Re: [asterisk-users] Hash Dial Pattern Problems
Now what does the 1.4 side (CLI) look like when you do this call?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Friday, May 07, 2010 7:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
I changed the dial pattern to %23|XXX and dialed #1234567. I was able to
trigger activity in the CLI:
Connected to Asterisk 1.2.1 currently running on aikphone (pid = 29352)
Verbosity is at least 22
-- Executing Macro("SIP/3000-ca1c", "dialout-trunk|3|3643873|") in new
stack
-- Executing GotoIf("SIP/3000-ca1c", "1?3:2)") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/3000-ca1c", "user-callerid") in new stack
-- Executing DBget("SIP/3000-ca1c", "AMPUSER=DEVICE/3000/user") in new
stack
-- DBget: varname=AMPUSER, family=DEVICE, key=3000/user
-- DBget: set variable AMPUSER to 3000
-- Executing DBget("SIP/3000-ca1c",
"AMPUSERCIDNAME=AMPUSER/3000/cidname") in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=3000/cidname
-- DBget: set variable AMPUSERCIDNAME to Augusta I.T Tes
-- Executing GotoIf("SIP/3000-ca1c", "0?5") in new stack
-- Executing SetCallerID("SIP/3000-ca1c", ""Augusta I.T Tes" <3000>") in
new stack
-- Executing NoOp("SIP/3000-ca1c", "Using CallerID "Augusta I.T Tes"
<3000>") in new stack
-- Executing Macro("SIP/3000-ca1c", "record-enable|3000|OUT") in new
stack
-- Executing GotoIf("SIP/3000-ca1c", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/3000-ca1c",
"recordingcheck|20100507-082747|1273235267.398") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20100507-082747|1273235267.398: Outbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/3000-ca1c", "No recording needed") in new stack
-- Executing Macro("SIP/3000-ca1c", "outbound-callerid|3") in new stack
-- Executing DBget("SIP/3000-ca1c",
"USEROUTCID=AMPUSER/3000/outboundcid") in new stack
-- DBget: varname=USEROUTCID, family=AMPUSER, key=3000/outboundcid
-- DBget: set variable USEROUTCID to
-- Executing GotoIf("SIP/3000-ca1c", "1?4") in new stack
-- Goto (macro-outbound-callerid,s,4)
-- Executing GotoIf("SIP/3000-ca1c", "1?6") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing NoOp("SIP/3000-ca1c", "CallerID set to "Augusta I.T Tes"
<3000>") in new stack
-- Executing SetGroup("SIP/3000-ca1c", "OUT_3") in new stack
-- Executing CheckGroup("SIP/3000-ca1c", "") in new stack
-- Executing SetVar("SIP/3000-ca1c", "DIAL_NUMBER=3643873") in new stack
-- Executing SetVar("SIP/3000-ca1c", "DIAL_TRUNK=3") in new stack
-- Executing AGI("SIP/3000-ca1c", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("SIP/3000-ca1c", "OUTNUM=3643873") in new stack
-- Executing Cut("SIP/3000-ca1c", "custom=OUT_3|:|1") in new stack
-- Executing GotoIf("SIP/3000-ca1c", "0?16") in new stack
-- Executing Dial("SIP/3000-ca1c", "IAX2/augusta/3643873") in new stack
-- Called augusta/3643873
-- Call accepted by 192.168.1.10 (format ulaw)
-- Format for call is ulaw
-- IAX2/augusta-16384 is making progress passing it to SIP/3000-ca1c
-- Hungup 'IAX2/augusta-16384'
== Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on
'SIP/3000-ca1c' in macro 'dialout-trunk'
== Spawn extension (from-internal, %233643873, 1) exited non-zero on
'SIP/3000-ca1c'
-- Executing Macro("SIP/3000-ca1c", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3000-ca1c", "w") in new stack
-- Executing NoCDR("SIP/3000-ca1c", "") in new stack
-- Executing Wait("SIP/3000-ca1c", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/3000-ca1c' in macro 'hangupcall'
== Spawn extension (from-inter
Re: [asterisk-users] Hash Dial Pattern Problems
I changed the dial pattern to %23|XXX and dialed #1234567. I was able to
trigger activity in the CLI:
Connected to Asterisk 1.2.1 currently running on aikphone (pid = 29352)
Verbosity is at least 22
-- Executing Macro("SIP/3000-ca1c", "dialout-trunk|3|3643873|") in new
stack
-- Executing GotoIf("SIP/3000-ca1c", "1?3:2)") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/3000-ca1c", "user-callerid") in new stack
-- Executing DBget("SIP/3000-ca1c", "AMPUSER=DEVICE/3000/user") in new
stack
-- DBget: varname=AMPUSER, family=DEVICE, key=3000/user
-- DBget: set variable AMPUSER to 3000
-- Executing DBget("SIP/3000-ca1c",
"AMPUSERCIDNAME=AMPUSER/3000/cidname") in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=3000/cidname
-- DBget: set variable AMPUSERCIDNAME to Augusta I.T Tes
-- Executing GotoIf("SIP/3000-ca1c", "0?5") in new stack
-- Executing SetCallerID("SIP/3000-ca1c", ""Augusta I.T Tes" <3000>") in
new stack
-- Executing NoOp("SIP/3000-ca1c", "Using CallerID "Augusta I.T Tes"
<3000>") in new stack
-- Executing Macro("SIP/3000-ca1c", "record-enable|3000|OUT") in new
stack
-- Executing GotoIf("SIP/3000-ca1c", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/3000-ca1c",
"recordingcheck|20100507-082747|1273235267.398") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20100507-082747|1273235267.398: Outbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/3000-ca1c", "No recording needed") in new stack
-- Executing Macro("SIP/3000-ca1c", "outbound-callerid|3") in new stack
-- Executing DBget("SIP/3000-ca1c",
"USEROUTCID=AMPUSER/3000/outboundcid") in new stack
-- DBget: varname=USEROUTCID, family=AMPUSER, key=3000/outboundcid
-- DBget: set variable USEROUTCID to
-- Executing GotoIf("SIP/3000-ca1c", "1?4") in new stack
-- Goto (macro-outbound-callerid,s,4)
-- Executing GotoIf("SIP/3000-ca1c", "1?6") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing NoOp("SIP/3000-ca1c", "CallerID set to "Augusta I.T Tes"
<3000>") in new stack
-- Executing SetGroup("SIP/3000-ca1c", "OUT_3") in new stack
-- Executing CheckGroup("SIP/3000-ca1c", "") in new stack
-- Executing SetVar("SIP/3000-ca1c", "DIAL_NUMBER=3643873") in new stack
-- Executing SetVar("SIP/3000-ca1c", "DIAL_TRUNK=3") in new stack
-- Executing AGI("SIP/3000-ca1c", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("SIP/3000-ca1c", "OUTNUM=3643873") in new stack
-- Executing Cut("SIP/3000-ca1c", "custom=OUT_3|:|1") in new stack
-- Executing GotoIf("SIP/3000-ca1c", "0?16") in new stack
-- Executing Dial("SIP/3000-ca1c", "IAX2/augusta/3643873") in new stack
-- Called augusta/3643873
-- Call accepted by 192.168.1.10 (format ulaw)
-- Format for call is ulaw
-- IAX2/augusta-16384 is making progress passing it to SIP/3000-ca1c
-- Hungup 'IAX2/augusta-16384'
== Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on
'SIP/3000-ca1c' in macro 'dialout-trunk'
== Spawn extension (from-internal, %233643873, 1) exited non-zero on
'SIP/3000-ca1c'
-- Executing Macro("SIP/3000-ca1c", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3000-ca1c", "w") in new stack
-- Executing NoCDR("SIP/3000-ca1c", "") in new stack
-- Executing Wait("SIP/3000-ca1c", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/3000-ca1c' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/3000-ca1c'
It stripped the hash and passed the number through the IAX2 trunk. I am just
getting a all "circuits are busy".
Thanks,
David
On Wed, May 5, 2010 at 6:53 PM, Philipp von Klitzing <
klitz...@pool.informatik.rwth-aachen.de> wrote:
> Hi!
>
> > I set: sip debug peer 3000 (my test extension) and dialed #3643873
>
> Your X-Lite softphone actually calls %233643873 and not #3643873.
> You would need to check the SIP RFCs in order to find out if Asterisk is
> behaving correctly here by not decoding %23 as #.
>
> In the meanwhile you could try to add the extension %233643873 to your
> dialplan, or find out if you can configure the way X-Lite handles the #
> within the dialstring.
>
> > To: "#3643873"
> >
> > ...
> > User-Agent: X-Lite release 1104o stamp 56125
>
> > (telephone-event) Looking for %233643873 in from-internal (domain
> > ...
> > SIP/2.0 404 Not Found Via: SIP/2.0/UDP
>
> Philipp
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a liv
Re: [asterisk-users] Hash Dial Pattern Problems
Hi! > I set: sip debug peer 3000 (my test extension) and dialed #3643873 Your X-Lite softphone actually calls %233643873 and not #3643873. You would need to check the SIP RFCs in order to find out if Asterisk is behaving correctly here by not decoding %23 as #. In the meanwhile you could try to add the extension %233643873 to your dialplan, or find out if you can configure the way X-Lite handles the # within the dialstring. > To: "#3643873" > ... > User-Agent: X-Lite release 1104o stamp 56125 > (telephone-event) Looking for %233643873 in from-internal (domain > ... > SIP/2.0 404 Not Found Via: SIP/2.0/UDP Philipp -- _ -- 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
Re: [asterisk-users] Hash Dial Pattern Problems
It doesnt seem to like the _X. . What is this suppose represent?
Thanks
On Wed, May 5, 2010 at 5:53 PM, Danny Nicholas wrote:
> From 1.2 CLI, do “dialplan show _...@default – this will tell you if your
> expected context is valid (may not work on 1.2, I started this ride at 1.4
> and therefore have no backward knowledge).
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
> *Sent:* Wednesday, May 05, 2010 4:41 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
>
>
>
> Your interpretation is right ownvery weird problem. The problem is
> when i dial #551212 there is absolutely no activity in the CLI. It is almost
> like there is a conflict somewhere.
>
> On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas 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 _XXX and do DAHDI dial?
>
>
>
> If this is correct, where is the IAX command in your CLI output.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 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-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 wrote:
>
> Set verbose to 5 and see if you get a CLI output.
>
>
> ----------
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 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-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
> *Sent:* Wednesday, May 05, 2010 8:23 AM
> *To:* asterisk-users@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 #|NXX
>
> exten => _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
> exten => _#.,2,Congestion
>
> I have been beating my end with the problem for three da
Re: [asterisk-users] Hash Dial Pattern Problems
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:%233643...@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: To: "#3643873" > From: "Test">;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:%233643...@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">;tag=76126b35 To: "#3643873" >;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: > Content-Length: 0 --- aikphone*CLI> <-- SIP read from 192.168.1.59:17456: ACK sip:%233643...@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" >;tag=as3d020428 From: "Hull Barrett" >;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 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 _XXX and do DAHDI dial? > > > > If this is correct, where is the IAX command in your CLI output. > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@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 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-boun...@lists.digium.com [mailto: > asterisk-users-boun...@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&qu
Re: [asterisk-users] Hash Dial Pattern Problems
>From 1.2 CLI, do "dialplan show _...@default - this will tell you if your
expected context is valid (may not work on 1.2, I started this ride at 1.4
and therefore have no backward knowledge).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
Your interpretation is right ownvery weird problem. The problem is when
i dial #551212 there is absolutely no activity in the CLI. It is almost like
there is a conflict somewhere.
On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas 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 _XXX and do DAHDI dial?
If this is correct, where is the IAX command in your CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 wrote:
Set verbose to 5 and see if you get a CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:23 AM
To: asterisk-users@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 #|NXX
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.
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Re: [asterisk-users] Hash Dial Pattern Problems
Your interpretation is right ownvery weird problem. The problem is when
i dial #551212 there is absolutely no activity in the CLI. It is almost like
there is a conflict somewhere.
On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas 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 _XXX and do DAHDI dial?
>
>
>
> If this is correct, where is the IAX command in your CLI output.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 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-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 wrote:
>
> Set verbose to 5 and see if you get a CLI output.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 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-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
> *Sent:* Wednesday, May 05, 2010 8:23 AM
> *To:* asterisk-users@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 #|NXX
>
> 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 mai
Re: [asterisk-users] Hash Dial Pattern Problems
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 _XXX and do DAHDI dial?
If this is correct, where is the IAX command in your CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 wrote:
Set verbose to 5 and see if you get a CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:23 AM
To: asterisk-users@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 #|NXX
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.
--
_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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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 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-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 wrote:
>
> Set verbose to 5 and see if you get a CLI output.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 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-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
> *Sent:* Wednesday, May 05, 2010 8:23 AM
> *To:* asterisk-users@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 #|NXX
>
> 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
>
--
_
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To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Hash Dial Pattern Problems
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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 wrote:
Set verbose to 5 and see if you get a CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:23 AM
To: asterisk-users@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 #|NXX
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
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asterisk-users mailing list
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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 wrote:
> Set verbose to 5 and see if you get a CLI output.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 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-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
> *Sent:* Wednesday, May 05, 2010 8:23 AM
> *To:* asterisk-users@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 #|NXX
>
> 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
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Re: [asterisk-users] Hash Dial Pattern Problems
Set verbose to 5 and see if you get a CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:23 AM
To: asterisk-users@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 #|NXX
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:
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--
_
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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 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-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
> *Sent:* Wednesday, May 05, 2010 8:23 AM
> *To:* asterisk-users@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 #|NXX
>
> exten => _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
> exten => _#.,2,Congestion
>
> I have been beating my end with the problem for three days. Any suggestions
> would be much appreciated.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Hash Dial Pattern Problems
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
other 2 1.4.30 boxes wouldn't talk to it properly.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with "unavailable". There
is no activity in the CLI. Other patterns for the trunk work just fine.
Dial pattern:
#|. or #|NXX
exten => _#.,1,Dial(IAX2/trunk/${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
