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[asterisk-users] asterisk-users Digest, Vol 126, Issue 18 mtr

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Message

marlonfca at me.com
Guest

Posted: Wed Jan 21, 2015 8:39 am Post subject: [asterisk-users] asterisk-users Digest, Vol 126, Issue 18 mt

You could use MTR command.
Its a trace route improved.

Marlon Araujo

Quote:

On Jan 20, 2015, at 08:59, asterisk-users-request@lists.digium.com wrote:

Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
asterisk-users-request@lists.digium.com

You can reach the person managing the list at
asterisk-users-owner@lists.digium.com

When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-users digest..."

Today's Topics:

1. sip show channelstats reliable? (Todd R.)
2. Re: sip show channelstats reliable? (Todd R.)
3. Re: sip show channelstats reliable? (Eric Wieling)
4. Re: sip show channelstats reliable? (Todd R.)
5. Re: sip show channelstats reliable? (Scott Griepentrog)
6. Re: SEMI-OFFTOPIC openvox (ricky gutierrez)
7. Re: SEMI-OFFTOPIC openvox (A J Stiles)
8. Re: MWI issue (Haley,Scott A)

----------------------------------------------------------------------

Message: 1
Date: Mon, 19 Jan 2015 12:17:25 -0600
From: Todd R. <tjrlist@live.com>
To: Asterisk-Users List <asterisk-users@lists.digium.com>
Subject: [asterisk-users] sip show channelstats reliable?
Message-ID: <BLU173-W265CCDC9CB89501E36210ECD4A0@phx.gbl>
Content-Type: text/plain; charset="iso-8859-1"

I am seeing lots of lost packets when running the command sip show channelstats at the CLI.
There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable.
Can I trust the info this command shows?
I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the same IP's to/from.
Since I don't 100% control the network my gear is on, I need something outside of Asterisk to show the network engineer to convince here and myself that there are network issues.
All I have is the loss that's shown from this command with no real network stats to back it up.
Is there a magic command in CentOS anyone can recommend to diagnose and match up the issues shown in Asterisk using this command?
Moving gear around on the network changes the info Asterisk shows a LOT. For example, if I point traffic to the main physical gateway I get loss to a particular customer's IP (their PBX), if I move it to another place on the network (as a VM) their IP is good and other customers IP's start showing loss using the channelstats info.
Driving me freakin' crazy. It does appear there are network issues causing my troubles but I can't get help if I can't point to some hard and fast issues outside of Asterisk.
The only thing I have right now is collissions showing on one of a few of our pfSense devices but they are virtual running on XenServer, still this would indicate a problem in my opinion.
Thanks in advance for any assistance on this issue. Stepping back from the ledge now LOL

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Message: 2
Date: Mon, 19 Jan 2015 12:44:33 -0600
From: Todd R. <tjrlist@live.com>
To: Asterisk-Users List <asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] sip show channelstats reliable?
Message-ID: <BLU173-W470794F737AECEA2FCD353CD4A0@phx.gbl>
Content-Type: text/plain; charset="iso-8859-1"

Additional info:
At the moment I am running 1.8.x but the other day I was getting the same results on 11.x
Here is a sample from show channelstats. I do think this command is showing that there is trouble between specific IP's and my Asterisk box but I don't know if the numbers are accurate and reliable.

Peer
Call ID
Duration
Recv: Pack
Lost
( %)
Jitter
Send: Pack
Lost
(
%)
Jitter

x.x.x.x
5531341d06b
00:07:42
0000023123
0000063836
(73.41%)
0.0000
0000023102
0000000000
(
0.00%)
0.0007

Peer IP changed to protect the innocent Smile

From: tjrlist@live.com
To: asterisk-users@lists.digium.com
Date: Mon, 19 Jan 2015 12:17:25 -0600
Subject: [asterisk-users] sip show channelstats reliable?

