[asterisk-users] Asterisk 13.13.1

2017-01-24 Thread Motty Cruz 
Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
starting to complaint about packets loss, conversations are choppy!


 

I don't even know where to start looking! Choppy conversations happened
within users. I am using sip.conf

 

[1091]

type=friend

context=sip-phone

call-limit=2

trustrpid=no

callerid="dev1" <1091>

disallow=all

allow=ulaw

allow=alaw

username=1091

secret=X

dtmfmode=rfc2833

host=dynamic

mailbox=10091@default

nat=force_rport,comedia

canreinvite=no

 

extensions.conf

exten => 1091,hint,SIP/${EXTEN}

exten => 1091,1,Dial(SIP/${EXTEN},15,t)

exten => 1091,2,Voicemail(${EXTEN}@default,u)

exten => 1091,102,Voicemail(${EXTEN}@default,b)

exten => 1091,103,Hangup

 

[2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: 

Retransmission timeout reached on transmission
7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

 

any ideas? 

 

Thanks!

Motty

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Re: [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

2017-01-24 Thread Joshua Colp 
On Tue, Jan 24, 2017, at 01:41 PM, Dan Cropp wrote:
> Thank you Joshua.
> 
> So there is no way to retrieve header information which may come in on
> subsequent packages?
> 
> If not, is there any way to make an Attended Transfer following the
> RFC5589?
> https://tools.ietf.org/html/rfc5589
> 
> Asking because we have a hospital with a Cisco switch.  Hospital has two
> calls from their Cisco switch into an Asterisk box.  Operator handling
> the two calls and needs to transfer Call A to be connected to call B. 
> Can obviously be patched inside of Asterisk.  However, the hospital wants
> the call to be Attended Transferred. Basically, we need to send the
> Transfer (REFER) with the Replaces containing the call ID, From tag, and
> the To Tag.
> 
> I am able to gather everything needed for the REFER field and pass that
> to the Transfer command (via AMI), except the To tag. 

There isn't that I can think of. Even if you are able to construct such
a REFER I'm not sure what exactly will happen inside of Asterisk with
the legs.

-- 
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] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

2017-01-24 Thread Dan Cropp 
Thank you Joshua.

So there is no way to retrieve header information which may come in on 
subsequent packages?

If not, is there any way to make an Attended Transfer following the RFC5589?
https://tools.ietf.org/html/rfc5589

Asking because we have a hospital with a Cisco switch.  Hospital has two calls 
from their Cisco switch into an Asterisk box.  Operator handling the two calls 
and needs to transfer Call A to be connected to call B.  Can obviously be 
patched inside of Asterisk.  However, the hospital wants the call to be 
Attended Transferred. Basically, we need to send the Transfer (REFER) with the 
Replaces containing the call ID, From tag, and the To Tag.

I am able to gather everything needed for the REFER field and pass that to the 
Transfer command (via AMI), except the To tag. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Tuesday, January 24, 2017 11:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to 
retrieve a PJSIP header To field for the SIP OK response to Trying?

On Tue, Jan 24, 2017, at 01:25 PM, Dan Cropp wrote:
> I place a call into Asterisk (from SIP phone) and the To header does 
> not have a tag.  Asterisk then sends it's Trying response, still no 
> tag in the To header.  The phone then replies with OK, this time the 
> To header includes a tag.
> 
> Is there any way to retrieve this response To header (including the 
> tag
> field) from the dial plan?
> I have tried the PJSIP-HEADER read of the To header, but it seems to 
> only have access to the initial To header.
> I even tried reading multiple layers of the To header, but it still 
> didn't retrieve the newer dialog To headers.

The PJSIP_HEADER dialplan function currently only looks at the initial message. 
It does not allow access to subsequent ones.

--
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] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

2017-01-24 Thread Joshua Colp 
On Tue, Jan 24, 2017, at 01:25 PM, Dan Cropp wrote:
> I place a call into Asterisk (from SIP phone) and the To header does not
> have a tag.  Asterisk then sends it's Trying response, still no tag in
> the To header.  The phone then replies with OK, this time the To header
> includes a tag.
> 
> Is there any way to retrieve this response To header (including the tag
> field) from the dial plan?
> I have tried the PJSIP-HEADER read of the To header, but it seems to only
> have access to the initial To header.
> I even tried reading multiple layers of the To header, but it still
> didn't retrieve the newer dialog To headers.

The PJSIP_HEADER dialplan function currently only looks at the initial
message. It does not allow access to subsequent ones.

-- 
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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[asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

2017-01-24 Thread Dan Cropp 
I place a call into Asterisk (from SIP phone) and the To header does not have a 
tag.  Asterisk then sends it's Trying response, still no tag in the To header.  
The phone then replies with OK, this time the To header includes a tag.

