* You are viewing Posts Tagged ‘avaya pbx’

Asterisk As A Media Gateway

I’m playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting.

This is what I was thinking of trying.

1. One asterisk server will contain the logic of the phone system (ex: queues, extensions…etc).

2. The mains server will not handle RTP traffic, it will send the RTP traffic to another system (another asterisk box?) for processing.

At the end of the day, what I am hoping for is to have 1 brain, and mutiple work horse audio gateways that can be added and removed as needed.

Has this been done? Can anyone point me to some documentation on how others have done this?

It’s always fun to play


Richard Seguin.

Transparent Elastix 2.2 fax receiving problem

Hi all,

First off all let me apologize for my English.

My company use an Avaya PBX with one E1 and about 250 extensions
(phones, faxes alarms, visa card readers etc).
My idea was to install a transparent asterisk between TELCO and PBX.

I use an OpenVox DE210E.
After install Elastix 2.2.0 and run dahdi_genconfig, i edit
/etc/asterisk/dahdi-channels.conf changing span 1 group from 0,11 to 0
and span 2 from 0,12 to 1.

Then from gui add all the active avaya extensions as custom extension
with dial DAHDI/i2/extension-number

The phones works fine but faxes open line; trying to synchronize (the
well known sound from faxes and modems). My home fax show the other
side number for about 90 sec and then display “comm. error” and hung up.

Any idea how to fix it ?

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

Asterisk refuses INVITE (401) and I don’t know why

Jonas,

May I suggest that you present us your sip.conf entry for this peer, properly redacted, of course. That might help more. What I do for “gateways” at known addresses is to put an
entry like this into the sip.conf entry:

[peer]
type=peer
defaultip=192.168.40.123
insecure=invite,port
context=some_context

On 11/22/2011 06:40 AM, Jonas Kellens wrote:
> Hello list,
>
> this is the communication between an Aastra 5000 PBX and Asterisk, where the Aastra makes a call to Asterisk. For some reason, Asterisk responds with 401-Unauthorized and I don’t
> know why.
>
> Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with this Aastra.
>
>
> A1.A1.A1.A1 = IP-address Asterisk PBX
> AS.AS.AS.AS = IP-address Aastra PBX
>
> Aastra PBX makes a call to the number 3221112233…
>
> This is all the sip debug trace gathered with asterisk :
>
>
> < --- SIP read from UDP:AS.AS.AS.AS:61490 --->
> INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport
> From: ;tag=310158BD
> To:
> Call-ID: 0201FFFFCEFEA742
> CSeq: 1 INVITE
> Contact:
> Proxy-Authorization: Digest username=”SIPPEERusername”, realm=”domain.tld”, nonce=”67105ac4″, uri=”sip:3221112233@A1.A1.A1.A1:5060″, response=”60be856773
> f86450fc9ddbaf7a568505″, algorithm=MD5
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE
> Max-Forwards: 70
> Privacy: none
> P-Asserted-Identity:
> User-Agent: A5000 R52-H2C0205
> P-Behind-Gsi: 192.168.6.1
> Content-Type: application/sdp
> Content-Length:195
>
> v=0
> o=- 0 0 IN IP4 sip.domain.tld
> s=-
> i=(o=IN IP4 10.1.2.35)
> c=IN IP4 AS.AS.AS.AS
> t=0 0
> m=audio 62654 RTP/AVP 8 0
> a=rtcp:65115
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=ptime:20
>
> < ------------->
>
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] — (16 headers 11 lines) —
> [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP TOS bits 184
> [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP CoS mark 5
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to AS.AS.AS.AS : 61490 (no NAT)
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE request as basis request – 0201FFFFCEFEA742
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer ‘SIPPEERusername’ for ‘SIPPEERusername’ from AS.AS.AS.AS:61490
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
> < --- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;received=AS.AS.AS.AS;rport=61490
> From: ;tag=310158BD
> To: ;tag=as68f71fe5
> Call-ID: 0201FFFFCEFEA742
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.6.2.20
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm=”domain.tld”, nonce=”46ef24d9″
> Content-Length: 0
>
> < ------------>
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling destruction of SIP dialog ‘0201FFFFCEFEA742′ in 32000 ms (Method: INVITE)
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
> < --- SIP read from UDP:AS.AS.AS.AS:61490 --->
> ACK sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport
> From: ;tag=310158BD
> To: ;tag=as68f71fe5
> Call-ID: 0201FFFFCEFEA742
> CSeq: 1 ACK
> Contact:
> Max-Forwards: 70
> User-Agent: A5000 R52-H2C0205
> P-Behind-Gsi: 192.168.6.1
> Content-Length: 0
>
>
> < ------------->
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] — (11 headers 0 lines) —
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
> < --- SIP read from UDP:AS.AS.AS.AS:61490 --->
> INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;rport
> From:
;tag=33015DBD
> To:
> Call-ID: 0201FFFFCCFEA242
> CSeq: 1 INVITE
> Contact:
> Proxy-Authorization: Digest username=”SIPPEERusername”, realm=”domain.tld”, nonce=”46ef24d9″, uri=”sip:3221112233@A1.A1.A1.A1:5060″, response=”14ecbfc7df24b49926151284c123ea11″,
> algorithm=MD5
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE
> Max-Forwards: 70
> Privacy: none
> P-Asserted-Identity:
> User-Agent: A5000 R52-H2C0205
> P-Behind-Gsi: 192.168.6.1
> Content-Type: application/sdp
> Content-Length:195
>
> v=0
> o=- 0 0 IN IP4 sip.domain.tld
> s=-
> i=(o=IN IP4 10.1.2.35)
> c=IN IP4 AS.AS.AS.AS
> t=0 0
> m=audio 62654 RTP/AVP 8 0
> a=rtcp:65115
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=ptime:20
>
>
> < ------------->
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] — (16 headers 11 lines) —
> [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP TOS bits 184
> [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP CoS mark 5
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to AS.AS.AS.AS : 61490 (no NAT)
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE request as basis request – 0201FFFFCCFEA242
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer ‘SIPPEERusername’ for ‘SIPPEERusername’ from AS.AS.AS.AS:61490
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
> < --- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;received=AS.AS.AS.AS;rport=61490
> From: ;tag=33015DBD
> To: ;tag=as1ba6ed56
> Call-ID: 0201FFFFCCFEA242
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.6.2.20
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm=”domain.tld”, nonce=”3df09f45″
> Content-Length: 0
>
>
> < ------------>
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling destruction of SIP dialog ‘0201FFFFCCFEA242′ in 32000 ms (Method: INVITE)
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
> < --- SIP read from UDP:AS.AS.AS.AS:61490 --->
> ACK sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;rport
> From: ;tag=33015DBD
> To: ;tag=as1ba6ed56
> Call-ID: 0201FFFFCCFEA242
> CSeq: 1 ACK
> Contact:
> Max-Forwards: 70
> User-Agent: A5000 R52-H2C0205
> P-Behind-Gsi: 192.168.6.1
> Content-Length: 0
>
>
> < ------------->
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] — (11 headers 0 lines) —
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
> < --- SIP read from UDP:AS.AS.AS.AS:61490 --->
> INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK847851481531358325;rport
> From:
;tag=340163BD
> To:
> Call-ID: 0201FFFFCBFE9C42
> CSeq: 1 INVITE
> Contact:
> Proxy-Authorization: Digest username=”SIPPEERusername”, realm=”domain.tld”, nonce=”3df09f45″, uri=”sip:3221112233@A1.A1.A1.A1:5060″, response=”80683cd640815b362f74afcfcb68809a”,
> algorithm=MD5
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE
> Max-Forwards: 70
> Privacy: none
> P-Asserted-Identity:
> User-Agent: A5000 R52-H2C0205
> P-Behind-Gsi: 192.168.6.1
> Content-Type: application/sdp
> Content-Length:195
>
> v=0
> o=- 0 0 IN IP4 sip.domain.tld
> s=-
> i=(o=IN IP4 10.1.2.35)
> c=IN IP4 AS.AS.AS.AS
> t=0 0
> m=audio 62654 RTP/AVP 8 0
> a=rtcp:65115
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=ptime:20
>
>
> < ------------->
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] — (16 headers 11 lines) —
> [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP TOS bits 184
> [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP CoS mark 5
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to AS.AS.AS.AS : 61490 (no NAT)
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE request as basis request – 0201FFFFCBFE9C42
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer ‘SIPPEERusername’ for ‘SIPPEERusername’ from AS.AS.AS.AS:61490
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
> < --- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK847851481531358325;received=AS.AS.AS.AS;rport=61490
> From: ;tag=340163BD
> To: ;tag=as26c6d395
> Call-ID: 0201FFFFCBFE9C42
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.6.2.20
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm=”domain.tld”, nonce=”6a7cfd54″
> Content-Length: 0
>
>
> < ------------>
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling destruction of SIP dialog ‘0201FFFFCBFE9C42′ in 32000 ms (Method: INVITE)
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
> < --- SIP read from UDP:AS.AS.AS.AS:61490 --->
> ACK sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK847851481531358325;rport
> From: ;tag=340163BD
> To: ;tag=as26c6d395
> Call-ID: 0201FFFFCBFE9C42
> CSeq: 1 ACK
> Contact:
> Max-Forwards: 70
> User-Agent: A5000 R52-H2C0205
> P-Behind-Gsi: 192.168.6.1
> Content-Length: 0
>
>
>
> Thanks.
>
> Kind regards,
> Jonas.
>
>
> –
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Avaya & Asterisk FreePBX Integration Problem

