Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
through a test .. i was able to send calls from Asterisk 1.4 to a PSTN number
through a cisco router with a channel bank.. Audio worked well.. i setup a
dial plan in asterisk to Dial(${ext...@ciscoip) and authorise the cisco
router's ip on the asterisk server and treat the calls comming from it like any
other SIP calls inside the server..
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
Date: Sat, 16 May 2009 14:46:27 +0300
From: timotsm...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Hi,
In our office, we're slowly migrating from a cisco call manager set up
to asterisk. Problem is management doesn't want to buy any other
hardware as they had already invested a lot in cisco. The main cause
of this is asterisk's added features like unique FAX number for
everyone in the company (which will be the same as phone DID), Voice
mail, Auto Answer etc yet we need thousands of dollars to add those to
our cisco call manager 4.1 set up.
I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
and also a dialpeer to forward on the router to forward calls to my
asterisk. It works properly but the problem is there is NO AUDIO! I
have tried to change codec but no sucess!
Has anyone had the above set up working successfully? Attached are some confs.
Thanks a lot for your assistance.
Kind Regards,
Wilson
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Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Thank David and Neeraj for your input.
Neeraj, I posted the configs in my first post, but i've also attached
some extracts here. they haven't changed much.
David, You're absolutely right and i think the problem could be the
reverse dial-peer or DTMF configuration. I think I have the
corresponding reverse dial-peer and the DTMF conf that you said.
However, I have checked my side and all seems to be ok. I've also
tried changing the dtmfmode to sip-notify on the gateway (and info in
sip.conf) but no luck!
Please look at the attached and give me some pointers.
Thanks,
Tim
On Sun, May 17, 2009 at 3:44 PM, David Backeberg dbackeb...@gmail.com wrote:
On Sat, May 16, 2009 at 10:22 AM, Timothy Smith timotsm...@gmail.com wrote:
I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these machines are atleast 2 switches from each other. I just enabled
it (nat=route or nat=yes) and it worked.
It's not yet done however. When I redirect a call to any Asterisk
application, it just hangs up! I have read some history and archives,
but none of the solutions has worked for me. e.g ip inspect udp
idle-time 900. My router (or IOS) doesn't have thet command.
Could you please assist point to what could be causing this and how to
solve it? Below are some logs and attached is the router log.
; This is the extension conf. Enter the extension you want to reach
now (something like auto attendant).
exten = _X.,1,Read(NUM,beep,4,2,3)
exten = _X.,n,Dial(SIP/${NUM})
; This is all i get when i call and the call hangs up!
Did you ever set up that reverse dial-peer? If not, do that first.
You put a three second timeout on the Read(). By any chance, is the
call hanging up 3 seconds after you call? That would be expected
behavior. Well, actually you give it two tries. So it should be
beep
three second wait
beep
three second wait
hangup
If you're actually entering numbers on your dialpad and they're not
getting read, you have a misconfiguration on your DTMF. If you enable
sip debugging on your asterisk side you can see exactly what's coming
over the wire from the Cisco side. There are a lot of choices for DTMF
on the asterisk side and the Cisco side, and they need to agree for
the button presses to be encoded and passed correctly. You can pass
them in-line as real audio, or you can convert them to a special dtmf
sip encoding. You'll notice all those choices when you go to configure
the Cisco dial-peer.
My personal preference:
on the Cisco dial-peer side
dtmf-relay rtp-nte
on the asterisk side
I left the dtmf config blank, and I don't remember which default you
end up with, but it worked in the default config for me.
___
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn negotiate-bchan resend-setup
no cdp enable
!
interface Serial0/0/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn negotiate-bchan resend-setup
no cdp enable
dial-peer voice 1 pots
destination-pattern 0T
port 0/0/1:15
forward-digits all
!
dial-peer voice 3 pots
incoming called-number .
direct-inward-dial
port 0/0/1:15
dial-peer voice 4 pots
incoming called-number .
direct-inward-dial
port 0/0/1:15
!
dial-peer voice 112 voip
destination-pattern 730732888
monitor probe icmp-ping
session protocol sipv2
session target ipv4:172.19.3.150
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
VG2# show dial-peer voice 112
VoiceOverIpPeer112
peer type = voice, system default peer = FALSE, information type =
voice,
description = `',
tag = 112, destination-pattern = `730732888',
voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 112, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = enabled,
modem transport = system,
URI classes:
Incoming (Request) =
Incoming (To) =
Incoming (From) =
Destination =
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
On Sat, May 16, 2009 at 10:22 AM, Timothy Smith timotsm...@gmail.com wrote:
I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these machines are atleast 2 switches from each other. I just enabled
it (nat=route or nat=yes) and it worked.
