Sonny Rajagopalan
2015-Jan-08 19:32 UTC
[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is now [transport-udp] type=transport protocol=udp bind=0.0.0.0 local_net=192.168.1.0/24 ;Templates for the necessary config sections [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!) type=aor max_contacts=1 remove_existing=yes ;Definitions for our phones, using the templates above [demo-alice](endpoint_internal) auth=demo-alice aors=demo-alice mailboxes=box_a rewrite_contact=yes [demo-alice](auth_userpass) password=demo-alice ; put a strong, unique password here instead username=demo-alice [demo-alice](aor_dynamic) [demo-bob](endpoint_internal) auth=demo-bob aors=demo-bob mailboxes=box_b rewrite_contact=yes [demo-bob](auth_userpass) password=demo-bob ; put a strong, unique password here instead username=demo-bob [demo-bob](aor_dynamic) Thank you for your help! On Thu, Jan 8, 2015 at 11:48 AM, Scott Griepentrog <sgriepentrog at digium.com> wrote:> It would appear that you have the Asterisk server on a public IP address, > your two endpoints are behind a NAT, and you have rewrite_contact enabled > in pjsip.conf. > > In which case, what you are seeing is correct. In order to be able to > send a call to an extension where it is behind NAT, Asterisk must update > the contact to have the current IP and port that the phone registered via > (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state > for). > > On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I am following the instructions in >> https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I >> am trying to make a call from extension Alice (6001) to extension for Bob >> (6002). When I make the call, I can hear the ringing on Alice's phone >> (caller), but Bob's phone (callee) doesn't ring, or show a call coming in >> from Alice. My setup and environment is as follows: Alice, Bob and Asterisk >> all in the same 192.168.1.0/24 network, and they are able to register to >> the Asterisk server running 13.1.0/PJSIP. The rest of the configuration is >> the same as the aforementioned wiki page, but is shown here for clarity: >> >> root at asterisk13FFP:/var/log/asterisk# more /etc/asterisk/extensions.conf >> [from-internal] >> exten=>6001,1,Dial(PJSIP/demo-alice) >> exten=>6002,1,Dial(PJSIP/demo-bob) >> exten=>6003,1,Answer() >> same =>6003,n,Playback(hello-world) >> same =>6003,n,Hangup() >> >> >> What I do observe is that I when I request the output of pjsip show >> endpoints, I get Contact information for the two SIP peers that have >> registered different from their actual IP addresses. I suspect this has >> something to do with their calls being routed elsewhere. If my assumption >> is correct--how do I fix this? Alice should be at 192.168.1.50 and Bob >> should be at 192.168.1.149, instead, they (both) show IP address >> 146.115.163.234. Any help is deeply appreciated. Thanks. >> >> asterisk13FFP*CLI> pjsip show endpoints >> >> Endpoint: <Endpoint/CID.....................................> >> <State.....> <Channels.> >> I/OAuth: >> <AuthId/UserName...........................................................> >> Aor: <Aor............................................> >> <MaxContact> >> Contact: <Aor/ContactUri...............................> >> <Status....> <RTT(ms)..> >> Transport: <TransportId........> <Type> <cos> <tos> >> <BindAddress..................> >> Identify: >> <Identify/Endpoint.........................................................> >> Match: <ip/cidr.........................> >> Channel: <ChannelId......................................> >> <State.....> <Time(sec)> >> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> >> >> ========================================================================================>> >> Endpoint: demo-alice >> Unavailable 0 of inf >> InAuth: demo-alice/demo-alice >> Aor: demo-alice 1 >> Contact: demo-alice/sip:demo-alice@*146.115.163.234*:38519 >> Unknown nan >> >> Endpoint: demo-bob Not in >> use 0 of inf >> InAuth: demo-bob/demo-bob >> Aor: demo-bob 1 >> Contact: demo-bob/sip:demo-bob@*146.115.163.234*:38321;tra >> Unknown nan >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > > -- > [image: Digium logo] > Scott Griepentrog > Digium, Inc ? Software Developer > 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US > direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090 > Check us out at: http://digium.com ? http://asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... 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Sonny Rajagopalan
2015-Jan-09 01:03 UTC
[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
Well, I thought it worked, but it actually doesn't--I am able to get the
caller pick up the phone, but for some reason, I cannot hear anything on
either side no matter who does the calling. Again, my two SIP phones are on
the local 192.168.1.0/24 network (do not go over the Internet) and the
Asterisk server is located in the same network (not accessed over the
Internet). Any help is appreciated.
Does the fact that Asterisk is running on a VirtualBox VM on the same
machine as one of the SIP phones matter? I am able to access the ARI REST
interface of the Asterisk server quite fine on the host machine.