I am seeing lots of lost packets when running the command sip show channelstats at the CLI.
There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable.
Can I trust the info this command shows?
I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the same IP's to/from.
Since I don't 100% control the network my gear is on, I need something outside of Asterisk to show the network engineer to convince here and myself that there are network issues.
All I have is the loss that's shown from this command with no real network stats to back it up.
Is there a magic command in CentOS anyone can recommend to diagnose and match up the issues shown in Asterisk using this command?
Moving gear around on the network changes the info Asterisk shows a LOT. For example, if I point traffic to the main physical gateway I get loss to a particular customer's IP (their PBX), if I move it to another place on the network (as a VM) their IP is good and other customers IP's start showing loss using the channelstats info.
Driving me freakin' crazy. It does appear there are network issues causing my troubles but I can't get help if I can't point to some hard and fast issues outside of Asterisk.
The only thing I have right now is collissions showing on one of a few of our pfSense devices but they are virtual running on XenServer, still this would indicate a problem in my opinion.
Thanks in advance for any assistance on this issue. Stepping back from the ledge now LOL

--
_____________________________________________________________________
-- 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:
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Message: 3
Date: Mon, 19 Jan 2015 13:55:33 -0500
From: Eric Wieling <EWieling@nyigc.com>
To: "tjrlist@live.com" <tjrlist@live.com>, Asterisk Users Mailing List
- Non-Commercial Discussion <asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] sip show channelstats reliable?
Message-ID:
<616B4ECE1290D441AD56124FEBB03D082F43F2E5E7@mailserver2007.nyigc.globe>

Content-Type: text/plain; charset="us-ascii"

I've seen something similar with Adtran SIP gateways. When a re-invite happens the Adtran gets all confused about call stats and marks the pre-reinvite leg of the call as losing large numbers of packets. BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38.

Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is very annoying. At some point I'll try and arrange a slugfest between Digium and Adtran and they can figure out why it doesn't work.

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Todd R.
Sent: Monday, January 19, 2015 1:45 PM
To: Asterisk-Users List
Subject: Re: [asterisk-users] sip show channelstats reliable?

Additional info:

At the moment I am running 1.8.x but the other day I was getting the same results on 11.x

Here is a sample from show channelstats. I do think this command is showing that there is trouble between specific IP's and my Asterisk box but I don't know if the numbers are accurate and reliable.

Peer

Call ID

Duration

Recv: Pack

Lost

( %)

Jitter

Send: Pack

Lost

(

%)

Jitter

x.x.x.x

5531341d06b

00:07:42

0000023123

0000063836

(73.41%)

0.0000

0000023102

0000000000

(

0.00%)

0.0007

Peer IP changed to protect the innocent Smile

________________________________
From: tjrlist@live.com<mailto:tjrlist@live.com>
To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Date: Mon, 19 Jan 2015 12:17:25 -0600
Subject: [asterisk-users] sip show channelstats reliable?
I am seeing lots of lost packets when running the command sip show channelstats at the CLI.

There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable.

Can I trust the info this command shows?

I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the same IP's to/from.

Since I don't 100% control the network my gear is on, I need something outside of Asterisk to show the network engineer to convince here and myself that there are network issues.

All I have is the loss that's shown from this command with no real network stats to back it up.

Is there a magic command in CentOS anyone can recommend to diagnose and match up the issues shown in Asterisk using this command?

Moving gear around on the network changes the info Asterisk shows a LOT. For example, if I point traffic to the main physical gateway I get loss to a particular customer's IP (their PBX), if I move it to another place on the network (as a VM) their IP is good and other customers IP's start showing loss using the channelstats info.

Driving me freakin' crazy. It does appear there are network issues causing my troubles but I can't get help if I can't point to some hard and fast issues outside of Asterisk.

The only thing I have right now is collissions showing on one of a few of our pfSense devices but they are virtual running on XenServer, still this would indicate a problem in my opinion.