Is there any way to retrieve this response To header (including the tag field) 
from the dial plan?
I have tried the PJSIP-HEADER read of the To header, but it seems to only have 
access to the initial To header.
I even tried reading multiple layers of the To header, but it still didn't 
retrieve the newer dialog To headers.

I am including the SIP messages reported by Asterisk for the call coming in...

*** Phone sends INVITE to Asterisk ***

INVITE sip:3...@xxx.xxx.xxx.xxx SIP/2.0^M
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5063;branch=z9hG4bK-18e552c3^M
From: "1004" ;tag=79e7940882a792ao2^M
To: ^M
Call-ID: 3162d378-ea2b2...@yyy.yyy.yyy.yyy^M
CSeq: 102 INVITE^M
Max-Forwards: 70^M
Authorization: Digest 
username="1004",realm="asterisk",nonce="1485271992/b1bebde5cb4a763ed85b1d8e52c8e30d",uri="sip:3...@xxx.xxx.xxx.xxx",algorithm=MD5,response="8dd827e9910c2446fb0b8497f5944b81",opaque="66e52\
68a2111e777",qop=auth,nc=0001,cnonce="9dda9e0d"^M
Contact: "1004" ^M
Expires: 240^M
User-Agent: Cisco/SPA504G-7.4.8a^M
Content-Length: 401^M
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE^M
Supported: replaces^M
Content-Type: application/sdp^M
^M
v=0^M
o=- 32730859 32730859 IN IP4 yyy.yyy.yyy.yyy^M
s=-^M
c=IN IP4 yyy.yyy.yyy.yyy^M
t=0 0^M
m=audio 16436 RTP/AVP 0 2 8 9 18 96 97 98 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:2 G726-32/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:9 G722/8000^M
a=rtpmap:18 G729a/8000^M
a=rtpmap:96 G726-40/8000^M
a=rtpmap:97 G726-24/8000^M
a=rtpmap:98 G726-16/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-15^M
a=ptime:30^M
a=sendrecv^M

*** reply from Asterisk to phone ***

SIP/2.0 100 Trying^M
Via: SIP/2.0/UDP 
yyy.yyy.yyy.yyy:5063;received=yyy.yyy.yyy.yyy;branch=z9hG4bK-18e552c3^M
Call-ID: 3162d378-ea2b2...@yyy.yyy.yyy.yyy^M
From: "1004" ;tag=79e7940882a792ao2^M
To: ^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 14.2.1^M
Content-Length:  0^M
^M


**
Asterisk receives this packet in response to the Trying.
Is it possible to retrieve this To header via the dial plan?  Specifically, I 
need the tag portion of the From
**

SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 
yyy.yyy.yyy.yyy:5063;received=yyy.yyy.yyy.yyy;branch=z9hG4bK-18e552c3^M
Call-ID: 3162d378-ea2b2...@yyy.yyy.yyy.yyy^M
From: "1004" ;tag=79e7940882a792ao2^M
To: ;tag=96156bd7-9e8e-4077-b6e4-f3eb12e39069^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 14.2.1^M
Contact: ^M
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, 
PRACK, REGISTER, MESSAGE, REFER^M
Supported: 100rel, timer, replaces, norefersub^M
Content-Type: application/sdp^M
Content-Length:   179^M
^M
v=0^M
o=- 32730859 32730861 IN IP4 xxx.xxx.xxx.xxx^M
s=Asterisk^M
c=IN IP4 xxx.xxx.xxx.xxx^M
t=0 0^M
m=audio 19384 RTP/AVP 0^M
a=rtpmap:0 PCMU/8000^M
a=ptime:20^M
a=maxptime:150^M
a=sendrecv^M


ACK sip:xxx.xxx.xxx.xxx:5060 SIP/2.0^M
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5063;branch=z9hG4bK-c38362b^M
From: "1004" ;tag=79e7940882a792ao2^M
To: ;tag=96156bd7-9e8e-4077-b6e4-f3eb12e39069^M
Call-ID: 3162d378-ea2b2...@yyy.yyy.yyy.yyy^M
CSeq: 102 ACK^M
Max-Forwards: 70^M
Authorization: Digest 
username="1004",realm="asterisk",nonce="1485271992/b1bebde5cb4a763ed85b1d8e52c8e30d",uri="sip:3...@xxx.xxx.xxx.xxx",algorithm=MD5,response="8dd827e9910c2446fb0b8497f5944b81",opaque="66e52\
68a2111e777",qop=auth,nc=0001,cnonce="9dda9e0d"^M
Contact: "1004" ^M
User-Agent: Cisco/SPA504G-7.4.8a^M
Content-Length: 0^M
^M

SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 
192.168.35.91:5063;received=192.168.35.91;branch=z9hG4bK-18e552c3^M
Call-ID: 3162d378-ea2b2452@192.168.35.91^M
From: "1004" ;tag=79e7940882a792ao2^M
To: ;tag=96156bd7-9e8e-4077-b6e4-f3eb12e39069^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 14.2.1^M
Contact: ^M
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, 
PRACK, REGISTER, MESSAGE, REFER^M
Supported: 100rel, timer, replaces, norefersub^M
Content-Type: application/sdp^M
Content-Length:   179^M
^M
v=0^M
o=- 32730859 32730861 IN IP4 192.168.33.30^M
s=Asterisk^M
c=IN IP4 192.168.33.30^M
t=0 0^M
m=audio 19384 RTP/AVP 0^M
a=rtpmap:0 PCMU/8000^M
a=ptime:20^M
a=maxptime:150^M
a=sendrecv^M
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Re: [asterisk-users] Setup DID

2017-01-24 Thread A J Stiles 
On Tuesday 24 Jan 2017, Zakir Mahomedy wrote:
> Hi I am trying to setup DDI for one of our servers
> Our Provider has given us one DDI for use for eg 080011.
> On my main server  A,  I use an IAX trunk to connect to Client Server
> B.When calls come in from the outside world on main server A for
> 080011In the dial plan, I pattern match and connect with IAX2 truck
> named 087XX eg) SERVER A DIALPLANexten => 080011,1, Verbose( 3, "
>  INCOMING CALLS  SERVER B )same => n,Dial(IAX2/087XX,,r) On
> server B I have an incoming context in which I have both a general IVR and
> the 080011 patternto route the DDI number to a particular extension.
> [incoming]
> exten =>080011.,1,Answer()same => n,Dial(PJSIP/200)same => n,Hangup()
> exten => s,1,Answer()same => n,Goto(main_ivr,,1)same => n,Hangup()
> I cant seem to get a match on 087, it always go to the s context in
> incomingAny ideas on how I can get DDI to work thanks Zakir

This probably is related to the format in which one server is presenting the 
extension to the other server, and your extension definition(s) not matching 
that format.  (For instance, something could be stripping the leading zero 
from the STD code.)  So you need to find out exactly what the far end is 
sending the number to you like, so you can make sure your extensions match.
 
In the "s" extension in your [incoming] context, put a line like
 
exten => s,n,NoOp(Dialled extension is "${EXTEN}")
 
Dial through to the first server, and watch the console on the second server to 
make sure the call comes through.  Note the format in which the extension is 
presented, and make sure your extension definitions match that.
 
 
Also, do not forget, "wildcard" extensions which can match more than one thing 
must begin with an underscore; e.g.
 
exten => _3xx,1,NoOp(Call to "${EXTEN}" on B site)
exten => _3xx,n,Dial(${BSITE}/${EXTEN})
exten => _3xx,n,Hangup()
 
Even although there be characters in the extension name which will mark it out 
as an obvious wildcard, Asterisk still expects for there to be an _ at the 
beginning in order to treat it as such.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] Setup DID

2017-01-24 Thread Feroz Ahmed 
You would need to explicitly defined the registration string / DDI, else it
will always take the s precedence.

Thanks,
Feroz Ahmed

On Tue, Jan 24, 2017 at 5:16 PM, Zakir Mahomedy  wrote:

> Hi I am trying to setup DDI for one of our servers
>
> Our Provider has given us one DDI for use for eg 080011.
>
> On my main server  A,  I use an IAX trunk to connect to Client Server B.
> When calls come in from the outside world on main server A for 080011
> In the dial plan, I pattern match and connect with IAX2 truck named
> 087XX
>
> eg) SERVER A DIALPLAN
> exten => 080011,1, Verbose( 3, "  INCOMING CALLS  SERVER B )
> same => n,Dial(IAX2/087XX,,r)
>
> On server B
> I have an incoming context in which I have both a general IVR and the
> 080011 pattern
> to route the DDI number to a particular extension.
>
> [incoming]
>
> exten =>080011.,1,Answer()
> same => n,Dial(PJSIP/200)
> same => n,Hangup()
>
> exten => s,1,Answer()
> same => n,Goto(main_ivr,,1)
> same => n,Hangup()
>
> I cant seem to get a match on 087, it always go to the s context in
> incoming
> Any ideas on how I can get DDI to work
> thanks
>
> Zakir
>
>
>
> --
> _
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>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Setup DID

