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Jerry Geis

2008-Nov-07 15:17 UTC

[asterisk-users] help with dialplan

I have a small system, server, client and 2 phones. Phones are polycom 
501's.
In general all is working fine. I can call the two phones, speak etc...
I can have the server call each phone and play a wave file.

However, when trying to setup a direct dial number of 1044 that
calls another machine running asterisk - something ODD is happening.

; This is not working....
[smvoice-sip]

exten => 1044,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044)
exten => 1044,n,Hangup

; changing 1044 to 10 works find.
[smvoice-sip]

exten => 10,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044)
exten => 10,n,Hangup


I am running 1.4.22 and DAHDI 2.0.0 complete.

Why is it picking up "10" when trying to dial "1044".

How can I determine what is going on here. Thanks,

Jerry

This is the SIP debug for the 1044 case that does not work.
-----------------

Use 'exit' when done

Asterisk 1.4.22, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
========================================================================  ==
Parsing '/etc/asterisk/asterisk.conf': Found
[0;37;40m[1;30;40m  == [0;37;40mParsing '/etc/asterisk/extconfig.conf':
Found
Connected to Asterisk 1.4.22 currently running on devcentos5x64 (pid = 3127)
devcentos5x64*CLI> 
Verbosity is at least 5

[Kdevcentos5x64*CLI> 
<--- SIP read from 192.168.1.89:5060 --->
INVITE sip:10 at 192.168.1.8;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bKb872214aDBECCC5D
From: "404" <sip:404 at 192.168.1.8>;tag=25AB8538-7BACFE71
To: <sip:10 at 192.168.1.8;user=phone>
CSeq: 1 INVITE
Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89
Contact: <sip:404 at 192.168.1.89>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258
Supported: 1?00rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 1226069152 1226069152 IN IP4 192.168.1.89
s=Polycom IP Phone
c=IN IP4 192.168.1.89
t=0 0
m=audio 2244 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.1.89 : 5060 (no NAT)
Using INVITE request as basis request - a4b0cbb4???-9737882e-815856ff at
192.168.1.89

<--- Reliably Transmitting (no NAT) to 192.168.1.89:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.1.89;branch=z9hG4bKb872214aDBECCC5D;received=192.168.1.89
From: "404" <sip:404 at 192.168.1.8>;tag=25AB8538-7BACFE71
To: <sip:10 at 192.168.1.8;user=phone>;tag=as5a3d998e
Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces?
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="1f1b706f"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a4b0cbb4-9737882e-815856ff at
192.168.1.89' in 32000 ms (Method: INVITE)
Found user '404'
??
<--- SIP read from 192.168.1.89:5060 --->
ACK sip:10 at 192.168.1.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bKb872214aDBECCC5D
From: "404" <sip:404 at 192.168.1.8>;tag=25AB8538-7BACFE71
To: <sip:10 at 192.168.1.8;user=phone>;tag=as5a3d998e
CSeq: 1 ACK
Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89
Contact: <sip:404 at 192.168.1.89>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258
Max-Forwards: ?70
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
?
[Kdevcentos5x64*CLI> 
<--- SIP read from 192.168.1.89:5060 --->
INVITE sip:10 at 192.168.1.8;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bK9c456f3360D09552
From: "404" <sip:404 at 192.168.1.8>;tag=25AB8538-7BACFE71
To: <sip:10 at 192.168.1.8;user=phone>
CSeq: 2 INVITE
Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89
Contact: <sip:404 at 192.168.1.89>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258
Supported: 1?00rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="404",
realm="asterisk", nonce="1f1b706f", uri="sip:10 at
192.168.1.8;user=phone",
response="c6e14f94fa0bbe3d742b6f570982ed79", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 1226069152 1226069152 IN IP4 192.168.1.89
s=Polycom IP Phone
c=IN IP4 192.168.1.89
t=0 0
m=audio 2244 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (15 headers 11 lines) ---
Sending to 192.168.1.89 : 5060 (no NAT)
Using INVITE request as basis request - a4b0cbb4-9737882e-815856ff at
192.168.1.89
Found user '404'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.89:2244
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for???????????? ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.89:2244
Looking for 10 in smvoice-sip (domain 192.168.1.8)

<--- Reliably Transmitting (no NAT) to 192.168.1.89:5060 --->
SIP/2.0 484 Address Incomplete
Via:?????? SIP/2.0/UDP
192.168.1.89;branch=z9hG4bK9c456f3360D09552;received=192.168.1.89
From: "404" <sip:404 at 192.168.1.8>;tag=25AB8538-7BACFE71
To: <sip:10 at 192.168.1.8;user=phone>;tag=as5a3d998e
Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a4b0cbb4-9737882e-815856ff at
192.168.1.89' in 32000 ms (Method: INV?ITE)

[Kdevcentos5x64*CLI> 
<--- SIP read from 192.168.1.89:5060 --->
ACK sip:10 at 192.168.1.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bK9c456f3360D09552
From: "404" <sip:404 at 192.168.1.8>;tag=25AB8538-7BACFE71
To: <sip:10 at 192.168.1.8;user=phone>;tag=as5a3d998e
CSeq: 2 ACK
Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89
Contact: <sip:404 at 192.168.1.89>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258
Proxy-Authoriz?ation: Digest username="404",
realm="asterisk", nonce="1f1b706f", uri="sip:10 at
192.168.1.8;user=phone",
response="c6e14f94fa0bbe3d742b6f570982ed79", algorithm=MD5
Max-Forwards: 70
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
?
[Kdevcentos5x64*CLI> 
<--- SIP read from 192.168.1.89:5060 --->
INVITE sip:1044 at 192.168.1.8;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bK51dc2d00EB13255B
From: "404" <sip:404 at 192.168.1.8>;tag=25AB8538-7BACFE71
To: <sip:1044 at 192.168.1.8;user=phone>
CSeq: 3 INVITE
Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89
Contact: <sip:404 at 192.168.1.89>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258
Supporte?d: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="404",
realm="asterisk", nonce="1f1b706f", uri="sip:10 at
192.168.1.8;user=phone",
response="75209201fe7d6f0854ecb918879e6049", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 1226069152 1226069153 IN IP4 192.168.1.89
s=Polycom IP Phone
c=IN IP4 192.168.1.89
t=0 0
m=audio 2244 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (15 headers 11 lines) ---
Sending to 192.168.1.89 : 5060 (no NAT)
Using INVITE request as basis request - a4b0cbb4-9737882e-815856ff at
192.168.1.89
Found user '404'
[Nov  7 09:48:02] NOTICE[3145]: chan_sip.c:14316 handle_request_invite: Failed
to authenticate user "404" <sip:404 at
192.168.1.8>;tag=25AB8538-7BACFE71

<--- Reliably Transmitting (no NAT) to 192.168.1.89:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
192.168.1.89;branch=z9hG4b?????K51dc2d00EB13255B;received=192.168.1.89
From: "404" <sip:404 at 192.168.1.8>;tag=25AB8538-7BACFE71
To: <sip:1044 at 192.168.1.8;user=phone>;tag=as5a3d998e
Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a4b0cbb4-9737882e-815856ff at
192.168.1.89' in 32000 ms (Method: INVITE)
?
[Kdevcentos5x64*CLI> 
<--- SIP read from 192.168.1.89:5060 --->
ACK sip:1044 at 192.168.1.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bK51dc2d00EB13255B
From: "404" <sip:404 at 192.168.1.8>;tag=25AB8538-7BACFE71
To: <sip:1044 at 192.168.1.8;user=phone>;tag=as5a3d998e
CSeq: 3 ACK
Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89
Contact: <sip:404 at 192.168.1.89>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258
Proxy-Auth?orization: Digest username="404",
realm="asterisk", nonce="1f1b706f", uri="sip:10 at
192.168.1.8;user=phone",
response="c6e14f94fa0bbe3d742b6f570982ed79", algorithm=MD5
Max-Forwards: 70
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
?
[Kdevcentos5x64*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).

Brent Davidson

2008-Nov-07 15:45 UTC

head link

[asterisk-users] help with dialplan

Jerry Geis wrote:> I have a small system, server, client and 2 phones. Phones are polycom 
> 501's.
> In general all is working fine. I can call the two phones, speak etc...
> I can have the server call each phone and play a wave file.
>
> However, when trying to setup a direct dial number of 1044 that
> calls another machine running asterisk - something ODD is happening.
>
> ; This is not working....
> [smvoice-sip]
>
> exten => 1044,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044)
> exten => 1044,n,Hangup
>
> ; changing 1044 to 10 works find.
> [smvoice-sip]
>
> exten => 10,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044)
> exten => 10,n,Hangup
>
>
> I am running 1.4.22 and DAHDI 2.0.0 complete.
>
> Why is it picking up "10" when trying to dial "1044".
>
> How can I determine what is going on here. Thanks,
>
> Jerry
>
> <debug snipped>
>   
Are your polycom phones set up for overlap dialing or do you dial the 
number then press a key to dial?

Jerry Geis

2008-Nov-07 15:51 UTC

head link

[asterisk-users] help with dialplan

>
> Are your polycom phones set up for overlap dialing or do you dial the 
> number then press a key to dial?
>
>    From you message I tried a couple things...

Clicking New call, then starting to dial this is when it messes up.

when I start entering the number first then click dial this successfull
does the 1044 and I am connected as I thought.

How do I turn off this overlap dial?

Thanks so much.

jerry

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