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Pavlidis Savas

2004-Oct-30 01:47 UTC

[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk

I have a peculiar problem.
I have installed asterisk
and also g729 (2 channels).
I have a Cisco7940 IP phone
with SIP installed (v6)
and a cisco router 2650xm
which has an isdn bri voice
interface that connects to
a legacy pbx system. Also
I installed a x-lite
to make some tests.

I have configured everything
after a lot of search and
trial and error. So I have
managed to make calls from the
7940 to x-lite and vice-versa
and also to make calls to
to legacy phones from the
7940 or the x-lite via the
cisco router using its voice
interface.
BUT the problem is that from
the legacy PBX phones I can call
the x-lite but not the cisco
7940 IP Phone.
Where is the problem????
Can anyone help me?

here are the configurations:
SIP.CONF
[general]
context=default                 ; Default context for incoming calls
port=5060                       ; UDP Port to bind to (SIP standard port 
is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds 
to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g729


[xlite1]
type=friend
regexten=1239                 ; When they register, create extension 1239
username=xlite1
callerid="Savas Pavlidis" <1239>
host=dynamic
;nat=yes                       ; X-Lite is behind a NAT router
canreinvite=no                ; Typically set to NO if behind NAT

[10.1.1.1]                      ; Cisco 2650XM router
type=friend
host=10.1.1.1
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=g729

[419]                           ; 7940G Cisco IP Phone
type=friend
username=419
host=dynamic
canreinvite=yes
dtmfmode=inband
disallow=all
allow=g729


EXTENSIONS.CONF (PART OF IT)
; The numbers 3XX belong to the traditional
; PBX telephones.
;
exten => _3XX,1,Dial(SIP/${EXTEN}@10.1.1.1)
exten => _3XX,n,Congestion

;
;
;
exten => 419,1,Dial(SIP/419)

exten => 420,1,Dial(SIP/xlite1)
exten => 420,2,Congestion


; as you may understand 419 is the cisco ip phone
; and extension 420 is the softp phone x-lite
; on the pc.


CISCO ROUTER CONFIGURATION (PART OF IT)
dial-peer voice 1 pots
 destination-pattern 3..
 direct-inward-dial
 port 1/0/0
 forward-digits all
!
dial-peer voice 2 pots
 destination-pattern 3..
 direct-inward-dial
 port 1/0/1
 forward-digits all
!
!
dial-peer voice 100 voip
 destination-pattern 9..
 session target ipv4:100.0.0.1
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs5 media
!
dial-peer voice 101 voip
 destination-pattern 8..
 session target ipv4:100.0.0.1
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs5 media
!
dial-peer voice 103 voip
 destination-pattern 1..
 session target ipv4:200.200.201.2
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs5 media
!
dial-peer voice 200 voip
 destination-pattern 40.
 session target ipv4:100.0.0.191
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs5 media
!
dial-peer voice 201 voip
 destination-pattern 5..
 session target ipv4:100.0.0.191
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs5 media
!
dial-peer voice 202 voip
 destination-pattern 42.
 session protocol sipv2
 session target sip-server
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
dial-peer voice 205 voip
 destination-pattern 41.
 session protocol sipv2
 session target sip-server
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
sip-ua
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 timers trying 1000
 sip-server ipv4:10.1.1.250:5060
!

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Pavlidis Savas

2004-Nov-01 02:30 UTC

head link

[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk

I have a peculiar problem.
I have installed asterisk
and also g729 (2 channels).
I have a Cisco7940 IP phone
with SIP installed (v6)
and a cisco router 2650xm
which has an isdn bri voice
interface that connects to
a legacy pbx system. Also
I installed a x-lite
to make some tests.