I suspect it has to do with RTP not being set up, but all the codec support
is there. Here's a log for the SIP request from 192.168.1.50:
<--- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009 --->
INVITE sip:6002 at 192.168.1.139;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 146.115.163.234:64009
;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-
Max-Forwards: 70
Contact: <sip:demo-alice at 146.115.163.234:64009;transport=UDP>
To: <sip:6002 at 192.168.1.139;transport=UDP>
From: <sip:demo-alice at 192.168.1.139;transport=UDP>;tag=b661670b
Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.21933 r21903
Authorization: Digest
username="demo-alice",realm="asterisk",nonce="[removed]",uri="
sip:6002 at 192.168.1.139
;transport=UDP",response="[removed]",cnonce="[removed]",nc=00000001,qop=auth,algorithm=md5,opaque="[removed]"
Allow-Events: presence, kpml
Content-Length: 245
v=0
o=Z 0 0 IN IP4 146.115.163.234
s=Z
c=IN IP4 146.115.163.234
t=0 0
m=audio 8000 RTP/AVP 0 3 110 8 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<--- Transmitting SIP response (319 bytes) to UDP:192.168.1.50:64009 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 146.115.163.234:64009
;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-
Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.
From: <sip:demo-alice at 192.168.1.139>;tag=b661670b
To: <sip:6002 at 192.168.1.139>
CSeq: 2 INVITE
Content-Length: 0
Any help is appreciated. A topology is shown below in ASCII.
< ( Big bad Internet ) >
GW/NAPT/Router
|
----------------------------------------------------------
/ \
| |
Host A Host B
-----------------
-----------------
| Alice | | Bob
|
| 192.168.1.50 | |
192.168.1.149 |
|---------------|
|---------------|
| Asterisk sr |
| (VM) |
| 192.168.1.239 |
|---------------|
On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> Thank you for your note, Scott.
>
> I set rewrite_contact=yes for both contacts, and I also had to do
> remove_existing=yes because I had to remove the existing contact
> information (max_contacts = 1 was preventing new contact information)
> using pjsip qualify demo-alice etc., after which the right IP addresses
> showed in pjsip show endpoints. Anyway, it works as expected now, I
> think. My pjsip.conf is now
>
> [transport-udp]
> type=transport
> protocol=udp
> bind=0.0.0.0
> local_net=192.168.1.0/24
> ;Templates for the necessary config sections
>
> [endpoint_internal](!)
> type=endpoint
> context=from-internal
> disallow=all
> allow=ulaw
>
> [auth_userpass](!)
> type=auth
> auth_type=userpass
>
> [aor_dynamic](!)
> type=aor
> max_contacts=1
> remove_existing=yes
> ;Definitions for our phones, using the templates above
>
> [demo-alice](endpoint_internal)
> auth=demo-alice
> aors=demo-alice
> mailboxes=box_a
> rewrite_contact=yes
> [demo-alice](auth_userpass)
> password=demo-alice ; put a strong, unique password here instead
> username=demo-alice
>
> [demo-alice](aor_dynamic)
>
> [demo-bob](endpoint_internal)
> auth=demo-bob
> aors=demo-bob
> mailboxes=box_b
> rewrite_contact=yes
> [demo-bob](auth_userpass)
> password=demo-bob ; put a strong, unique password here instead
> username=demo-bob
>
> [demo-bob](aor_dynamic)
>
>
> Thank you for your help!
>
> On Thu, Jan 8, 2015 at 11:48 AM, Scott Griepentrog <
> sgriepentrog at digium.com> wrote:
>
>> It would appear that you have the Asterisk server on a public IP
address,
>> your two endpoints are behind a NAT, and you have rewrite_contact
enabled
>> in pjsip.conf.
>>
>> In which case, what you are seeing is correct. In order to be able to
>> send a call to an extension where it is behind NAT, Asterisk must
update
>> the contact to have the current IP and port that the phone registered
via
>> (i.e. the WAN IP of the NAT, and the WAN port that it is retaining
state
>> for).