2017-01-24 Thread Zakir Mahomedy 
Hi I am trying to setup DDI for one of our servers
Our Provider has given us one DDI for use for eg 080011.
On my main server  A,  I use an IAX trunk to connect to Client Server B.When 
calls come in from the outside world on main server A for 080011In the dial 
plan, I pattern match and connect with IAX2 truck named 087XX
eg) SERVER A DIALPLANexten => 080011,1, Verbose( 3, "  INCOMING CALLS  
SERVER B )same => n,Dial(IAX2/087XX,,r)      
On server B I have an incoming context in which I have both a general IVR and 
the 080011 patternto route the DDI number to a particular extension.
[incoming]
exten =>080011.,1,Answer()same => n,Dial(PJSIP/200)same => n,Hangup()
exten => s,1,Answer()same => n,Goto(main_ivr,,1)same => n,Hangup()
I cant seem to get a match on 087, it always go to the s context in incomingAny 
ideas on how I can get DDI to work thanks
Zakir

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Re: [asterisk-users] Understanding how LLDP works with DHCP [SOLVED]

2017-01-24 Thread Olivier 
2017-01-19 18:19 GMT+01:00 Jose Flores Galicia :

> 2017-01-19 4:09 GMT-06:00 Olivier :
>
>> Hello,
>>
>> For years, I used to configure SIP phone VLAN membership through a DHCP
>> server.
>>
>> Here are the details:
>> - I dedicate a LAN port on a switch to voice VLAN
>> - somewhere else, I configure a DHCP server to serve LAN addresses within
>> voice VLAN
>> - any other switch port connected to an other DHCP server is explicitely
>> excluded from voice VLAN
>> - new SIP hardphones are first connected to the dedicated voice VLAN
>> port: after several reboots, they get an address within voice VLAN address
>> range and save VLAN tag somewhere within their persistent memory
>> - SIP phones are then moved to an other switch port: as they boots, they
>> request a LAN address using previously received VLAN tag.
>>
>> Now I would like to improve this process using LLDP.
>> I ran a couple of tests in my lab and still have some questions:
>>
>>
>> 1. My lab switch sends within LLDP frames, a list of VLANs. One is named
>> "default" and the other is named "voice".
>> Do LLDP-capable phones look for a specific name to elect the VLAN tag
>> they will later use to build DHCPDISCOVER request or do they look for
>> something else (medPolicy) ?
>>
>> 2. With LLDP, do you still need your DHCP server to embed VLAN membership
>> data within DHCPOFFER or is it a thing of the past ?
>>
>> 3. Have you been successfull with LLDP on a KVM guest networked to an
>> LLDP-enabled switch through a linux bridge (see [1]) ?
>> Where can I find information regarding the line bellow:
>> echo 16384 > /sys/class/net//bridge/group_fwd_mask
>>
>>
>> [1] https://thenetworkway.wordpress.com/2016/01/04/lldp-traffic-
>> and-linux-bridges/
>>
>>
>> Best regards
>>
>>
>> --
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>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> Hi Oliver.
>
> 1. Actually there are 2 protocols which must be supported on switch and ip
> phone, one is LLDP wich inventories both ways. IP Phone <-> Switch, to
> ellaborate on both devices MIB database with switching/routing/app
> capabilities of their partner; second is the LLDP-MED (Media Endpoint
> Discovery) which is capable of sending L2/L3 settings to devices. These
> settings are in several categories, concerning your question is "Network
> Policy" settings wich will be sent to the ip phone based on their app
> capabilities (LLDP).
> "Network Policy" settings can contain VLAN ID for voice and other ID for
> Data, and other for Video, etc. Once the LLDP-MED "Network Policy" settings
> are received on ip phone, will tag the phone traffic on the specified VLAN,
> On switching capable ip phones (2 or more ethernet interfaces), probably
> will only tag phone traffic, and leave the switched traffic on the access
> vlan.
> So ip phone sends DHCPREQUEST on the VLAN ID set by LLDP-MED.
>
> 2. DHCP VLAN settings will probably being ignored since most of devices
> will prefer LLDP-MED settings.
>
> 3. With LLDP enabled on KVM guest, you can obtain information about
> network devices attached, their capabilities, brand, model, etc. I have
> never tried but LLDP-MED supposed only to work on next switch device (Link
> Layer), not a propagation protocol.
>
>
> I always have deployed LLDP-MED capable ip phones on a LLDP-MED capable
> network is:
> 1.- Enable LLDP on all access switches so they can advertise and receive
> LLDP information.
> 2.- Configure LLDP-MED on all ports where will be connected ip phones and
> set the correct Network Policy, sending LLDP-MED capable voice devices to
> voice VLAN ID
> 3.- DHCP server exists attached to voice VLAN ID, as LLDP-MED will provide
> same VLAN ID, phone will receive at the first boot their DHCP settings.
>
> Best Regards.
>
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Hello,

Combining a Yealink phone with a Ubiquiti switch, and thanks to above
explanation,I could at last have my very first working LLDP experiment !

To sum up things: it seems an LLDP-enabled switch port would send a frame
including an LLDP policy. This LLDP policy (which can be specific to each
port) describe which tag to use to "enter" Voice VLAN.

In turn LLDP-enabled