I have configured everything
after a lot of search and
trial and error. So I have
managed to make calls from the
7940 to x-lite and vice-versa
and also to make calls to
to legacy phones from the
7940 or the x-lite via the
cisco router using its voice
interface.
BUT the problem is that from
the legacy PBX phones I can call
the x-lite but not the cisco
7940 IP Phone. I place the call
and the cisco phone rings just once
(and it shows for a fraction of
second the caller id) and then
the connections closes as if
the called has hunged up.
Where is the problem????
Can anyone help me?

here are the configurations:
SIP.CONF
[general]
context=default                 ; Default context for incoming calls
port=5060                       ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g729


[xlite1]
type=friend
regexten=1239                 ; When they register, create extension 1239
username=xlite1
callerid="Savas Pavlidis" <1239>
host=dynamic
;nat=yes                       ; X-Lite is behind a NAT router
canreinvite=no                ; Typically set to NO if behind NAT

[10.1.1.1]                      ; Cisco 2650XM router
type=friend
host=10.1.1.1
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=g729

[419]                           ; 7940G Cisco IP Phone
type=friend
username=419
host=dynamic
canreinvite=yes
dtmfmode=inband
disallow=all
allow=g729


EXTENSIONS.CONF (PART OF IT)
; The numbers 3XX belong to the traditional
; PBX telephones.
;
exten => _3XX,1,Dial(SIP/${EXTEN}@10.1.1.1)
exten => _3XX,n,Congestion

;
;
;
exten => 419,1,Dial(SIP/419)

exten => 420,1,Dial(SIP/xlite1)
exten => 420,2,Congestion


; as you may understand 419 is the cisco ip phone
; and extension 420 is the softp phone x-lite
; on the pc.


CISCO ROUTER CONFIGURATION (PART OF IT)
dial-peer voice 1 pots
  destination-pattern 3..
  direct-inward-dial
  port 1/0/0
  forward-digits all
!
dial-peer voice 2 pots
  destination-pattern 3..
  direct-inward-dial
  port 1/0/1
  forward-digits all
!
!
dial-peer voice 100 voip
  destination-pattern 9..
  session target ipv4:100.0.0.1
  dtmf-relay h245-signal h245-alphanumeric
  ip qos dscp cs5 media
!
dial-peer voice 101 voip
  destination-pattern 8..
  session target ipv4:100.0.0.1
  dtmf-relay h245-signal h245-alphanumeric
  ip qos dscp cs5 media
!
dial-peer voice 103 voip
  destination-pattern 1..
  session target ipv4:200.200.201.2
  dtmf-relay h245-signal h245-alphanumeric
  ip qos dscp cs5 media
!
dial-peer voice 200 voip
  destination-pattern 40.
  session target ipv4:100.0.0.191
  dtmf-relay h245-signal h245-alphanumeric
  ip qos dscp cs5 media
!
dial-peer voice 201 voip
  destination-pattern 5..
  session target ipv4:100.0.0.191
  dtmf-relay h245-signal h245-alphanumeric
  ip qos dscp cs5 media
!
dial-peer voice 202 voip
  destination-pattern 42.
  session protocol sipv2
  session target sip-server
  session transport udp
  dtmf-relay rtp-nte
  codec g711alaw
  no vad
!
dial-peer voice 205 voip
  destination-pattern 41.
  session protocol sipv2
  session target sip-server
  session transport udp
  dtmf-relay rtp-nte
  codec g711alaw
  no vad
!
sip-ua
  retry invite 3
  retry response 3
  retry bye 3
  retry cancel 3
  timers trying 1000
  sip-server ipv4:10.1.1.250:5060
!

This is the SIP transaction
Sip read:
INVITE sip:419@10.1.1.250:5060 SIP/2.0
Via: SIP/2.0/UDP  10.1.1.1:5060
From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5
To: <sip:419@10.1.1.250>
Date: Mon, 01 Nov 2004 08:12:53 GMT
Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 2846660713-722735577-3128754082-2282147162
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, 
SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: <sip:319@10.1.1.1>;party=calling;screen=no;privacy=off
Timestamp: 1099296773
Contact: <sip:319@10.1.1.1:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 235

v=0
o=CiscoSystemsSIP-GW-UserAgent 1213 9211 IN IP4 10.1.1.1
s=SIP Call
c=IN IP4 10.1.1.1
t=0 0
m=audio 18644 RTP/AVP 8 101
c=IN IP4 10.1.1.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