Thanks in advance for any assistance on this issue. Stepping back from the ledge now LOL

-- _____________________________________________________________________ -- 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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Message: 4
Date: Mon, 19 Jan 2015 13:00:37 -0600
From: Todd R. <tjrlist@live.com>
To: Eric Wieling <ewieling@nyigc.com>, Asterisk-Users List
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] sip show channelstats reliable?
Message-ID: <BLU173-W166D750594A2E845C58840CD4A0@phx.gbl>
Content-Type: text/plain; charset="windows-1252"

Thanks but no Adtran here.
I do think these stats are indicating an issue, I just don't know how to prove it outside Asterisk.

From: EWieling@nyigc.com
To: tjrlist@live.com; asterisk-users@lists.digium.com
Date: Mon, 19 Jan 2015 13:55:33 -0500
Subject: RE: [asterisk-users] sip show channelstats reliable?

I?ve seen something similar with Adtran SIP gateways. When a re-invite happens the Adtran gets all confused about call stats and marks the pre-reinvite leg of the call as losing large numbers of packets. BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38. Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is very annoying. At some point I?ll try and arrange a slugfest between Digium and Adtran and they can figure out why it doesn?t work. From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Todd R.
Sent: Monday, January 19, 2015 1:45 PM
To: Asterisk-Users List
Subject: Re: [asterisk-users] sip show channelstats reliable? Additional info: At the moment I am running 1.8.x but the other day I was getting the same results on 11.x Here is a sample from show channelstats. I do think this command is showing that there is trouble between specific IP's and my Asterisk box but I don't know if the numbers are accurate and reliable. PeerCall IDDurationRecv: PackLost( %)JitterSend: PackLost(%)Jitterx.x.x.x5531341d06b00:07:4200000231230000063836(73.41%)0.000000000231020000000000(0.00%)0.0007 Peer IP changed to protect the innocent Smile

From: tjrlist@live.com
To: asterisk-users@lists.digium.com
Date: Mon, 19 Jan 2015 12:17:25 -0600
Subject: [asterisk-users] sip show channelstats reliable?I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable. Can I trust the info this command shows? I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the same IP's to/from. Since I don't 100% control the network my gear is on, I need something outside of Asterisk to show the network engineer to convince here and myself that there are network issues. All I have is the loss that's shown from this command with no real network stats to back it up. Is there a magic command in CentOS anyone can recommend to diagnose and match up the issues shown in Asterisk using this command? Moving gear around on the network changes the info Asterisk shows a LOT. For example, if I point traffic to the main
physical gateway I get loss to a particular customer's IP (their PBX), if I move it to another place on the network (as a VM) their IP is good and other customers IP's start showing loss using the channelstats info. Driving me freakin' crazy. It does appear there are network issues causing my troubles but I can't get help if I can't point to some hard and fast issues outside of Asterisk. The only thing I have right now is collissions showing on one of a few of our pfSense devices but they are virtual running on XenServer, still this would indicate a problem in my opinion. Thanks in advance for any assistance on this issue. Stepping back from the ledge now LOL
-- _____________________________________________________________________ -- 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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Message: 5
Date: Mon, 19 Jan 2015 13:13:01 -0600
From: Scott Griepentrog <sgriepentrog@digium.com>
To: tjrlist@live.com, Asterisk Users Mailing List - Non-Commercial
Discussion <asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] sip show channelstats reliable?
Message-ID:
<CACrpESbTXJXAuPLNdbBMTWUMyH4ksv_zRL0aSrM-QnjHrmOVUg@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

I would recommend capturing traffic outside your Asterisk server with
Wireshark, then running the Telephony/Rtp/Analysize Streams option to
determine if you have packet loss at that point in the network.

Quote:

On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist@live.com> wrote:

Thanks but no Adtran here.

I do think these stats are indicating an issue, I just don't know how to
prove it outside Asterisk.

------------------------------
From: EWieling@nyigc.com
To: tjrlist@live.com; asterisk-users@lists.digium.com
Date: Mon, 19 Jan 2015 13:55:33 -0500
Subject: RE: [asterisk-users] sip show channelstats reliable?

I?ve seen something similar with Adtran SIP gateways. When a re-invite
happens the Adtran gets all confused about call stats and marks the
pre-reinvite leg of the call as losing large numbers of packets. BTW,
IIRC reinvites happen when a codec changes or the channel switches to T.38.

Also Adtran SIP gateways appear not to support OPTIONS packets when
running in SIP proxy mode, which is very annoying. At some point I?ll
try and arrange a slugfest between Digium and Adtran and they can figure
out why it doesn?t work.

*From:* asterisk-users-bounces@lists.digium.com [mailto:
asterisk-users-bounces@lists.digium.com] *On Behalf Of *Todd R.
*Sent:* Monday, January 19, 2015 1:45 PM
*To:* Asterisk-Users List
*Subject:* Re: [asterisk-users] sip show channelstats reliable?

Additional info:

At the moment I am running 1.8.x but the other day I was getting the same
results on 11.x

Here is a sample from show channelstats. I do think this command is
showing that there is trouble between specific IP's and my Asterisk box but
I don't know if the numbers are accurate and reliable.

Peer

Call ID

Duration

Recv: Pack

Lost

( %)

Jitter

Send: Pack

Lost

(

%)

Jitter

x.x.x.x

5531341d06b

00:07:42

0000023123

0000063836

(73.41%)

0.0000

0000023102

0000000000

(

0.00%)

0.0007

Peer IP changed to protect the innocent Smile

------------------------------

From: tjrlist@live.com
To: asterisk-users@lists.digium.com
Date: Mon, 19 Jan 2015 12:17:25 -0600
Subject: [asterisk-users] sip show channelstats reliable?

I am seeing lots of lost packets when running the command sip show
channelstats at the CLI.

There are issues across multiple Asterisk servers I am trying to diagnose
but everything I read seems to point to this command being pretty
unreliable.

Can I trust the info this command shows?

I am showing lots of lost packets in sip show channelstats but I can't see
any packet loss when pinging the same IP's to/from.

Since I don't 100% control the network my gear is on, I need something
outside of Asterisk to show the network engineer to convince here and
myself that there are network issues.

All I have is the loss that's shown from this command with no real network
stats to back it up.

Is there a magic command in CentOS anyone can recommend to diagnose and
match up the issues shown in Asterisk using this command?

Moving gear around on the network changes the info Asterisk shows a LOT.
For example, if I point traffic to the main physical gateway I get loss to
a particular customer's IP (their PBX), if I move it to another place on
the network (as a VM) their IP is good and other customers IP's start
showing loss using the channelstats info.

Driving me freakin' crazy. It does appear there are network issues causing
my troubles but I can't get help if I can't point to some hard and fast
issues outside of Asterisk.

The only thing I have right now is collissions showing on one of a few of
our pfSense devices but they are virtual running on XenServer, still this
would indicate a problem in my opinion.

Thanks in advance for any assistance on this issue. Stepping back from the
ledge now LOL

-- _____________________________________________________________________
-- 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
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Message: 6
Date: Mon, 19 Jan 2015 14:37:34 -0600
From: ricky gutierrez <xserverlinux@gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] SEMI-OFFTOPIC openvox
Message-ID:
<CAL_GE3Q=bF6sngOsS=5dUEK5oe5pH3p7=R=nyN=buNqeAc5Nbg@mail.gmail.com>
Content-Type: text/plain; charset=UTF-8