>>
>> On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan <
>> sonny.rajagopalan at gmail.com> wrote:
>>
>>> I am following the instructions in
>>> https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality
and
>>> I am trying to make a call from extension Alice (6001) to extension
for Bob
>>> (6002). When I make the call, I can hear the ringing on Alice's
phone
>>> (caller), but Bob's phone (callee) doesn't ring, or show a
call coming in
>>> from Alice. My setup and environment is as follows: Alice, Bob and
Asterisk
>>> all in the same 192.168.1.0/24 network, and they are able to
register
>>> to the Asterisk server running 13.1.0/PJSIP. The rest of the
configuration
>>> is the same as the aforementioned wiki page, but is shown here for
clarity:
>>>
>>> root at asterisk13FFP:/var/log/asterisk# more
/etc/asterisk/extensions.conf
>>> [from-internal]
>>> exten=>6001,1,Dial(PJSIP/demo-alice)
>>> exten=>6002,1,Dial(PJSIP/demo-bob)
>>> exten=>6003,1,Answer()
>>> same =>6003,n,Playback(hello-world)
>>> same =>6003,n,Hangup()
>>>
>>>
>>> What I do observe is that I when I request the output of pjsip show
>>> endpoints, I get Contact information for the two SIP peers that
have
>>> registered different from their actual IP addresses. I suspect this
has
>>> something to do with their calls being routed elsewhere. If my
assumption
>>> is correct--how do I fix this? Alice should be at 192.168.1.50 and
Bob
>>> should be at 192.168.1.149, instead, they (both) show IP address
>>> 146.115.163.234. Any help is deeply appreciated. Thanks.
>>>
>>> asterisk13FFP*CLI> pjsip show endpoints
>>>
>>> Endpoint:
<Endpoint/CID.....................................>
>>> <State.....> <Channels.>
>>> I/OAuth:
>>>
<AuthId/UserName...........................................................>
>>> Aor:
<Aor............................................>
>>> <MaxContact>
>>> Contact:
<Aor/ContactUri...............................>
>>> <Status....> <RTT(ms)..>
>>> Transport: <TransportId........> <Type>
<cos> <tos>
>>> <BindAddress..................>
>>> Identify:
>>>
<Identify/Endpoint.........................................................>
>>> Match: <ip/cidr.........................>
>>> Channel:
<ChannelId......................................>
>>> <State.....> <Time(sec)>
>>> Exten: <DialedExten...........> CLCID:
<ConnectedLineCID.......>
>>>
>>>
========================================================================================>>>
>>> Endpoint: demo-alice
>>> Unavailable 0 of inf
>>> InAuth: demo-alice/demo-alice
>>> Aor: demo-alice 1
>>> Contact: demo-alice/sip:demo-alice@*146.115.163.234*:38519
>>> Unknown nan
>>>
>>> Endpoint: demo-bob
Not in
>>> use 0 of inf
>>> InAuth: demo-bob/demo-bob
>>> Aor: demo-bob 1
>>> Contact: demo-bob/sip:demo-bob@*146.115.163.234*:38321;tra
>>> Unknown nan
>>>
>>>
>>> --
>>>
_____________________________________________________________________
>>> -- 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
>>>
>>
>>
>>
>> --
>> [image: Digium logo]
>> Scott Griepentrog
>> Digium, Inc ? Software Developer
>> 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
>> direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
>> Check us out at: http://digium.com ? http://asterisk.org
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Patrick Beaumont
2015-Jan-09 09:01 UTC
[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue [Spam score:10%]
My suspicion would be that the line
o=Z 0 0 IN IP4 146.115.163.234?
is causing the problem. Your SIP client is reporting it's external IP
address for the audio stream rather than it's internal one. I would look at
the settings in your sip client to see if it has any automatic NAT stuff (like
using a STUN server) and disable it.
Regards,
Patrick.
________________________________
From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at
lists.digium.com> on behalf of Sonny Rajagopalan <sonny.rajagopalan at
gmail.com>
Sent: 09 January 2015 01:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue [Spam
score:10%]
Well, I thought it worked, but it actually doesn't--I am able to get the
caller pick up the phone, but for some reason, I cannot hear anything on either
side no matter who does the calling. Again, my two SIP phones are on the local
192.168.1.0/24<http://192.168.1.0/24> network (do not go over the
Internet) and the Asterisk server is located in the same network (not accessed
over the Internet). Any help is appreciated.
Does the fact that Asterisk is running on a VirtualBox VM on the same machine as
one of the SIP phones matter? I am able to access the ARI REST interface of the
Asterisk server quite fine on the host machine.
I suspect it has to do with RTP not being set up, but all the codec support is
there. Here's a log for the SIP request from
192.168.1.50<http://192.168.1.50/>:
<--- Received SIP request (1229 bytes) from
UDP:192.168.1.50:64009<http://192.168.1.50:64009/> --->
INVITE sip:6002 at 192.168.1.139<mailto:sip%3A6002 at
192.168.1.139>;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
146.115.163.234:64009;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-
Max-Forwards: 70
Contact: <sip:demo-alice at 146.115.163.234:64009;transport=UDP>
To: <sip:6002 at 192.168.1.139<mailto:sip%3A6002 at
192.168.1.139>;transport=UDP>
From: <sip:demo-alice at 192.168.1.139<mailto:sip%3Ademo-alice at
192.168.1.139>;transport=UDP>;tag=b661670b
Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.21933 r21903
Authorization: Digest
username="demo-alice",realm="asterisk",nonce="[removed]",uri="sip:6002
at 192.168.1.139<mailto:sip%3A6002 at
192.168.1.139>;transport=UDP",response="[removed]",cnonce="[removed]",nc=00000001,qop=auth,algorithm=md5,opaque="[removed]"
Allow-Events: presence, kpml
Content-Length: 245
v=0
o=Z 0 0 IN IP4 146.115.163.234
s=Z
c=IN IP4 146.115.163.234
t=0 0
m=audio 8000 RTP/AVP 0 3 110 8 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<--- Transmitting SIP response (319 bytes) to
UDP:192.168.1.50:64009<http://192.168.1.50:64009/> --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
146.115.163.234:64009;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-
Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.