20 headers, 11 lines
Using latest request as basis request
Sending to 10.1.1.1 : 5060 (non-NAT)
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.1.1.1:18644
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x10e(GSM|ULAW|ALAW|G729A), peer - 
audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 
0x1(G723)
Found no matching peer or user for '10.1.1.1:56528'
Looking for 419 in default
list_route: hop: <sip:319@10.1.1.1:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP  10.1.1.1:5060
From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5
To: <sip:419@10.1.1.250>;tag=as4996755e
Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:419@10.1.1.250>
Content-Length: 0


  to 10.1.1.1:5060
We're at 10.1.1.250 port 16084
Answering with preferred capability 0x100(G729A)
Answering with non-codec capability 0x1(G723)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:419@10.1.1.18:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f
From: "319" <sip:319@10.1.1.250>;tag=as339f0f84
To: <sip:419@10.1.1.18:5060;user=phone>
Contact: <sip:319@10.1.1.250>
Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 01 Nov 2004 08:22:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 28878 28878 IN IP4 10.1.1.250
s=session
c=IN IP4 10.1.1.250
t=0 0
m=audio 16084 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
  (no NAT) to 10.1.1.18:5060


Sip read:
CANCEL sip:419@10.1.1.250:5060 SIP/2.0
Via: SIP/2.0/UDP  10.1.1.1:5060
From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5
To: <sip:419@10.1.1.250>
Date: Mon, 01 Nov 2004 08:12:53 GMT
Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1
CSeq: 101 CANCEL
Max-Forwards: 6
Timestamp: 1099296773
Content-Length: 0


10 headers, 0 lines
Sending to 10.1.1.1 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP  10.1.1.1:5060
From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5
To: <sip:419@10.1.1.250>;tag=as4996755e
Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:419@10.1.1.250>
Content-Length: 0


  to 10.1.1.1:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP  10.1.1.1:5060
From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5
To: <sip:419@10.1.1.250>;tag=as4996755e
Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:419@10.1.1.250>
Content-Length: 0


  to 10.1.1.1:5060
Reliably Transmitting:
CANCEL sip:419@10.1.1.18:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f
From: "319" <sip:319@10.1.1.250>;tag=as339f0f84
To: <sip:419@10.1.1.18:5060;user=phone>
Contact: <sip:319@10.1.1.250>
Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

  (no NAT) to 10.1.1.18:5060
Scheduling destruction of call 
'05785b314473c336521dbfa40a4b8e7e@10.1.1.250' in 15000 ms


Sip read:
ACK sip:419@10.1.1.250:5060 SIP/2.0
Via: SIP/2.0/UDP  10.1.1.1:5060
From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5
To: <sip:419@10.1.1.250>;tag=as4996755e
Date: Mon, 01 Nov 2004 08:12:53 GMT
Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK


9 headers, 0 lines
Destroying call 'A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1'


Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f
From: "319" <sip:319@10.1.1.250>;tag=as339f0f84
To: <sip:419@10.1.1.18:5060;user=phone>
Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250
Date: Mon, 01 Nov 2004 08:12:52 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:419@10.1.1.18:5060>
Content-Length: 0


10 headers, 0 lines


Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f
From: "319" <sip:319@10.1.1.250>;tag=as339f0f84
To: 
<sip:419@10.1.1.18:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e
Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250
Date: Mon, 01 Nov 2004 08:12:52 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:419@10.1.1.18:5060>
Content-Length: 0


10 headers, 0 lines


Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f
From: "319" <sip:319@10.1.1.250>;tag=as339f0f84
To: 
<sip:419@10.1.1.18:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e
Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250
Date: Mon, 01 Nov 2004 08:12:52 GMT
CSeq: 102 CANCEL
Server: CSCO/6
Content-Length: 0


9 headers, 0 lines


Sip read:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f
From: "319" <sip:319@10.1.1.250>;tag=as339f0f84
To: 
<sip:419@10.1.1.18:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e
Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250
Date: Mon, 01 Nov 2004 08:12:52 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:419@10.1.1.18:5060>
Content-Length: 0