Hi,

I’m currently testing my FreePbx Box to work with our Avaya PBX to allow
dialing outgoing international call and FreePBX extensions to avaya PBX
Extensions calling.
Unfortunately no luck to do it successfully. Any help would be much be
appreciated, here is the sample codes I already tried:

On FreePBX GUI:
1. I created a custom Trunk called AvayaPBXTrunk with custom dial string
OOH323/$OUTNUM$/Avaya
2. Created an Outbound route called InternationalCall and select
AvayaPBXTrunk on the trunk sequence.
3. Created an Extension 1000 with dial extension OOH323/$OUTNUM$@Avaya

On Asterisk CLI:
1. Edit ooh323.conf with the following codes:
[general]
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
bindaddr=127.0.0.1
port=1720
callerID=”Asterisk PBX”
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
callerid=asterisk
context=default
disallow=all
allow=ulaw

[Avaya]
type=friend
context=from-internal
host=X.X.X.X ‘IP Address of our Avaya PBX
port=1720
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=inband
rtptimeout=60
e164=50

2. Edit sip_custom.conf with the following code:

[general]
context=from-internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=127.0.0.1
srvlookup=yes
canreinvite=no

Below also the log result during the call:

need to be able to pass a call to the pstn from another pbx trunk

pstn                                                           pstn

asterisk                 link between                 avaya pbx

both systems tied together by 2 pri’s
both have trunks out to the pstn
want to get rid of the avaya pstn trunk and send thru my asterisk box
avaya still has inbound calls on this trunk until late november (at&t
is dragging their feet doing the porting – 8 weeks between ) and still
has stations that we are not in a position to migrate to asterisk just
yet (about 500)
can get the call to show on the link between systems, but asterisk has
no station associate it with so it drops?
any suggestions out there?