From: <sip:demo-alice at 192.168.1.139<mailto:sip%3Ademo-alice at
192.168.1.139>>;tag=b661670b
To: <sip:6002 at 192.168.1.139<mailto:sip%3A6002 at 192.168.1.139>>
CSeq: 2 INVITE
Content-Length: 0
Any help is appreciated. A topology is shown below in ASCII.
< ( Big bad Internet ) >
GW/NAPT/Router
|
----------------------------------------------------------
/ \
| |
Host A Host B
-----------------
-----------------
| Alice | | Bob
|
| 192.168.1.50 | | 192.168.1.149
|
|---------------|
|---------------|
| Asterisk sr |
| (VM) |
| 192.168.1.239 |
|---------------|
On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan <sonny.rajagopalan at
gmail.com<mailto:sonny.rajagopalan at gmail.com>> wrote:
Thank you for your note, Scott.
I set rewrite_contact=yes for both contacts, and I also had to do
remove_existing=yes because I had to remove the existing contact information
(max_contacts = 1 was preventing new contact information) using pjsip qualify
demo-alice etc., after which the right IP addresses showed in pjsip show
endpoints. Anyway, it works as expected now, I think. My pjsip.conf is now
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.1.0/24<http://192.168.1.0/24>
;Templates for the necessary config sections
[endpoint_internal](!)
type=endpoint
context=from-internal
disallow=all
allow=ulaw
[auth_userpass](!)
type=auth
auth_type=userpass
[aor_dynamic](!)
type=aor
max_contacts=1
remove_existing=yes
;Definitions for our phones, using the templates above
[demo-alice](endpoint_internal)
auth=demo-alice
aors=demo-alice
mailboxes=box_a
rewrite_contact=yes
[demo-alice](auth_userpass)
password=demo-alice ; put a strong, unique password here instead
username=demo-alice
[demo-alice](aor_dynamic)
[demo-bob](endpoint_internal)
auth=demo-bob
aors=demo-bob
mailboxes=box_b
rewrite_contact=yes
[demo-bob](auth_userpass)
password=demo-bob ; put a strong, unique password here instead
username=demo-bob
[demo-bob](aor_dynamic)
Thank you for your help!
On Thu, Jan 8, 2015 at 11:48 AM, Scott Griepentrog <sgriepentrog at
digium.com<mailto:sgriepentrog at digium.com>> wrote:
It would appear that you have the Asterisk server on a public IP address, your
two endpoints are behind a NAT, and you have rewrite_contact enabled in
pjsip.conf.
In which case, what you are seeing is correct. In order to be able to send a
call to an extension where it is behind NAT, Asterisk must update the contact to
have the current IP and port that the phone registered via (i.e. the WAN IP of
the NAT, and the WAN port that it is retaining state for).
On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan <sonny.rajagopalan at
gmail.com<mailto:sonny.rajagopalan at gmail.com>> wrote:
I am following the instructions in
https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am
trying to make a call from extension Alice (6001) to extension for Bob (6002).
When I make the call, I can hear the ringing on Alice's phone (caller), but
Bob's phone (callee) doesn't ring, or show a call coming in from Alice.
My setup and environment is as follows: Alice, Bob and Asterisk all in the same
192.168.1.0/24<http://192.168.1.0/24> network, and they are able to
register to the Asterisk server running 13.1.0/PJSIP. The rest of the
configuration is the same as the aforementioned wiki page, but is shown here for
clarity:
root at asterisk13FFP:/var/log/asterisk# more /etc/asterisk/extensions.conf
[from-internal]
exten=>6001,1,Dial(PJSIP/demo-alice)
exten=>6002,1,Dial(PJSIP/demo-bob)
exten=>6003,1,Answer()
same =>6003,n,Playback(hello-world)
same =>6003,n,Hangup()
What I do observe is that I when I request the output of pjsip show endpoints, I
get Contact information for the two SIP peers that have registered different
from their actual IP addresses. I suspect this has something to do with their
calls being routed elsewhere. If my assumption is correct--how do I fix this?
Alice should be at 192.168.1.50 and Bob should be at 192.168.1.149, instead,
they (both) show IP address 146.115.163.234. Any help is deeply appreciated.
Thanks.
asterisk13FFP*CLI> pjsip show endpoints
Endpoint: <Endpoint/CID.....................................>
<State.....> <Channels.>
I/OAuth:
<AuthId/UserName...........................................................>
Aor: <Aor............................................>
<MaxContact>
Contact: <Aor/ContactUri...............................>
<Status....> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos>
<tos> <BindAddress..................>
Identify:
<Identify/Endpoint.........................................................>
Match: <ip/cidr.........................>
Channel: <ChannelId......................................>
<State.....> <Time(sec)>
Exten: <DialedExten...........> CLCID:
<ConnectedLineCID.......>
========================================================================================
Endpoint: demo-alice Unavailable 0
of inf
InAuth: demo-alice/demo-alice
Aor: demo-alice 1
Contact: demo-alice/sip:demo-alice at 146.115.163.234:38519 Unknown
nan
Endpoint: demo-bob Not in use 0
of inf
InAuth: demo-bob/demo-bob
Aor: demo-bob 1
Contact: demo-bob/sip:demo-bob at 146.115.163.234:38321;tra Unknown
nan
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by
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Scott Griepentrog
2015-Jan-09 14:01 UTC
[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
?To sort out RTP problems, I would recommend: 1) on all endpoints use codec of allow=!all,ulaw -- this is or should be supported by all endpoints and eliminates any issues of mismatch, translation, etc., and can be adjusted later once everything is working 2) add an Echo() application to your dialplan so you can call it and check ?RTP to and from Asterisk 3) start with direct_media=no to run all the RTP through Asterisk first 4) packet capture at/on the asterisk server, as well as at endpoints if need be, to identfy if and where RTP streams are being sent and received. The goal being to get two way audio calls up through Asterisk, and then change one thing at a time towards your desired configuration and retest. On Thu, Jan 8, 2015 at 7:03 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote:> Well, I thought it worked, but it actually doesn't--I am able to get the > caller pick up the phone, but for some reason, I cannot hear anything on > either side no matter who does the calling. Again, my two SIP phones are on > the local 192.168.1.0/24 network (do not go over the Internet) and the > Asterisk server is located in the same network (not accessed over the > Internet). Any help is appreciated. > > Does the fact that Asterisk is running on a VirtualBox VM on the same > machine as one of the SIP phones matter? I am able to access the ARI REST > interface of the Asterisk server quite fine on the host machine. > > I suspect it has to do with RTP not being set up, but all the codec > support is there. Here's a log for the SIP request from 192.168.1.50: > > <--- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009 ---> > INVITE sip:6002 at 192.168.1.139;transport=UDP SIP/2.0 > Via: SIP/2.0/UDP 146.115.163.234:64009 > ;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- > Max-Forwards: 70 > Contact: <sip:demo-alice at 146.115.163.234:64009;transport=UDP> > To: <sip:6002 at 192.168.1.139;transport=UDP> > From: <sip:demo-alice at 192.168.1.139;transport=UDP>;tag=b661670b > Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. > CSeq: 2 INVITE > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, > SUBSCRIBE > Content-Type: application/sdp > Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri > User-Agent: Z 3.3.21933 r21903 > > Authorization: Digest > username="demo-alice",realm="asterisk",nonce="[removed]",uri=" > sip:6002 at 192.168.1.139 > ;transport=UDP",response="[removed]",cnonce="[removed]",nc=00000001,qop=auth,algorithm=md5,opaque="[removed]" > > Allow-Events: presence, kpml > Content-Length: 245 > > > v=0 > o=Z 0 0 IN IP4 146.115.163.234 > s=Z > c=IN IP4 146.115.163.234 > t=0 0 > m=audio 8000 RTP/AVP 0 3 110 8 98 101 > a=rtpmap:110 speex/8000 > a=rtpmap:98 iLBC/8000 > a=fmtp:98 mode=20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > > > <--- Transmitting SIP response (319 bytes) to UDP:192.168.1.50:64009 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 146.115.163.234:64009 > ;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- > Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. > From: <sip:demo-alice at 192.168.1.139>;tag=b661670b > To: <sip:6002 at 192.168.1.139> > CSeq: 2 INVITE > Content-Length: 0 > > Any help is appreciated. A topology is shown below in ASCII. > > > < ( Big bad Internet ) > > > GW/NAPT/Router > | > ---------------------------------------------------------- > / \ > > | | > Host A Host B > ----------------- > ----------------- > | Alice | | Bob > | > | 192.168.1.50 | | > 192.168.1.149 | > |---------------| > |---------------| > | Asterisk sr | > | (VM) | > | 192.168.1.239 | > |---------------| > > On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> Thank you for your note, Scott. >> >> I set rewrite_contact=yes for both contacts, and I also had to do >> remove_existing=yes because I had to remove the existing contact >> information (max_contacts = 1 was preventing new contact information) >> using pjsip qualify demo-alice etc., after which the right IP addresses >> showed in pjsip show endpoints. Anyway, it works as expected now, I >> think. My pjsip.conf is now >> >> [transport-udp] >> type=transport >> protocol=udp >> bind=0.0.0.0 >> local_net=192.168.1.0/24 >> ;Templates for the necessary config sections >> >> [endpoint_internal](!) >> type=endpoint >> context=from-internal >> disallow=all >> allow=ulaw >> >> [auth_userpass](!) >> type=auth >> auth_type=userpass >> >> [aor_dynamic](!) >> type=aor >> max_contacts=1 >> remove_existing=yes >> ;Definitions for our phones, using the templates above >> >> [demo-alice](endpoint_internal) >> auth=demo-alice >> aors=demo-alice >> mailboxes=box_a >> rewrite_contact=yes >> [demo-alice](auth_userpass) >> password=demo-alice ; put a strong, unique password here instead >> username=demo-alice >> >> [demo-alice](aor_dynamic) >> >> [demo-bob](endpoint_internal) >> auth=demo-bob >> aors=demo-bob >> mailboxes=box_b >> rewrite_contact=yes >> [demo-bob](auth_userpass) >> password=demo-bob ; put a strong, unique password here instead >> username=demo-bob >> >> [demo-bob](aor_dynamic) >> >> >> Thank you for your help! >> >> On Thu, Jan 8, 2015 at 11:48 AM, Scott Griepentrog < >> sgriepentrog at digium.com> wrote: >> >>> It would appear that you have the Asterisk server on a public IP >>> address, your two endpoints are behind a NAT, and you have rewrite_contact >>> enabled in pjsip.conf. >>> >>> In which case, what you are seeing is correct. In order to be able to >>> send a call to an extension where it is behind NAT, Asterisk must update >>> the contact to have the current IP and port that the phone registered via >>> (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state >>> for). >>> >>> On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan < >>> sonny.rajagopalan at gmail.com> wrote: >>> >>>> I am following the instructions in >>>> https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and >>>> I am trying to make a call from extension Alice (6001) to extension for Bob >>>> (6002). When I make the call, I can hear the ringing on Alice's phone >>>> (caller), but Bob's phone (callee) doesn't ring, or show a call coming in >>>> from Alice. My setup and environment is as follows: Alice, Bob and Asterisk >>>> all in the same 192.168.1.0/24 network, and they are able to register >>>> to the Asterisk server running 13.1.0/PJSIP. The rest of the configuration >>>> is the same as the aforementioned wiki page, but is shown here for clarity: >>>> >>>> root at asterisk13FFP:/var/log/asterisk# more >>>> /etc/asterisk/extensions.conf >>>> [from-internal] >>>> exten=>6001,1,Dial(PJSIP/demo-alice) >>>> exten=>6002,1,Dial(PJSIP/demo-bob) >>>> exten=>6003,1,Answer() >>>> same =>6003,n,Playback(hello-world) >>>> same =>6003,n,Hangup() >>>> >>>> >>>> What I do observe is that I when I request the output of pjsip show >>>> endpoints, I get Contact information for the two SIP peers that have >>>> registered different from their actual IP addresses. I suspect this has >>>> something to do with their calls being routed elsewhere. If my assumption >>>> is correct--how do I fix this? Alice should be at 192.168.1.50 and Bob >>>> should be at 192.168.1.149, instead, they (both) show IP address >>>> 146.115.163.234. Any help is deeply appreciated. Thanks. >>>> >>>> asterisk13FFP*CLI> pjsip show endpoints >>>> >>>> Endpoint: <Endpoint/CID.....................................> >>>> <State.....> <Channels.> >>>> I/OAuth: >>>> <AuthId/UserName...........................................................> >>>> Aor: <Aor............................................> >>>> <MaxContact> >>>> Contact: <Aor/ContactUri...............................> >>>> <Status....> <RTT(ms)..> >>>> Transport: <TransportId........> <Type> <cos> <tos> >>>> <BindAddress..................> >>>> Identify: >>>> <Identify/Endpoint.........................................................> >>>> Match: <ip/cidr.........................> >>>> Channel: <ChannelId......................................> >>>> <State.....> <Time(sec)> >>>> Exten: <DialedExten...........> CLCID: >>>> <ConnectedLineCID.......