It's not yet done however. When I redirect a call to any Asterisk
application, it just hangs up! I have read some history and archives,
but none of the solutions has worked for me. e.g ip inspect udp
idle-time 900. My router (or IOS) doesn't have thet command.
Could you please assist point to what could be causing this and how to
solve it? Below are some logs and attached is the router log.
; This is the extension conf. Enter the extension you want to reach
now (something like auto attendant).
exten = _X.,1,Read(NUM,beep,4,2,3)
exten = _X.,n,Dial(SIP/${NUM})
; This is all i get when i call and the call hangs up!
Did you ever set up that reverse dial-peer? If not, do that first.
You put a three second timeout on the Read(). By any chance, is the
call hanging up 3 seconds after you call? That would be expected
behavior. Well, actually you give it two tries. So it should be
beep
three second wait
beep
three second wait
hangup
If you're actually entering numbers on your dialpad and they're not
getting read, you have a misconfiguration on your DTMF. If you enable
sip debugging on your asterisk side you can see exactly what's coming
over the wire from the Cisco side. There are a lot of choices for DTMF
on the asterisk side and the Cisco side, and they need to agree for
the button presses to be encoded and passed correctly. You can pass
them in-line as real audio, or you can convert them to a special dtmf
sip encoding. You'll notice all those choices when you go to configure
the Cisco dial-peer.
My personal preference:
on the Cisco dial-peer side
dtmf-relay rtp-nte
on the asterisk side
I left the dtmf config blank, and I don't remember which default you
end up with, but it worked in the default config for me.
___
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Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
On 16 May 2009, at 12:46, Timothy Smith wrote: blah Has anyone had the above set up working successfully? Attached are some confs. Thanks a lot for your assistance. Check about the sip.conf 'insecure' option. I have had to use it in the past for similar stuff. I think it was 'insecure=very' but that might be deprecated by now.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Thanks Steve for this tip. I have insecure=very is not yet deprecated. I have added it but still no good. I personally think the problem could be with the codecs. Any ideas? I have attached some debug info. Regards, Tim On Sat, May 16, 2009 at 3:25 PM, Steve Howes st...@geekinter.net wrote: On 16 May 2009, at 12:46, Timothy Smith wrote: blah Has anyone had the above set up working successfully? Attached are some confs. Thanks a lot for your assistance. Check about the sip.conf 'insecure' option. I have had to use it in the past for similar stuff. I think it was 'insecure=very' but that might be deprecated by now.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users VG2# VG2# VG2# VG2# VG2# May 16 12:41:40.237: ISDN Se0/0/1:15 Q931: RX - SETUP pd = 8 callref = 0x0C73 Sending Complete Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98392 Exclusive, Channel 18 Calling Party Number i = 0x2183, '730230199' Plan:ISDN, Type:National Called Party Number i = 0xA1, '790792888' Plan:ISDN, Type:National May 16 12:41:40.237: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_INCOMING May 16 12:41:40.237: ISDN Se0/0/1:15 EVENT: call_incoming: call_id 0x33B3, Guid = BCBB464BB098 VG2# May 16 12:41:40.241: fb_get_reject_cause_code: ERROR cause_code NULL May 16 12:41:40.245: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD May 16 12:41:40.245: ISDN Se0/0/1:15 Q931: TX - CALL_PROC pd = 8 callref = 0x8C73 Channel ID i = 0xA98392 Exclusive, Channel 18 May 16 12:41:40.353: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD May 16 12:41:40.353: ISDN Se0/0/1:15 Q931: TX - ALERTING pd = 8 callref = 0x8C73 VG2# May 16 12:41:46.697: ISDN Se0/0/0:15 SERROR: isdn_get_name_from_gtd: false ret May 16 12:41:46.697: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD May 16 12:41:46.697: ISDN Se0/0/1:15 Q931: TX - CONNECT pd = 8 callref = 0x8C73 May 16 12:41:46.713: ISDN Se0/0/1:15 Q931: RX - CONNECT_ACK pd = 8 callref = 0x0C73 May 16 12:41:46.713: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_PROGRESS OULKLAVG2# May 16 12:41:46.713: %ISDN-6-CONNECT: Interface Serial0/0/1:17 is now connected to 730230199 N/A VG2# May 16 12:41:52.373: ISDN Se0/0/1:15 Q931: RX - DISCONNECT pd = 8 callref = 0x0C73 Cause i = 0x8490 - Normal call clearing May 16 12:41:52.373: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_DISC May 16 12:41:52.373: %ISDN-6-CONNECT: Interface Serial0/0/1:17 is now connected to 730230199 N/A May 16 12:41:52.373: %ISDN-6-DISCONNECT: Interface Serial0/0/1:17 disconnected from 730230199 , call lasted 5 seconds VG2# May 16 12:41:52.373: ISDN Se0/0/1:15 Q931: TX - RELEASE pd = 8 callref = 0x8C73 May 16 12:41:52.385: ISDN Se0/0/1:15 Q931: RX - RELEASE_COMP pd = 8 callref = 0x0C73 May 16 12:41:52.385: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_CLEARED From: 730230199 sip:730230...@172.19.3.150;tag=as5f114784 To: sip:1...@172.19.4.102:32544;rinstance=e6a140ee2d1dee0f;tag=ae700477 Contact: sip:730230...@172.19.3.150 Call-ID: 7beff1bd661329c643aa69ec43628...@172.19.3.150 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.9 Content-Length: 0 --- -- SIP/100-00820520 answered SIP/172.17.3.248-007fc920 Audio is at 172.19.3.150 port 13312 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Reliably Transmitting (no NAT) to 172.17.3.248:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK501204;received=172.17.3.248 From: sip:730230...@172.17.3.248;tag=D8FE7BF8-4CA To: sip:730232...@172.19.3.150;tag=as0fb38dd9 Call-ID: 4a137712-414d11de-9606c927-51af5...@172.17.3.248 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:730232...@172.19.3.150 Content-Type: application/sdp Content-Length: 261 v=0 o=root 544232458 544232458 IN IP4 172.19.3.150 s=Asterisk PBX 1.6.0.9 c=IN IP4 172.19.3.150 t=0 0 m=audio 13312 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- Packet2Packet bridging SIP/172.17.3.248-007fc920 and SIP/100-00820520 cs-intranet*CLI --- SIP read from UDP://172.17.3.248:62582 --- ACK sip:730232...@172.19.3.150:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK511A76 From: sip:730230...@172.17.3.248;tag=D8FE7BF8-4CA To:
Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
On Sat, May 16, 2009 at 7:46 AM, Timothy Smith timotsm...@gmail.com wrote: I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9), and also a dialpeer to forward on the router to forward calls to my asterisk. It works properly but the problem is there is NO AUDIO! I have tried to change codec but no sucess! Has anyone had the above set up working successfully? Yes. You have been caught by a not-very-well-documented issue with setting up voice routing on the 3845, and probably other similar Cisco gear. And I'm not sure how you've done your test. This is the closest I've ever seen to a document that explains your problem: http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml Did you have a SIP phone on one side of asterisk and a POTS phone on the outside of the 3845? If you did, and you could talk on both at the same time, I think you would discover in fact that you do have some audio, in fact, one-way audio to be precise. But I don't remember for sure, because it's been a while since I've done this to myself. At any rate, your problem is you have dial-peers to get voice packets out from the 3845 to Cisco, but no dial-peers to get the packets from SIP back to a physical circuit on the 3845. Think about this. What should happen to a call inbound from asterisk, to the 3845? Should it go out an E1 to the outside phones world? If so, you need to build a dial-peer that does that. Until you do, you won't be getting two-way audio. you need another rule something like: dial-peer voice 790792888 pots map this back to the proper E1 circuit A secondary problem could also be with the way you're managing your DSPs. I don't know how many physical DSPs you have in your router, but usually it's a GOOD thing to enable DSP farming. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
David,
Thanks a lot for your input. I will enable DSP farming. Like some
other techies, I just wanted to see it work before i consider others
things.
I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these machines are atleast 2 switches from each other. I just enabled
it (nat=route or nat=yes) and it worked.
It's not yet done however. When I redirect a call to any Asterisk
application, it just hangs up! I have read some history and archives,
but none of the solutions has worked for me. e.g ip inspect udp
idle-time 900. My router (or IOS) doesn't have thet command.
Could you please assist point to what could be causing this and how to
solve it? Below are some logs and attached is the router log.
; This is the extension conf. Enter the extension you want to reach
now (something like auto attendant).
exten = _X.,1,Read(NUM,beep,4,2,3)
exten = _X.,n,Dial(SIP/${NUM})
; This is all i get when i call and the call hangs up!
cs-intranet*CLI
== Using SIP RTP CoS mark 5
-- Executing [730732...@default:1]
Read(SIP/172.17.3.248-30069280, NUM,beep,4,2,3) in new stack
-- Accepting a maximum of 4 digits.
== Using SIP RTP CoS mark 5
-- Executing [730732...@default:1]
Read(SIP/172.17.3.248-30069280, NUM,beep,4,2,3) in new stack
-- Accepting a maximum of 4 digits.
cs-intranet*CLI
Thanks alot for your assistance.
On Sat, May 16, 2009 at 4:02 PM, David Backeberg dbackeb...@gmail.com wrote:
On Sat, May 16, 2009 at 7:46 AM, Timothy Smith timotsm...@gmail.com wrote:
I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
and also a dialpeer to forward on the router to forward calls to my
asterisk. It works properly but the problem is there is NO AUDIO! I
have tried to change codec but no sucess!
Has anyone had the above set up working successfully?
Yes.
You have been caught by a not-very-well-documented issue with setting
up voice routing on the 3845, and probably other similar Cisco gear.
And I'm not sure how you've done your test.
This is the closest I've ever seen to a document that explains your problem:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml
Did you have a SIP phone on one side of asterisk and a POTS phone on
the outside of the 3845?
If you did, and you could talk on both at the same time, I think you
would discover in fact that you do have some audio, in fact, one-way
audio to be precise. But I don't remember for sure, because it's been
a while since I've done this to myself.
At any rate, your problem is you have dial-peers to get voice packets
out from the 3845 to Cisco, but no dial-peers to get the packets from
SIP back to a physical circuit on the 3845. Think about this. What
should happen to a call inbound from asterisk, to the 3845? Should it
go out an E1 to the outside phones world? If so, you need to build a
dial-peer that does that. Until you do, you won't be getting two-way
audio.
you need another rule something like:
dial-peer voice 790792888 pots
map this back to the proper E1 circuit
A secondary problem could also be with the way you're managing your
DSPs. I don't know how many physical DSPs you have in your router, but
usually it's a GOOD thing to enable DSP farming.
___
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May 16 14:18:28.640: ISDN Se0/0/1:15 Q931: RX - RELEASE_COMP pd = 8 callref =
0x0CAB
Cause i = 0x80D1 - Invalid call reference value
May 16 14:18:28.640: ISDN Se0/0/1:15 SERROR: L3_GetUser_NLCB: EVENT 0X5A No
NLCB 2
May 16 14:18:28.640: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr
0x8CAB callid 0x0
May 16 14:18:28.660: ISDN Se0/0/1:15 Q931: RX - SETUP pd = 8 callref = 0x0CAC
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9838D
Exclusive, Channel 13
Calling Party Number i = 0x2183, '730730199'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '730732888'
Plan:ISDN, Type:National
May 16 14:18:28.664: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x3407 calltype 2 CALL_INCOMING
May 16 14:18:28.664: ISDN Se0/0/1:15 EVENT: call_incoming: call_id 0x3407, Guid
= 42D2FC70B0D1
May 16 14:18:28.664: fb_get_reject_cause_code: ERROR cause_code NULL
May 16 14:18:28.668: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD
May 16 14:18:28.668: ISDN Se0/0/1:15 Q931: TX - CALL_PROC pd = 8 callref =
0x8CAC
Channel ID i = 0xA9838D
Exclusive, Channel 13
May 16 14:18:28.676:
Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Steve Howes schrieb: Check about the sip.conf 'insecure' option. I have had to use it in the past for similar stuff. I think it was 'insecure=very' but that might be deprecated by now.. insecure=very should now be written as insecure=port,invite Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
