIONS=tr) in new stack
 -- Executing [s@macro-user-callerid:11] Set(SIP/101-0103,
 CALLERID(all)=101 101) in new stack
 -- Executing [s@macro-user-callerid:12] GotoIf(SIP/101-0103,
 0?limit) in new stack
 -- Executing [s@macro-user-callerid:13] ExecIf(SIP/101-0103,
 1?Set(GROUP(concurrency_limit)=101)) in new stack
 -- Executing [s@macro-user-callerid:14] ExecIf(SIP/101-0103,
 0?Set(CHANNEL(language)=)) in new stack
 -- Executing [s@macro-user-callerid:15] GotoIf(SIP/101-0103,
 1?continue) in new stack
 -- Goto (macro-user-callerid,s,28)
 -- Executing [s@macro-user-callerid:28] Set(SIP/101-0103,
 CALLERID(number)=101) in new stack
 -- Executing [s@macro-user-callerid:29] Set(SIP/101-0103,
 CALLERID(name)=101) in new stack
 -- Executing [s@macro-user-callerid:30] Set(SIP/101-0103,
 CDR(cnum)=101) in new stack
 -- Executing [s@macro-user-callerid:31] Set(SIP/101-0103,
 CDR(cnam)=101) in new stack
 -- Executing [s@macro-user-callerid:32] Set(SIP/101-0103,
 CHANNEL(language)=en) in new stack
 -- Executing [0149xx@from-internal:2] Set(SIP/101-0103,
 MOHCLASS=default) in new stack
 -- Executing [0149xx@from-internal:3] Set(SIP/101-0103,
 _NODEST=) in new stack
 -- Executing [0149xx@from-internal:4] Gosub(SIP/101-0103,
 sub-record-check,s,1(out,0149xx,)) in new stack
 -- Executing [s@sub-record-check:1] Set(SIP/101-0103,
 REC_POLICY_MODE_SAVE=) in new stack
 -- Executing [s@sub-record-check:2] GotoIf(SIP/101-0103,
 1?check) in new stack
 -- Goto (sub-record-check,s,7)
 -- Executing [s@sub-record-check:7] Set(SIP/101-0103,
 __MON_FMT=wav) in new stack
 -- Executing [s@sub-record-check:8] GotoIf(SIP/101-0103,
 1?next) in new stack
 -- Goto (sub-record-check,s,11)
 -- Executing [s@sub-record-check:11] ExecIf(SIP/101-0103,
 0?Return()) in new stack
 -- Executing [s@sub-record-check:12] ExecIf(SIP/101-0103,
 0?Set(__REC_POLICY_MODE=)) in new stack
 -- Executing [s@sub-record-check:13] GotoIf(SIP/101-0103,
 0?out,1) in new stack
 -- Executing [s@sub-record-check:14] Set(SIP/101-0103,
 __REC_STATUS=INITIALIZED) in new stack
 -- Executing [s@sub-record-check:15] Set(SIP/101-0103,
 NOW=1426869820) in new stack
 -- Executing [s@sub-record-check:16] Set(SIP/101-0103,
 __DAY=20) in new stack
 -- Executing [s@sub-record-check:17] Set(SIP/101-0103,
 __MONTH=03) in new stack
 -- Executing [s@sub-record-check:18] Set(SIP/101-0103,
 __YEAR=2015) in new stack
 -- Executing [s@sub-record-check:19] Set(SIP/101-0103,
 __TIMESTR=20150320-164340) in new stack
 -- Executing [s@sub-record-check:20] Set(SIP/101-0103,
 __FROMEXTEN=101) in new stack
 -- Executing [s@sub-record-check:21]

Re: [asterisk-users] outbound calls

2015-03-20 Thread Trey Hilyard 
So you are saying that it resolved the issue to activate voicemail on the
device that sits past your trunk provider? That confuses me a little, but
if your calls are working, that's great news.

On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit 
salah.elharit...@gmail.com wrote:

 i noticed that when i active the voicemail in the IP-phone where the
 number 0033149xx is configured i can call this number without issue

 Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
 -- Called SIP/FD/0033149xx == Begin MixMonitor Recording
 SIP/101-010d
 -- SIP/FD-010e is making progress passing it to SIP/101-010d
 0x2b393cfc2610 -- Probation passed - setting RTP source address
 to 192.
168.1.138:55542
 0x1d08efa0 -- Probation passed - setting RTP source address to
  217.195.xx.xx:46346
 -- SIP/FD-010e answered SIP/101-010d
 0x1d08efa0 -- Probation passed - setting RTP source address to
  217.195.xx.xx:46346
 thanks and regards.


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