10 headers, 0 lines
Transmitting:
ACK sip:419@10.1.1.18:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f
From: "319" <sip:319@10.1.1.250>;tag=as339f0f84
To: 
<sip:419@10.1.1.18:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e
Contact: <sip:319@10.1.1.250>
Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

  (no NAT) to 10.1.1.18:5060
Destroying call '05785b314473c336521dbfa40a4b8e7e@10.1.1.250'
sip no debug
SIP Debugging Disabled



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Pavlidis Savas

2004-Nov-01 05:38 UTC

head link

[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk

Propably this is a cisco router issue.
I discovered that if a put
play line in the extensions.conf
so that it can play something before
the call is done, even
for one second, the call is working
normally.

I also played with other variables
in the sip.conf but have not succeeded
except with the play line in extensions.conf



exten => 419,1,Playback(pbx-transfer)
exten => 419,2,Dial(SIP/419)

Pavlidis Savas

2004-Nov-01 07:14 UTC

head link

[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk

Thanks for your reply.
The cisco router is 2650xm
with the following
==================================================================Cisco
Internetwork Operating System Software
IOS (tm) C2600 Software (C2600-IS-M), Version 12.2(15)ZJ3, EARLY
DEPLOYMENT RELE
ASE SOFTWARE (fc2)
TAC Support: http://www.cisco.com/tac
Copyright (c) 1986-2003 by cisco Systems, Inc.
Compiled Fri 26-Sep-03 02:05 by eaarmas
Image text-base: 0x80008098, data-base: 0x81ADBC0C

ROM: System Bootstrap, Version 12.2(8r) [cmong 8r], RELEASE SOFTWARE (fc1)
ROM: C2600 Software (C2600-IS-M), Version 12.2(15)ZJ3, EARLY DEPLOYMENT
RELEASE
SOFTWARE (fc2)

Yalko-Thess uptime is 11 weeks, 3 hours, 1 minute
System returned to ROM by reload
System restarted at 10:52:42 UTC Mon Aug 16 2004
System image file is "flash:c2600-is-mz.122-15.ZJ3.bin"

cisco 2650XM (MPC860P) processor (revision 0x200) with 125952K/5120K
bytes of me
mory.
Processor board ID JAE080102SE (2906160658)
M860 processor: part number 5, mask 2
Bridging software.
X.25 software, Version 3.0.0.
Basic Rate ISDN software, Version 1.1.
1 FastEthernet/IEEE 802.3 interface(s)
1 Serial network interface(s)
3 ISDN Basic Rate interface(s)
32K bytes of non-volatile configuration memory.
32768K bytes of processor board System flash (Read/Write)

Configuration register is 0x2102
=================================================================it has a VIC
2BRI-ISDN interface  (NT/TE)
which connects to the PBX which is
a Bosch  Tenovis Integral 33EW2 EuroISDN.
We use already another similar cisco at another
office via a WAN link to transfer data and voice
channels between the two PBX's (same).
I made some changes to experiment with
SIP as the hardware already exists.

I somewhere red that my version of Cisco IOS
cannot handle the INVITE command according
to RFC's.

You can see a previous mail with the same
header which has the configuration
and the SIP transaction.

I have already got the 12.3(9)
and I will flash it in the first occasion
and see if it corrects. In the meantime
by playing a line of a small audio
like transfer, does the trick and fools
the cisco and works correctly.

Thanks, and if you got any clues
by my configs I would greatly
appreciate it.

Savas Pavlidis

Henry Devito wrote:
>We need a little more info such as type of PBX, how it connected to 
>Router type, IOS version, Router config, and at least your extensions.conf.
>
>I have setup several * using different Cisco routers and PBX's and not
had
>any problems. I have a lab where I can setup your config with *.  
>  
>

asterisk users - Oct 2004 - HELP: problem making calls from legacy pbx to cisco sip phone via asterisk

[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk

[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk

[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk

[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk