[asterisk-users] Hash Dial Pattern Problems
David Nickel
dnickel at gmail.com
Wed May 5 17:04:44 CDT 2010
I set: sip debug peer 3000 (my test extension) and dialed #3643873
Here is the output:
<-- SIP read from 192.168.1.59:17456:
INVITE sip:%233643873 at 192.168.2.10 <sip%3A%25233643873 at 192.168.2.10> SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:17456
;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:3000 at 192.168.1.59:17456>
To: "#3643873"<sip:%233643873 at 192.168.2.10 <sip%3A%25233643873 at 192.168.2.10>
>
From: "Test"<sip:3000 at 192.168.2.10 <sip%3A3000 at 192.168.2.10>>;tag=76126b35
Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
Proxy-Authorization: Digest
username="3000",realm="asterisk",nonce="6a7a2c99",uri="
sip:%233643873 at 192.168.2.10 <sip%3A%25233643873 at 192.168.2.10>
",response="7bcf9339c154ef939bd575aeaaef1860",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 317
v=0
o=- 8 2 IN IP4 192.168.1.59
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.59
t=0 0
m=audio 34194 RTP/AVP 107 0 8 101
a=alt:1 2 : MsRCET/S fNqrHReN 192.168.200.113 34194
a=alt:2 1 : pf8wX3Si UdjtGUj2 192.168.1.59 34194
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (13 headers 12 lines)---
Using INVITE request as basis request -
NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.
Sending to 192.168.1.59 : 17456 (non-NAT)
Found user '3000'
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.59:34194
Found description format BV32
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for %233643873 in from-internal (domain 192.168.2.10)
Reliably Transmitting (no NAT) to 192.168.1.59:17456:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.59:17456
;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport;received=192.168.1.59
From: "Test"<sip:3000 at 192.168.2.10 <sip%3A3000 at 192.168.2.10>>;tag=76126b35
To: "#3643873"<sip:%233643873 at 192.168.2.10 <sip%3A%25233643873 at 192.168.2.10>
>;tag=as3d020428
Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:%233643873 at 192.168.2.10 <sip%3A%25233643873 at 192.168.2.10>>
Content-Length: 0
---
aikphone*CLI>
<-- SIP read from 192.168.1.59:17456:
ACK sip:%233643873 at 192.168.2.10 <sip%3A%25233643873 at 192.168.2.10> SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:17456
;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport
To: "#3643873"<sip:%233643873 at 192.168.2.10 <sip%3A%25233643873 at 192.168.2.10>
>;tag=as3d020428
From: "Hull Barrett"<sip:3000 at 192.168.2.10 <sip%3A3000 at 192.168.2.10>
>;tag=76126b35
Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.
CSeq: 2 ACK
Content-Length: 0
--- (7 headers 0 lines)---
Destroying call 'NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.'
On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas <danny at debsinc.com> wrote:
> Ok. I’m confused. I was interpreting what you wrote to say that you are
> doing this:
>
> 1. pick up sip phone attached to pbx1 (1.2 box)
> 2. dial #5551212
> 3. command dial(iax2/trunk/5551212,30,r) gets executed on 1.2 box
> 4. 1.4 box should fall into _XXXXXXX and do DAHDI dial?
>
>
>
> If this is correct, where is the IAX command in your CLI output.
>
>
> ------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *David Nickel
> *Sent:* Wednesday, May 05, 2010 10:11 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
>
>
>
> I am on the 1.2 box and see nothing with the verbose cranked up. I do see
> the following when tailing the asterisk full log during the calls:
>
> May 5 11:09:46 DEBUG[26538] chan_sip.c: Setting NAT on RTP to 0
>
> May 5 11:09:46 DEBUG[26538] chan_sip.c: Checking SIP call limits for
> device 3000
>
> May 5 11:09:46 DEBUG[26538] chan_sip.c: Stopping retransmission on
> 'Njg3MjI5N2IzNDk3NWYxZTMzMzFmMzEwNzc2ZDE1NTE.' of Response 2: Match Found
>
>
>
>
>
> On Wed, May 5, 2010 at 10:39 AM, Danny Nicholas <danny at debsinc.com> wrote:
>
> Ok – you have to be getting something or you wouldn’t get that message.
> You are looking at CLI on the 1.2 or 1.4 box? If you’re looking at the 1.4
> side, you won’t see anything until a connection is made (although you should
> see some kind of credential reject or something??)
>
>
> ------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *David Nickel
> *Sent:* Wednesday, May 05, 2010 9:31 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
>
>
>
> Nothing..goes directly to "The person you are calling is unavailable".
>
> On Wed, May 5, 2010 at 9:46 AM, Danny Nicholas <danny at debsinc.com> wrote:
>
> Set verbose to 5 and see if you get a CLI output.
>
>
> ------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *David Nickel
> *Sent:* Wednesday, May 05, 2010 8:39 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
>
>
>
> I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
>
> The other box is running 1.2.1
>
> Thanks,
>
> David
>
> On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas <danny at debsinc.com> wrote:
>
> Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
> other 2 1.4.30 boxes wouldn’t talk to it properly.
>
>
> ------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *David Nickel
> *Sent:* Wednesday, May 05, 2010 8:23 AM
> *To:* asterisk-users at lists.digium.com
> *Subject:* [asterisk-users] Hash Dial Pattern Problems
>
>
>
> I have two Asterisk boxe. One is running 1.6 and the other 1.2
>
> The users on the 1.2 system press # plus a local 7 digit number to place
> local calls through the trunk to the 1.6 box.
>
> For some reason this dial pattern fails right away with "unavailable".
> There is no activity in the CLI. Other patterns for the trunk work just
> fine.
>
> Dial pattern:
> #|. or #|NXXXXXX
>
> exten => _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
> exten => _#.,2,Congestion
>
> I have been beating my end with the problem for three days. Any suggestions
> would be much appreciated.
>
>
> --
> _____________________________________________________________________
> -- 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
>
>
>
>
> --
> _____________________________________________________________________
> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100505/cf2ec207/attachment.htm
More information about the asterisk-users
mailing list