> >>>> >>>> ========================================================================================>>>> >>>> Endpoint: demo-alice >>>> Unavailable 0 of inf >>>> InAuth: demo-alice/demo-alice >>>> Aor: demo-alice 1 >>>> Contact: demo-alice/sip:demo-alice@*146.115.163.234*:38519 >>>> Unknown nan >>>> >>>> Endpoint: demo-bob Not in >>>> use 0 of inf >>>> InAuth: demo-bob/demo-bob >>>> Aor: demo-bob 1 >>>> Contact: demo-bob/sip:demo-bob@*146.115.163.234*:38321;tra >>>> Unknown nan >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >>>> >>> >>> >>> >>> -- >>> [image: Digium logo] >>> Scott Griepentrog >>> Digium, Inc ? Software Developer >>> 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US >>> direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090 >>> Check us out at: http://digium.com ? http://asterisk.org >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> > > -- > _____________________________________________________________________ > -- 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 >-- [image: Digium logo] Scott Griepentrog Digium, Inc ? Software Developer 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090 Check us out at: http://digium.com ? http://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... 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Sonny Rajagopalan
2015-Jan-09 14:47 UTC
[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
Solved! The issue was that RTP flows were being established to the wrong IP address. I figured out this issue--I had to disable STUN in both SIP phones for this to work correctly. Still, I wish a working configuration for Asterisk, and two SIP phones in the same 192.168.1.0/24 network would have helped me tremendously. On Thu, Jan 8, 2015 at 8:03 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote:> Well, I thought it worked, but it actually doesn't--I am able to get the > caller pick up the phone, but for some reason, I cannot hear anything on > either side no matter who does the calling. Again, my two SIP phones are on > the local 192.168.1.0/24 network (do not go over the Internet) and the > Asterisk server is located in the same network (not accessed over the > Internet). Any help is appreciated. > > Does the fact that Asterisk is running on a VirtualBox VM on the same > machine as one of the SIP phones matter? I am able to access the ARI REST > interface of the Asterisk server quite fine on the host machine. > > I suspect it has to do with RTP not being set up, but all the codec > support is there. Here's a log for the SIP request from 192.168.1.50: > > <--- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009 ---> > INVITE sip:6002 at 192.168.1.139;transport=UDP SIP/2.0 > Via: SIP/2.0/UDP 146.115.163.234:64009 > ;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- > Max-Forwards: 70 > Contact: <sip:demo-alice at 146.115.163.234:64009;transport=UDP> > To: <sip:6002 at 192.168.1.139;transport=UDP> > From: <sip:demo-alice at 192.168.1.139;transport=UDP>;tag=b661670b > Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. > CSeq: 2 INVITE > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, > SUBSCRIBE > Content-Type: application/sdp > Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri > User-Agent: Z 3.3.21933 r21903 > > Authorization: Digest > username="demo-alice",realm="asterisk",nonce="[removed]",uri=" > sip:6002 at 192.168.1.139 > ;transport=UDP",response="[removed]",cnonce="[removed]",nc=00000001,qop=auth,algorithm=md5,opaque="[removed]" > > Allow-Events: presence, kpml > Content-Length: 245 > > > v=0 > o=Z 0 0 IN IP4 146.115.163.234 > s=Z > c=IN IP4 146.115.163.234 > t=0 0 > m=audio 8000 RTP/AVP 0 3 110 8 98 101 > a=rtpmap:110 speex/8000 > a=rtpmap:98 iLBC/8000 > a=fmtp:98 mode=20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > > > <--- Transmitting SIP response (319 bytes) to UDP:192.168.1.50:64009 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 146.115.163.234:64009 > ;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- > Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. > From: <sip:demo-alice at 192.168.1.139>;tag=b661670b > To: <sip:6002 at 192.168.1.139> > CSeq: 2 INVITE > Content-Length: 0 > > Any help is appreciated. A topology is shown below in ASCII. > > > < ( Big bad Internet ) > > > GW/NAPT/Router > | > ---------------------------------------------------------- > / \ > > | | > Host A Host B > ----------------- > ----------------- > | Alice | | Bob > | > | 192.168.1.50 | | > 192.168.1.149 | > |---------------| > |---------------| > | Asterisk sr | > | (VM) | > | 192.168.1.239 | > |---------------| > > On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> Thank you for your note, Scott. >> >> I set rewrite_contact=yes for both contacts, and I also had to do >> remove_existing=yes because I had to remove the existing contact >> information (max_contacts = 1 was preventing new contact information) >> using pjsip qualify demo-alice etc., after which the right IP addresses >> showed in pjsip show endpoints. Anyway, it works as expected now, I >> think. My pjsip.conf is now >> >> [transport-udp] >> type=transport >> protocol=udp >> bind=0.0.0.0 >> local_net=192.168.1.0/24 >> ;Templates for the necessary config sections >> >> [endpoint_internal](!) >> type=endpoint >> context=from-internal >> disallow=all >> allow=ulaw >> >> [auth_userpass](!) >> type=auth >> auth_type=userpass >> >> [aor_dynamic](!) >> type=aor >> max_contacts=1 >> remove_existing=yes >> ;Definitions for our phones, using the templates above >> >> [demo-alice](endpoint_internal) >> auth=demo-alice >> aors=demo-alice >> mailboxes=box_a >> rewrite_contact=yes >> [demo-alice](auth_userpass) >> password=demo-alice ; put a strong, unique password here instead >> username=demo-alice >> >> [demo-alice](aor_dynamic) >> >> [demo-bob](endpoint_internal) >> auth=demo-bob >> aors=demo-bob >> mailboxes=box_b >> rewrite_contact=yes >> [demo-bob](auth_userpass) >> password=demo-bob ; put a strong, unique password here instead >> username=demo-bob >> >> [demo-bob](aor_dynamic) >> >> >> Thank you for your help! >> >> On Thu, Jan 8, 2015 at 11:48 AM, Scott Griepentrog < >> sgriepentrog at digium.com> wrote: >> >>> It would appear that you have the Asterisk server on a public IP >>> address, your two endpoints are behind a NAT, and you have rewrite_contact >>> enabled in pjsip.conf. >>> >>> In which case, what you are seeing is correct. In order to be able to >>> send a call to an extension where it is behind NAT, Asterisk must update >>> the contact to have the current IP and port that the phone registered via >>> (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state >>> for). >>> >>> On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan < >>> sonny.rajagopalan at gmail.com> wrote: >>> >>>> I am following the instructions in >>>> https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and >>>> I am trying to make a call from extension Alice (6001) to extension for Bob >>>> (6002). When I make the call, I can hear the ringing on Alice's phone >>>> (caller), but Bob's phone (callee) doesn't ring, or show a call coming in >>>> from Alice. My setup and environment is as follows: Alice, Bob and Asterisk >>>> all in the same 192.168.1.0/24 network, and they are able to register >>>> to the Asterisk server running 13.1.0/PJSIP. The rest of the configuration >>>> is the same as the aforementioned wiki page, but is shown here for clarity: >>>> >>>> root at asterisk13FFP:/var/log/asterisk# more >>>> /etc/asterisk/extensions.conf >>>> [from-internal] >>>> exten=>6001,1,Dial(PJSIP/demo-alice) >>>> exten=>6002,1,Dial(PJSIP/demo-bob) >>>> exten=>6003,1,Answer() >>>> same =>6003,n,Playback(hello-world) >>>> same =>6003,n,Hangup() >>>> >>>> >>>> What I do observe is that I when I request the output of pjsip show >>>> endpoints, I get Contact information for the two SIP peers that have >>>> registered different from their actual IP addresses. I suspect this has >>>> something to do with their calls being routed elsewhere. If my assumption >>>> is correct--how do I fix this? Alice should be at 192.168.1.50 and Bob >>>> should be at 192.168.1.149, instead, they (both) show IP address >>>> 146.115.163.234. Any help is deeply appreciated. Thanks. >>>> >>>> asterisk13FFP*CLI> pjsip show endpoints >>>> >>>> Endpoint: <Endpoint/CID.....................................> >>>> <State.....> <Channels.> >>>> I/OAuth: >>>> <AuthId/UserName...........................................................> >>>> Aor: <Aor............................................> >>>> <MaxContact> >>>> Contact: <Aor/ContactUri...............................> >>>> <Status....> <RTT(ms)..> >>>> Transport: <TransportId........> <Type> <cos> <tos> >>>> <BindAddress..................> >>>> Identify: >>>> <Identify/Endpoint.........................................................> >>>> Match: <ip/cidr.........................> >>>> Channel: <ChannelId......................................> >>>> <State.....> <Time(sec)> >>>> Exten: <DialedExten...........> CLCID: >>>> <ConnectedLineCID.......> >>>> >>>> ========================================================================================>>>> >>>> Endpoint: demo-alice >>>> Unavailable 0 of inf >>>> InAuth: demo-alice/demo-alice >>>> Aor: demo-alice 1 >>>> Contact: demo-alice/sip:demo-alice@*146.115.163.234*:38519 >>>> Unknown nan >>>> >>>> Endpoint: demo-bob Not in >>>> use 0 of inf >>>> InAuth: demo-bob/demo-bob >>>> Aor: demo-bob 1 >>>> Contact: demo-bob/sip:demo-bob@*146.115.163.234*:38321;tra >>>> Unknown nan >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >>>> >>> >>> >>> >>> -- >>> [image: Digium logo] >>> Scott Griepentrog >>> Digium, Inc ? Software Developer >>> 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US >>> direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090 >>> Check us out at: http://digium.com ? http://asterisk.org >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150109/21de1d84/attachment-0001.html>