Hi, when I make an outgoing call sends me a busy here, and no one is making call

Contact: <sip:984783842@50.X.X.X:5060>
Content-Length: 0

<------------>
-- Executing [984783842@to_pstn:1] Dial("SIP/101-0000004e",
"SIP/5001/84783842@,40,rRT") in new stack
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 13780
Video is at 50.X.X.X:18488
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding video codec 200003 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 190.53.38.203:5060:
INVITE sip:84783842%40@190.53.38.203 SIP/2.0
Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport
Max-Forwards: 70
From: "Operadora" <sip:101@50.X.X.X>;tag=as3708c762
To: <sip:84783842%40@190.53.38.203>
Contact: <sip:101@50.X.X.X:5060>
Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7@50.X.X.X:5060
CSeq: 102 INVITE
User-Agent: inmaconsa-Voice-Sip-ipbx
Date: Mon, 19 Jan 2015 20:17:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Operadora"
<sip:101@50.X.X.X>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 507

v=0
o=root 541548714 541548714 IN IP4 50.X.X.X
s=inamaconsa-Voice-Sip-pbx
c=IN IP4 50.X.X.X
b=CT:384
t=0 0
m=audio 13780 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 18488 RTP/AVP 99 98
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv

---
-- Called SIP/5001/84783842@

<--- Transmitting (NAT) to 190.X.X.1:41316 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316
From: "101" <sip:101@50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842@50.X.X.X>;tag=as77fb37e2
Call-ID: 7f55e32e-e4c6e11a@172.16.8.179
CSeq: 102 INVITE
Server: inmaconsa-Voice-Sip-ipbx
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:984783842@50.X.X.X:5060>
Content-Length: 0

<------------>

<--- SIP read from UDP:190.53.38.203:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
50.X.X.X:5060;branch=z9hG4bK374c2247;received=50.X.X.X;rport=5060
From: "Operadora" <sip:101@50.X.X.X>;tag=as3708c762
To: <sip:84783842%40@190.53.38.203>;tag=as4bb74f30
Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7@50.X.X.X:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 190.53.38.203:5060:
ACK sip:84783842%40@190.53.38.203 SIP/2.0
Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport
Max-Forwards: 70
From: "Operadora" <sip:101@50.X.X.X>;tag=as3708c762
To: <sip:84783842%40@190.53.38.203>;tag=as4bb74f30
Contact: <sip:101@50.X.X.X:5060>
Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7@50.X.X.X:5060
CSeq: 102 ACK
User-Agent: inmaconsa-Voice-Sip-ipbx
Content-Length: 0

---
[Jan 19 14:17:53] WARNING[11596][C-0000003d]: chan_sip.c:23037
handle_response_invite: Received response: "Forbidden" from
'"Operadora" <sip:101@50.X.X.X>;tag=as3708c762'
Scheduling destruction of SIP dialog
'0c9236b922c5a99f6a1a797c7c3f9eb7@50.X.X.X:5060' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [984783842@to_pstn:2] Busy("SIP/101-0000004e", "3")
in new stack

<--- Reliably Transmitting (NAT) to 190.X.X.1:41316 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP
190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316
From: "101" <sip:101@50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842@50.X.X.X>;tag=as77fb37e2
Call-ID: 7f55e32e-e4c6e11a@172.16.8.179
CSeq: 102 INVITE
Server: inmaconsa-Voice-Sip-ipbx
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0

<------------>
== Spawn extension (to_pstn, 984783842, 2) exited non-zero on
'SIP/101-0000004e'

<--- SIP read from UDP:190.X.X.1:41316 --->
ACK sip:984783842@50.X.X.X SIP/2.0
Via: SIP/2.0/UDP 190.X.X.1:41316;branch=z9hG4bK-61b74f36
From: "101" <sip:101@50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842@50.X.X.X>;tag=as30070ac7
Call-ID: 7f55e32e-e4c6e11a@172.16.8.179
CSeq: 101 ACK
Max-Forwards: 70
Contact: "101" <sip:101@190.X.X.1:41316>
User-Agent: Cisco/SPA508G-7.5.6
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Retransmitting #1 (NAT) to 190.X.X.1:41316:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP
190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316
From: "101" <sip:101@50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842@50.X.X.X>;tag=as77fb37e2
Call-ID: 7f55e32e-e4c6e11a@172.16.8.179
CSeq: 102 INVITE
Server: inmaconsa-Voice-Sip-ipbx
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0

2015-01-19 10:24 GMT-06:00 ricky gutierrez <xserverlinux@gmail.com>:

Quote:

Hi list, I write on the list looking for help, buy a openvox gw gsm
for four channels and I'm a little disappointed with the support
openvox, for some reason , The call doesn?t get trough

support tells me it was my asterisk server, but does not really work
me and my internal calls are working perfectly, I tested with another
sangoma FXO gateway and works perfectly.

the problem is that support openvox is Chinese and the difference in
time zone is high.

my trunk is connected

5001/5001 X.X.X.X D Yes
Yes 5060

Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline]

I follow this guide , but not work

http://www.lojamundi.com.br/download/gateways-gsm/openvox/Quickstart_Guide_of_OpenVox_GSM_Gateway_VS-GW2120_Series_Connect_with_Asterisk_Server.pdf

--
rickygm

http://gnuforever.homelinux.com

--
rickygm

http://gnuforever.homelinux.com

------------------------------

Message: 7
Date: Tue, 20 Jan 2015 09:39:58 +0000
From: A J Stiles <asterisk_list@earthshod.co.uk>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] SEMI-OFFTOPIC openvox
Message-ID: <201501200939.58525.asterisk_list@earthshod.co.uk>
Content-Type: Text/Plain; charset="utf-8"

Quote:

On Monday 19 Jan 2015, ricky gutierrez wrote:
Hi list, I write on the list looking for help, buy a openvox gw gsm
for four channels and I'm a little disappointed with the support
openvox, for some reason , The call doesn?t get trough

support tells me it was my asterisk server, but does not really work
me and my internal calls are working perfectly, I tested with another
sangoma FXO gateway and works perfectly.

the problem is that support openvox is Chinese and the difference in
time zone is high.

my trunk is connected

5001/5001 X.X.X.X D Yes
Yes 5060

Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline]

I follow this guide , but not work

http://www.lojamundi.com.br/download/gateways-gsm/openvox/Quickstart_Guide_
of_OpenVox_GSM_Gateway_VS-GW2120_Series_Connect_with_Asterisk_Server.pdf

I've had some experience with OpenVox GSM cards and chan_extra. Their support
isn't great; they like if you can give them ssh access to your box, and you
will need to ask questions afterwards to find out what they did in there, but
they did manage to sort out an obscure problem for me and explained enough for
me to work out what had been the matter in the first place.

As far as I can work out, their GSM gateway appliances seem to be some kind of
server motherboard with GSM cards and a pre-installed Linux, Asterisk and
chan_extra; but I've not had direct experience of them, having built my own
boxes using G400P and/or G400E cards in my favourite supplier's motherboards.

Oh, and finally, if you're using any kind of GSM gateway, be careful!
Otherwise, you will end up incurring the wrath of your telco -- "unlimited"
often does not really mean unlimited, and the only way to find out what the
limit actually is is to exceed it.

--
AJS

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

------------------------------

Message: 8
Date: Tue, 20 Jan 2015 13:59:36 +0000
From: "Haley,Scott A" <scott.haley@edwardjones.com>
To: "asterisk-users@lists.digium.com"
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] MWI issue
Message-ID: <E0917100-D825-4336-8EF0-6961913A3C20@edwardjones.com>
Content-Type: text/plain; charset="utf-8"

I have a situation that I need help with. I have 2 phone systems, 1 Asterisk and 1 Avaya. All voicemail is kept on the Avaya system. Whenever a call comes into an extension that the Asterisk server owns, I re-direct it to a different number that is owned by the Avaya System. If that Avaya extension does not answer it, I send it to the voicemail on the Avaya Messaging system for the extension that it came in on the Asterisk box.

Once that happens, I need to send a MWI indicator to an application on the desktop of the Avaya User that there is a voicemail for that mailbox.

I see the SIP Notify come in from Avaya for the extension (I did this with a tcpdump). My question is how do I configure Asterisk to act on that request and call an agi program to do what I want.

Any help would be appreciated.

Thanks,
Scott Haley

If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments.

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